[VoiceOps] FW: Linksys PAP2T (Scott Berkman)

Wed Feb 1 12:49:32 EST 2012

I have had the same problem with the Cisco SPA ATAