Re: [asterisk-users] So long, and thanks for all the fish!

2012-07-31 Thread Alex Oniciuc 
No Kevin, we thank you for the fish, for sharing the knowledge and for
having the patience...
Good luck on your new journey!

P.S. As a token of our appreciation, just a word and we'll make the life of
the new Director of Software Technologies miserable!


2012/7/31 Kevin P. Fleming 

> I've been with Digium for just over seven years, and it's been an
> incredible experience that I wouldn't have traded for anything. When
> Mark Spencer invited me to visit Digium (and Huntsville) in early
> 2005, I could not have dreamed that I'd end up working for such an
> exciting, innovative company, finding a wife, and meeting hundreds of
> people (many of whom are now friends) around the world. It's been a
> time of tremendous personal and career growth, and my wonderful
> colleagues at Digium and in the Asterisk open source community have
> been directly responsible for most of that.
>
> Recently, though, I've been presented an opportunity to take on a new
> challenge and this has resulted in my acceptance of a new job, in a
> new industry. In the middle of September, I'll start working for
> Bloomberg, L.P., in the Office of the CTO, helping to lead their
> nascent open source initiative. I'll be working to bring the power of
> open source software, open standards, and community building to the
> financial market data services industry, where it is sorely needed
> (and overdue). Michelle and I will be relocating to the greater New
> York City area, but Michelle will continue in her role as Digium's
> in-house counsel. Because of our need to relocate, I'll only be at
> Digium until August 8th, although I'll be in Huntsville until around
> Labor Day.
>
> This is yet another incredibly exciting, career changing opportunity
> in my life, and I can't wait to see what it will bring. I'll be
> forever thankful for the opportunity that Digium and the Asterisk
> community provided me to learn, grow and find the place where my skills
> and experience are the most valuable (to both myself and my employer).
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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[asterisk-users] Problem provisioning Cisco SPA303

2012-07-31 Thread Support 
Hello.
I've got a Cisco SPA303 that I'm trying to provision via http.  I noticed that 
this device looks very similar to a PAP2T, so I used that as a template for my 
provisioning file.

However, the result is less than stellar.  Line 1 registers and works.  
However, lines 2 and 3 also register as line 1, effectively giving me a 1-line 
phone with 3 buttons.

Also, the line name is the same on all 3 phone lies.

I've looked on Cisco's website and Googled around, but I can not find a true 
example of a provisioning file for this device.  Anything I could find would be 
enough for me to make a template.

Does anyone know of an example provisioning file for this device?

I've included, below, a (sanitized) copy of the file I'm using.  The user_id is 
simply the device's MAC address, followed by a dash, then the line number.

Any help you could give me would be most appreciated.


  

hostnamexxx



No
No

3CCE73D241A9-1
3CCE73D241A9-1
3CCE73D241A9-1
DICLFUXOT1SWNQEW
Yes
yes
Yes

3CCE73D241A9-2
3CCE73D241A9-2
3CCE73D241A9-2
KVJJZPAMSI[WM1ISR
Yes
yes
Yes

 example.com 
example.com 

 example.com 
 example.com 

 http://example.com/index.pl?mac=$MA 

0
0
0




-- 

Mike Diehl.

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Re: [asterisk-users] So long, and thanks for all the fish!

2012-07-31 Thread Raj Mathur (राज माथुर) 
On Tuesday 31 Jul 2012, Kevin P. Fleming wrote:
> [snip]
> This is yet another incredibly exciting, career changing opportunity
> in my life, and I can't wait to see what it will bring. I'll be
> forever thankful for the opportunity that Digium and the Asterisk
> community provided me to learn, grow and find the place where my
> skills and experience are the most valuable (to both myself and my
> employer).

Thanks for all the pertinent and helpful advice and suggestions on the 
list, Kevin, and all the best in your new assignment.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] So long, and thanks for all the fish!

2012-07-31 Thread Duncan Turnbull 

On 1/08/2012, at 1:59 AM, "Kevin P. Fleming"  wrote:

> I've been with Digium for just over seven years, and it's been an
> incredible experience that I wouldn't have traded for anything. When
> Mark Spencer invited me to visit Digium (and Huntsville) in early
> 2005, I could not have dreamed that I'd end up working for such an
> exciting, innovative company, finding a wife, and meeting hundreds of
> people (many of whom are now friends) around the world. It's been a
> time of tremendous personal and career growth, and my wonderful
> colleagues at Digium and in the Asterisk open source community have
> been directly responsible for most of that.
> 
> Recently, though, I've been presented an opportunity to take on a new
> challenge and this has resulted in my acceptance of a new job, in a
> new industry. In the middle of September, I'll start working for
> Bloomberg, L.P., in the Office of the CTO, helping to lead their
> nascent open source initiative. I'll be working to bring the power of
> open source software, open standards, and community building to the
> financial market data services industry, where it is sorely needed
> (and overdue). Michelle and I will be relocating to the greater New
> York City area, but Michelle will continue in her role as Digium's
> in-house counsel. Because of our need to relocate, I'll only be at
> Digium until August 8th, although I'll be in Huntsville until around
> Labor Day.
> 
> This is yet another incredibly exciting, career changing opportunity
> in my life, and I can't wait to see what it will bring. I'll be
> forever thankful for the opportunity that Digium and the Asterisk
> community provided me to learn, grow and find the place where my skills and 
> experience are the most valuable (to both myself and my employer).
> 
Thanks very much Kevin, I have sincerely appreciated your insights and ability 
to help. I wish you great success in your next role

Best wishes

Duncan

> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>  http://www.asterisk.org/hello
> 
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> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users


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[asterisk-users] As Kevin Fleming says "So long, and thanks for all the fish!", we say thank you - and look to the future

2012-07-31 Thread David Duffett 



It's amazing what you can learn in a few days...

Having just found out that Queen Elizabeth has a great sense of humor, it has 
now emerged that Kevin Fleming - a man who (both with and without his 
moustache) has been an amazing contributor and influencer in the Asterisk 
project is set to move on to a new challenge outside the project - but still 
within the realms of Open Source.

Kevin has been involved with Asterisk for 7+ years, and has been both a thought 
leader and a powerful voice in the Asterisk world during that time. I first met 
Kevin at a TMC event called VoIP Developer in California (old school, well 
before the days of IT Expo), where he was speaking about Asterisk as well as 
helping to man the Digium booth at the event.

I've also followed Kevin around Berlin looking for great gelato during the 
AstriCon Europe 2006 tour - and it was well worth it, that man knows his gelato!

I'd like to take this opportunity to say thanks to Kevin for his enormous 
contribution to the Asterisk Project. Without his efforts, Asterisk would not 
be the success it is today ... Anyway, back to main theme - when someone in a 
senior role like Kevin moves on, it is important that others are there to pick 
up his responsibilities and move the project forward.

As it turns out, we've already been working on this, and have some very 
talented people that will be taking up the key responsibilities of the project 
going forward. Some of them have been involved with Asterisk for several years, 
and some are recent additions, but together they form a great team to lead 
Asterisk into the future.



Matt Jordan has assumed the project leader role for Asterisk, and is 
responsible for managing the releases of Asterisk, as well as all of the 
development efforts within Digium.
Mark Michelson is serving as the Technical Lead for the project, responsible 
for architecture and design direction.
We have also recently created the role of Community Support Manager, which 
Rusty Newton has filled. Rusty is a long time Digium employee with many years 
supporting Asterisk and Digium products, and will be the day to day interface 
for community technical issues.

As you know, I recently joined Digium to look after the interests of the 
worldwide Open Source Asterisk community and I will therefore also be working 
alongside the good people identified above, especially Rusty.

So while we wish Kevin all the best as he moves on, we are also confident that 
the good work he and the rest of the team have done continues to be in the best 
hands going forwards.

To the future...

David

Digium logo
David Duffett
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] Digium IP Phone D40 quality, very bad

2012-07-31 Thread A J Stiles 
On Tuesday 31 July 2012, bilal ghayyad wrote:
> Even at the web based configuration at the phone it self, I am not able to
> do reboot (there is no reboot button) and I can do this only from the
> Phone it self.

Real question:  Why are you needing to reboot your phones?

You should never be getting the phone into a state where the only way out is 
to reboot it.  The correct solution is *not* to make rebooting easier, but to 
avoid ever getting into that state in the first place.

By the way, we have recently acquired some D40s.  How do you change the time 
format from AM/PM to VCR-style?  (i.e. 16:41, not 4:41 pm).

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-31 Thread Carlos Alvarez 
Particularly, what virtualization software are you using?


On Tue, Jul 31, 2012 at 8:19 AM, Leandro Dardini  wrote:

> Hello Bryant,
> it is nice to hear someone with different experience, so I am happy to
> know the "cloud" is indeed a feasible environment even for VoIP.
>
> Can you share with us some of your configuration magic? Like the cloud
> service you are using, the power of each node and the load you are
> experiencing on them in regards to the number of channels active and phone
> registered?
>
> Leandro
>
> 2012/7/31 Bryant Zimmerman 
>
>> Kannan
>>
>> I have to disagree with Leanrod. We are a hosted (cloud) PBX company we
>> successfully run our Multi-tenant systems in Virtual machines and have no
>> issues with them. It comes down to designing your virtual environment for
>> your target loads and then not exceeding them. This allows for fail over of
>> hardware and scalability. We have moved our virtual phone switches live
>> with full call loads and have no call drops.   We do not usually dedicate a
>> single Virtual Machine to each customer either. We have built our own
>> Multi-tenant PBX on top of asterisk. We achieve many of the features
>> available in freepbx/trixbox (not all). This method allows us to cost
>> effectively service our customers with a presence of scale in mind. It is
>> not uncommon to have 5000 + extensions per virtual switch. This method does
>> require highly skilled engineering to achieve stability.
>>
>> Bryant
>>
>> --
>> *From*: "Kannan" 
>> *Sent*: Tuesday, July 31, 2012 12:37 AM
>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
>> asterisk-users@lists.digium.com>
>> *Subject*: Re: [asterisk-users] Multi-Tenant PBX with Asterisk
>>
>>
>> Thanks Leandro for your comments.
>>
>>
>> On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini wrote:
>>
>>>
>>>
>>> 2012/7/30 Kannan 
>>>
 Hi

  I came across couple of pointers on the Internet regarding solutions
 available for providing hosted PBX service.

  1. Multiple PBXs: Using separate hardware to host each PBX. Pretty
 straightforward, but no hosting company wants to use it.
 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance
 of Asterisk. I.e. partitioning a single instance of Asterisk into multiple
 PBXs by way of configurations, using unique landing context for each 
 tenant.
 3. Virtual PBX: Multiple virtual machines within the same hardware,
 each host an instance of Asterisk.

  Which one of the method above is generally used by hosted PBX service
 providers?

  Isn't the second option with ARA a good choice for dynamic creation
 of multiple "small" PBX tenants?

  Is the last option alone or combination of options 2 and 3 good for
 cloud based hosted PBX service offering?

  Thanks,
 Kannan.

>>>
>>>  Working in the voip field from a lots of years, I have found all three
>>> type of business.
>>>
>>>  The first is maybe the easier and most common. Hardware is cheap and
>>> it is easier to "sell" a service like the PBX if it is sold together with a
>>> piece of iron. Usually the hardware is placed on client's network, using
>>> the bandwidth of the client. Usually together with the PBX is sold also a
>>> router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic
>>> on the client's DSL.
>>>
>>>  The major pros about this solution is you can use a normal PBX like
>>> freepbx/trixbox,  the client can mess the config how he likes, without
>>> disrupting other services, you can install VoIP card to connect landlines,.
>>>
>>>  The major cons is the cost of the hardware, the cost of the g.729
>>> licenses (if any) and the maintenance cost of replacing hardware failures
>>> and the need to be physically near each client.
>>>
>>>  The second is the holy grail of the VoIP providers.
>>>
>>>  The major pros is the cost. Having a single hardware is cheap and it
>>> is still cheap also if you decide to get two to be ready in case of an
>>> hardware failure.
>>>
>>>  The major cons is the software. You cannot use the award winning
>>> freepbx/trixbox family and you need to deal with sometime limited or
>>> incomplete developed interfaces. The client always asks for the missing
>>> feature. One other major cons is the "reload". If the PBX software is not
>>> made using ARA, then every time you add a new peer or a new DID, you need
>>> to reload the entire PBX and that is a resource killer. Again, if the pbx
>>> interface is not made using ARA, then you cannot let your clients to change
>>> the configuration or they will trigger continuous reload (and delaying
>>> reload for example every 10 minutes is not a solution)
>>>
>>>  The last one is sometime the chosen compromise, but from my point of
>>> view, pbxes are not good software to virtualize. They are too sensible to
>>> delays and your voice quality can go down if the real server is overloa

Re: [asterisk-users] Asterisk 1.8.15.0 Now Available

2012-07-31 Thread Eric Germann 
Is there an ETA on when this will show up on packages?

Thanks for the work!

EKG 

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Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-31 Thread Leandro Dardini 
Hello Bryant,
it is nice to hear someone with different experience, so I am happy to know
the "cloud" is indeed a feasible environment even for VoIP.

Can you share with us some of your configuration magic? Like the cloud
service you are using, the power of each node and the load you are
experiencing on them in regards to the number of channels active and phone
registered?

Leandro

2012/7/31 Bryant Zimmerman 

> Kannan
>
> I have to disagree with Leanrod. We are a hosted (cloud) PBX company we
> successfully run our Multi-tenant systems in Virtual machines and have no
> issues with them. It comes down to designing your virtual environment for
> your target loads and then not exceeding them. This allows for fail over of
> hardware and scalability. We have moved our virtual phone switches live
> with full call loads and have no call drops.   We do not usually dedicate a
> single Virtual Machine to each customer either. We have built our own
> Multi-tenant PBX on top of asterisk. We achieve many of the features
> available in freepbx/trixbox (not all). This method allows us to cost
> effectively service our customers with a presence of scale in mind. It is
> not uncommon to have 5000 + extensions per virtual switch. This method does
> require highly skilled engineering to achieve stability.
>
> Bryant
>
> --
> *From*: "Kannan" 
> *Sent*: Tuesday, July 31, 2012 12:37 AM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Subject*: Re: [asterisk-users] Multi-Tenant PBX with Asterisk
>
>
> Thanks Leandro for your comments.
>
>
> On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini wrote:
>
>>
>>
>> 2012/7/30 Kannan 
>>
>>> Hi
>>>
>>>  I came across couple of pointers on the Internet regarding solutions
>>> available for providing hosted PBX service.
>>>
>>>  1. Multiple PBXs: Using separate hardware to host each PBX. Pretty
>>> straightforward, but no hosting company wants to use it.
>>> 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance
>>> of Asterisk. I.e. partitioning a single instance of Asterisk into multiple
>>> PBXs by way of configurations, using unique landing context for each tenant.
>>> 3. Virtual PBX: Multiple virtual machines within the same hardware, each
>>> host an instance of Asterisk.
>>>
>>>  Which one of the method above is generally used by hosted PBX service
>>> providers?
>>>
>>>  Isn't the second option with ARA a good choice for dynamic creation of
>>> multiple "small" PBX tenants?
>>>
>>>  Is the last option alone or combination of options 2 and 3 good for
>>> cloud based hosted PBX service offering?
>>>
>>>  Thanks,
>>> Kannan.
>>>
>>
>>  Working in the voip field from a lots of years, I have found all three
>> type of business.
>>
>>  The first is maybe the easier and most common. Hardware is cheap and it
>> is easier to "sell" a service like the PBX if it is sold together with a
>> piece of iron. Usually the hardware is placed on client's network, using
>> the bandwidth of the client. Usually together with the PBX is sold also a
>> router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic
>> on the client's DSL.
>>
>>  The major pros about this solution is you can use a normal PBX like
>> freepbx/trixbox,  the client can mess the config how he likes, without
>> disrupting other services, you can install VoIP card to connect landlines,.
>>
>>  The major cons is the cost of the hardware, the cost of the g.729
>> licenses (if any) and the maintenance cost of replacing hardware failures
>> and the need to be physically near each client.
>>
>>  The second is the holy grail of the VoIP providers.
>>
>>  The major pros is the cost. Having a single hardware is cheap and it is
>> still cheap also if you decide to get two to be ready in case of an
>> hardware failure.
>>
>>  The major cons is the software. You cannot use the award winning
>> freepbx/trixbox family and you need to deal with sometime limited or
>> incomplete developed interfaces. The client always asks for the missing
>> feature. One other major cons is the "reload". If the PBX software is not
>> made using ARA, then every time you add a new peer or a new DID, you need
>> to reload the entire PBX and that is a resource killer. Again, if the pbx
>> interface is not made using ARA, then you cannot let your clients to change
>> the configuration or they will trigger continuous reload (and delaying
>> reload for example every 10 minutes is not a solution)
>>
>>  The last one is sometime the chosen compromise, but from my point of
>> view, pbxes are not good software to virtualize. They are too sensible to
>> delays and your voice quality can go down if the real server is overloaded.
>>
>>  The same for the cloud based solutions (I have yet to found). I suspect
>> the "cloud" is good for services like http, not for real time applications.
>>
>>  Leandro
>>
>>
>> --
>> __

Re: [asterisk-users] Digium IP Phone D40 quality, very bad

2012-07-31 Thread Chris Bagnall 

However, there's no "reboot"
button in the web GUI of the phone.


I have no experience with the phone in question and so will make no 
comment regarding the OP's original problem, but the absence of a 
software reboot function in the web GUI seems to be a pretty major 
oversight in my view.


I do hope that one gets added to the "we should really add this to the 
firmware ASAP" list :-)


Kind regards,

Chris
--
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Re: [asterisk-users] Digium IP Phone D40 quality, very bad

2012-07-31 Thread Rusty Newton 

On 7/31/2012 7:22 AM, bilal ghayyad wrote:

Hi All;

Really it is miserable.

I bring 8 Digium Phone D40 and I used them with a customer, the voice quality 
is bad internally (between the extension), there is no clearance at all ! We 
are hearing the voice like another person.

The used codec is ulaw.

The firmware version is: 1_1_0_0_48178

Even at the web based configuration at the phone it self, I am not able to do 
reboot (there is no reboot button) and I can do this only from the Phone it 
self.

 From the speaker, the voice is very bad and weak.

I am really feel disappointed why I did not use Polycom.

Can someone help me or advise me what to do in this?



Many members of this list will testify that audio quality issues can 
happen on *any* SIP phone. Our support department regularly 
troubleshoots audio quality issues with many makes and models of phone 
(for those customers of our paid Asterisk Support). You can have a 
variety of audio quality problems caused from various root issues such 
as networking misconfiguration and even bad room acoustics or cheap 
headsets plugged into the phone.


If only it were as simple as "buy this phone it's good" or "don't buy 
this phone it's bad". Unfortunately as we all know, technology is often 
more complex.


In regards to rebooting the phone it sounds like you found the method 
within the phones physical interface. However, there's no "reboot" 
button in the web GUI of the phone.  There are two additional ways of 
doing it:


1) (I'm not using the DPMA), see - 
https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phones+when+used+without+the+DPMA

You'll send it a SIP NOTIFY telling it to reconfigure.
2) (I'm using the DPMA), see - 
https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+and+the+Asterisk+CLI
You'll do digium_phones reconfigure phone  ...or... 
digium_phones reconfigure all


Your best next step is to call Digium, the manufacturer of the phone; 
They offer *free technical support* on the Digium phones and can best 
help you troubleshoot the issue. For support call  +1 (256) 428-6000 , 
option 2 and then 2 again.


Thanks,

--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at:www.digium.com  www.asterisk.org


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Re: [asterisk-users] So long, and thanks for all the fish!

2012-07-31 Thread Thiago Coutinho 
On Tue, Jul 31, 2012 at 10:59 AM, Kevin P. Fleming  wrote:
> I've been with Digium for just over seven years, and it's been an
> incredible experience that I wouldn't have traded for anything. When
> Mark Spencer invited me to visit Digium (and Huntsville) in early
> 2005, I could not have dreamed that I'd end up working for such an
> exciting, innovative company, finding a wife, and meeting hundreds of
> people (many of whom are now friends) around the world. It's been a
> time of tremendous personal and career growth, and my wonderful
> colleagues at Digium and in the Asterisk open source community have
> been directly responsible for most of that.
>
> Recently, though, I've been presented an opportunity to take on a new
> challenge and this has resulted in my acceptance of a new job, in a
> new industry. In the middle of September, I'll start working for
> Bloomberg, L.P., in the Office of the CTO, helping to lead their
> nascent open source initiative. I'll be working to bring the power of
> open source software, open standards, and community building to the
> financial market data services industry, where it is sorely needed
> (and overdue). Michelle and I will be relocating to the greater New
> York City area, but Michelle will continue in her role as Digium's
> in-house counsel. Because of our need to relocate, I'll only be at
> Digium until August 8th, although I'll be in Huntsville until around
> Labor Day.
>
> This is yet another incredibly exciting, career changing opportunity
> in my life, and I can't wait to see what it will bring. I'll be
> forever thankful for the opportunity that Digium and the Asterisk
> community provided me to learn, grow and find the place where my skills and
> experience are the most valuable (to both myself and my employer).
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org

Good luck and thanks for you contributions to the Asterisk community!


-- 
thiagoc

"O povo não deveria temer o governo. O governo é quem deveria temer o povo."
V de Vingança

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[asterisk-users] So long, and thanks for all the fish!

2012-07-31 Thread Kevin P. Fleming 

I've been with Digium for just over seven years, and it's been an
incredible experience that I wouldn't have traded for anything. When
Mark Spencer invited me to visit Digium (and Huntsville) in early
2005, I could not have dreamed that I'd end up working for such an
exciting, innovative company, finding a wife, and meeting hundreds of
people (many of whom are now friends) around the world. It's been a
time of tremendous personal and career growth, and my wonderful
colleagues at Digium and in the Asterisk open source community have
been directly responsible for most of that.

Recently, though, I've been presented an opportunity to take on a new
challenge and this has resulted in my acceptance of a new job, in a
new industry. In the middle of September, I'll start working for
Bloomberg, L.P., in the Office of the CTO, helping to lead their
nascent open source initiative. I'll be working to bring the power of
open source software, open standards, and community building to the
financial market data services industry, where it is sorely needed
(and overdue). Michelle and I will be relocating to the greater New
York City area, but Michelle will continue in her role as Digium's
in-house counsel. Because of our need to relocate, I'll only be at
Digium until August 8th, although I'll be in Huntsville until around
Labor Day.

This is yet another incredibly exciting, career changing opportunity
in my life, and I can't wait to see what it will bring. I'll be
forever thankful for the opportunity that Digium and the Asterisk
community provided me to learn, grow and find the place where my skills 
and experience are the most valuable (to both myself and my employer).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Digium IP Phone D40 quality, very bad

2012-07-31 Thread Paul Belanger 

On 12-07-31 08:22 AM, bilal ghayyad wrote:

Hi All;

Really it is miserable.

I bring 8 Digium Phone D40 and I used them with a customer, the voice quality 
is bad internally (between the extension), there is no clearance at all ! We 
are hearing the voice like another person.

The used codec is ulaw.

The firmware version is: 1_1_0_0_48178

Even at the web based configuration at the phone it self, I am not able to do 
reboot (there is no reboot button) and I can do this only from the Phone it 
self.

 From the speaker, the voice is very bad and weak.

I am really feel disappointed why I did not use Polycom.

Can someone help me or advise me what to do in this?

1. I'm curious why you went with the Digium D40 vs Polycom?  I assume 
from your post you have already successfully implemented Polycom phones 
on site.


2. Did you not test the phones before going on site?  If yes, what has 
changed between the two sites.  I get the impression when you say 
'Really it is miserable' this is the first time you have set them up and 
are truly surprised with the quality.


3. There are many things that can affect audio quality.  What have you 
done to isolate the specific issue to the Digium D40?


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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Re: [asterisk-users] libpri error

2012-07-31 Thread Tzafrir Cohen 
On Mon, Jul 30, 2012 at 10:50:20AM +, Kamlesh Kumar wrote:
> 
> when I issue 'make' command, below output comes. [root@localhost 
> libpri-1.4.11.3]# make
> gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -O2 -MD 
> -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c

[snip]

Looks OK.

Make is quite noisy when it encounters an error:

$ echo "something invalid" >>q921.c 

$ LANG=C make
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC  -O2 -MD 
-MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC  -O2 -MD 
-MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC  -O2 -MD 
-MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c
q921.c:3113:1: error: unknown type name 'something'
q921.c:3113:1: error: expected '=', ',', ';', 'asm' or '__attribute__' at end 
of input
make: *** [q921.o] Error 1

$ echo $?
2


Had I not sabotaged the code:

$ make
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC  -O2 -MD 
-MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC  -O2 -MD 
-MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c
[snip]
gcc -shared -Wl,-hlibpri.so.1.4  -o libpri.so.1.4 copy_string.lo pri.lo q921.lo 
prisched.lo q931.lo pri_aoc.lo pri_cc.lo pri_facility.lo asn1_primitive.lo 
rose.lo rose_address.lo rose_etsi_aoc.lo rose_etsi_cc.lo rose_etsi_diversion.lo 
rose_etsi_ect.lo rose_etsi_mwi.lo rose_other.lo rose_q931.lo rose_qsig_aoc.lo 
rose_qsig_cc.lo rose_qsig_ct.lo rose_qsig_diversion.lo rose_qsig_mwi.lo 
rose_qsig_name.lo version.lo
/sbin/ldconfig -n .
ln -sf libpri.so.1.4 libpri.so

$ echo $?
0


'$?' (the value of the shell variable named '?') holds the return status
of the last command. If it's not 0, it means that this command returned
an error.


-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-31 Thread Bryant Zimmerman 
Kannan

I have to disagree with Leanrod. We are a hosted (cloud) PBX company we 
successfully run our Multi-tenant systems in Virtual machines and have no 
issues with them. It comes down to designing your virtual environment for 
your target loads and then not exceeding them. This allows for fail over of 
hardware and scalability. We have moved our virtual phone switches live 
with full call loads and have no call drops.   We do not usually dedicate a 
single Virtual Machine to each customer either. We have built our own 
Multi-tenant PBX on top of asterisk. We achieve many of the features 
available in freepbx/trixbox (not all). This method allows us to cost 
effectively service our customers with a presence of scale in mind. It is 
not uncommon to have 5000 + extensions per virtual switch. This method does 
require highly skilled engineering to achieve stability. 

Bryant 


 From: "Kannan" 
Sent: Tuesday, July 31, 2012 12:37 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Multi-Tenant PBX with Asterisk

Thanks Leandro for your comments.  

On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini  
wrote:

2012/7/30 Kannan 
Hi 
 I came across couple of pointers on the Internet regarding solutions 
available for providing hosted PBX service. 
 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty 
straightforward, but no hosting company wants to use it. 2. Multi-tenant 
PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. 
partitioning a single instance of Asterisk into multiple PBXs by way of 
configurations, using unique landing context for each tenant. 3. Virtual 
PBX: Multiple virtual machines within the same hardware, each host an 
instance of Asterisk. 
 Which one of the method above is generally used by hosted PBX service 
providers? 
 Isn't the second option with ARA a good choice for dynamic creation of 
multiple "small" PBX tenants? 
 Is the last option alone or combination of options 2 and 3 good for cloud 
based hosted PBX service offering? 
 Thanks, Kannan.  
  Working in the voip field from a lots of years, I have found all three 
type of business. 
 The first is maybe the easier and most common. Hardware is cheap and it is 
easier to "sell" a service like the PBX if it is sold together with a piece 
of iron. Usually the hardware is placed on client's network, using the 
bandwidth of the client. Usually together with the PBX is sold also a 
router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic 
on the client's DSL. 
 The major pros about this solution is you can use a normal PBX like 
freepbx/trixbox,  the client can mess the config how he likes, without 
disrupting other services, you can install VoIP card to connect landlines,. 

 The major cons is the cost of the hardware, the cost of the g.729 licenses 
(if any) and the maintenance cost of replacing hardware failures and the 
need to be physically near each client. 
 The second is the holy grail of the VoIP providers.  
 The major pros is the cost. Having a single hardware is cheap and it is 
still cheap also if you decide to get two to be ready in case of an 
hardware failure.  
 The major cons is the software. You cannot use the award winning 
freepbx/trixbox family and you need to deal with sometime limited or 
incomplete developed interfaces. The client always asks for the missing 
feature. One other major cons is the "reload". If the PBX software is not 
made using ARA, then every time you add a new peer or a new DID, you need 
to reload the entire PBX and that is a resource killer. Again, if the pbx 
interface is not made using ARA, then you cannot let your clients to change 
the configuration or they will trigger continuous reload (and delaying 
reload for example every 10 minutes is not a solution) 
 The last one is sometime the chosen compromise, but from my point of view, 
pbxes are not good software to virtualize. They are too sensible to delays 
and your voice quality can go down if the real server is overloaded. 
 The same for the cloud based solutions (I have yet to found). I suspect 
the "cloud" is good for services like http, not for real time applications. 
 
 Leandro 

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[asterisk-users] Digium IP Phone D40 quality, very bad

2012-07-31 Thread bilal ghayyad 
Hi All;

Really it is miserable.

I bring 8 Digium Phone D40 and I used them with a customer, the voice quality 
is bad internally (between the extension), there is no clearance at all ! We 
are hearing the voice like another person.

The used codec is ulaw.

The firmware version is: 1_1_0_0_48178

Even at the web based configuration at the phone it self, I am not able to do 
reboot (there is no reboot button) and I can do this only from the Phone it 
self.

>From the speaker, the voice is very bad and weak.

I am really feel disappointed why I did not use Polycom.

Can someone help me or advise me what to do in this?
Regards
Bilal

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[asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

2012-07-31 Thread Sebastian Arcus 
I have two setups with SIP hardware phones as extensions and POTS lines 
as trunks. Internal SIP to SIP calls are crystal clear, but all calls 
bridged to POTS have a significant amount of static noise. The problem 
is that if I plug a POTS phone directly into the line, there is almost 
no static noise - the line is clean. It's like Asterisk (or the 
hardware) amplifies the static noise. What I've tried so far:


1. Connect Asterisk with a short cable directly into the master phone 
socket, where it enters the building.

2. One of the lines carries ADSL - so I double filtered it.
3. Tried three different phone sets (one Grandstream, two Cisco models).
4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB adapter as 
analogue-to-digital interfaces.
5. Reduced the software echo canceller in chan_dahdi.conf to 32 and even 
16 - until I could actually start to hear echo. Still no difference.
6. Reduced the rxgain and txgain in chan_dahdi.conf to 0 - but the 
static noise is still there.

7. Tried different phone cables for the pots line.
8. Tried a different motherboard on the computer with Asterisk and 
checked there is no IRQ sharing. Tried when there was no other load on 
the Asterisk computer.

9. Tried Asterisk 1.6, 1.8 and 10

Is there anything else I can do - or should I just give in to the static 
noise? Is that how other hybrid setups work - do you get static noise on 
the line - more than if plugged directly? The client is adamant that the 
noise on the line is too high - by comparison with the quality on mobile 
phone calls (which are digital, incidentally) - so if I don't find a 
solution, I suppose I will just have to rip it all out and let one of 
the companies with proprietary phone systems install one.


Any hints appreciated.

Sebastian

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[asterisk-users] AGI not generating sip 180/183 status

2012-07-31 Thread Marek Cervenka 

hello,

i have strange problem with AGI (asterisk 1.8.10.0)
when i use Dial from dialplan everything is ok
when i dial from AGI script there is missing SIP Status 180 ringing and 
183 session progress


any ideas?

DIAL without AGI

196.356479 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE 
sip:222333...@some.pbx.org, with session description

196.356768 10.0.0.213 -> 10.0.0.193 SIP Status: 401 Unauthorized
196.365709 10.0.0.193 -> 10.0.0.213 SIP Request: ACK 
sip:222333...@some.pbx.org
196.370028 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE 
sip:222333...@some.pbx.org, with session description

196.370503 10.0.0.213 -> 10.0.0.193 SIP Status: 100 Trying
199.797325 10.0.0.213 -> 10.0.0.193 SIP Status: 180 Ringing
199.797932 10.0.0.213 -> 10.0.0.193 SIP/SDP Status: 183 Session 
Progress, with session description
199.878441 10.0.0.193 -> 10.0.0.213 RTCP Receiver Report   Source 
description
199.988259 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, 
SSRC=0xD2C6DEB8, Seq=7289, Time=3171500, Mark
200.004139 10.0.0.213 -> 10.0.0.193 RTP PT=ITU-T G.711 PCMA, 
SSRC=0x279E385A, Seq=50775, Time=28960
200.008118 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, 
SSRC=0xD2C6DEB8, Seq=7290, Time=3171660


201.504218 10.0.0.213 -> 10.0.0.193 RTP PT=ITU-T G.711 PCMA, 
SSRC=0x279E385A, Seq=50850, Time=40960
201.519477 10.0.0.193 -> 10.0.0.213 SIP Request: BYE 
sip:222333444@10.0.0.213:5060

201.519611 10.0.0.213 -> 10.0.0.193 SIP Status: 487 Request Terminated
201.519800 10.0.0.213 -> 10.0.0.193 SIP Status: 200 OK
201.528465 10.0.0.193 -> 10.0.0.213 SIP Request: ACK 
sip:222333...@some.pbx.org



DIAL from AGI
66.581752 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE 
sip:222333...@some.pbx.org, with session description

66.581958 10.0.0.213 -> 10.0.0.193 SIP Status: 401 Unauthorized
66.590738 10.0.0.193 -> 10.0.0.213 SIP Request: ACK 
sip:222333...@some.pbx.org
66.59 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE 
sip:222333...@some.pbx.org, with session description

66.596167 10.0.0.213 -> 10.0.0.193 SIP Status: 100 Trying
66.652571 10.0.0.213 -> 10.0.0.193 SIP/SDP Status: 200 OK, with session 
description

66.676485 10.0.0.193 -> 10.0.0.213 RTCP Receiver Report   Source description
66.750371 10.0.0.193 -> 10.0.0.213 SIP Request: ACK 
sip:222333444@10.0.0.213:5060
66.844392 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, 
SSRC=0xE842E26F, Seq=3869, Time=1120100, Mark
66.854430 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, 
SSRC=0xE842E26F, Seq=3870, Time=1120260

...
69.404625 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, 
SSRC=0xE842E26F, Seq=3998, Time=1140740
69.516390 10.0.0.193 -> 10.0.0.213 SIP Request: BYE 
sip:222333444@10.0.0.213:5060

69.516669 10.0.0.213 -> 10.0.0.193 SIP Status: 200 OK

--
---
Marek Cervenka
===


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Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)

2012-07-31 Thread Ishfaq Malik 
On Mon, 2012-07-30 at 08:39 -0500, Matthew Jordan wrote:
> 
> - Original Message -
> > From: "Ishfaq Malik" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> > 
> > Sent: Wednesday, July 18, 2012 9:58:47 AM
> > Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 
> > 1.8)
> > 
> > On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote:
> > > Hi
> > > 
> > > I'm having a problem with the entirety of a call being recorded in
> > > the
> > > following scenario
> > > I'm using asterisk 1.8.7.0
> > > 
> > > Person A (asterisk peer) calls Person B (not on asterisk, real
> > > world
> > > number via a SIP trunk)
> > > Mixmonitor is invoked by Person A in the outbound context and
> > > AUDIOHOOK_INHERIT(MixMonitor)=yes is also set
> > > Person a transfers Person B to Person C (another asterisk peer)
> > > Person A is no longer involved in the call and the call is bridged
> > > between Person B and Person C
> > > 
> > > The call recording stops as soon as Person A hangs up, even though
> > > AUDIOHOOK_INHERIT is set
> > > 
> > > Is there any way we can get the entire call recorded in one file?
> > > 
> > > Thanks in advance
> > > 
> > > Ish
> 
> Ish:
> 
> Leif had a pretty good explanation of why AUDIOHOOK_INHERIT behaves this
> way in the comments of ASTERISK-16013.  I'll quote it here:
> 
> "Well I just tested this scenario. ... after a bit of testing I determined the
> scenarios.
> 
> Working:
> 
> * Party A places a call to Party B
> * Party B places an attended transfer to Party C
> * Party A and C are not talking
> * Call recording works as expected
> 
> Not working:
> 
> * Party A places a call to Party B
> * Party A places an attended transfer to Party C
> Call recording works up to this point – the recording of the conversation
> between Party A and Party B, and the portion of the conversation between 
> Party A
> and Party C is recorded
> * Party A now hangs up
> * Call recording is now stopped
> * Party B and Party C are now speaking (unrecorded)
> 
> To me, this is actually the intended and expected behavior. The 
> AUDIOHOOK_INHERIT() function is executed on the channel created by Party A, 
> and 
> thus the call recording is going to follow Party A around when it is 
> transferred
> around the system.
> 
> However, once Party A is kicked out of the conversation (i.e. they hangup) 
> then
> the call recording stops because that is the channel the recording is 
> associated
> with."
> 
> Note that if you read the scenario description of AUDIOHOOK_INHERIT at
> https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT, the
> function works in the transfer scenarios where the called party initiates the
> transfer, not the callee.
> 
> For your scenario, you could try setting the MixMonitor on the called party
> channel as opposed to the callee channel, using one of the Dial GoSub/Macro
> options (U,M,b).
> 
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
> 
> Note that Macro is deprecated in more recent versions of Asterisk, and the 'b'
> option will only be available in Asterisk 11.
> 

Thank you for the hints at the end, using the M option has sorted my
issue out

Ish

-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-31 Thread Ishfaq Malik 
On Mon, 2012-07-30 at 15:06 +0530, Kannan wrote:
> Hi
> 
> 
> I came across couple of pointers on the Internet regarding solutions
> available for providing hosted PBX service.
> 
> 
> 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty
> straightforward, but no hosting company wants to use it.
> 2. Multi-tenant PBX: Configuring multiple PBXs within the same
> instance of Asterisk. I.e. partitioning a single instance of Asterisk
> into multiple PBXs by way of configurations, using unique landing
> context for each tenant.
> 3. Virtual PBX: Multiple virtual machines within the same hardware,
> each host an instance of Asterisk.
> 
> 
> Which one of the method above is generally used by hosted PBX service
> providers?
> 
> 
> Isn't the second option with ARA a good choice for dynamic creation of
> multiple "small" PBX tenants?
> 
> 
> Is the last option alone or combination of options 2 and 3 good for
> cloud based hosted PBX service offering?
> 
We use 2 and I'd have to agree with most of what the previous replies
have said. You really need to nail down your conventions and stick to
them. We did this by creating our own custom front end so our
conventions are built in to the front end code.

ARA is really useful for this type of thing. If you're expanding to the
point that you need to add new servers for extra capacity, ARA enables
you to retain all your config on a single (pair of) machine(s). It also
means that, if you have the framework to allow it, your customers can
make changes to their own account themselves.


-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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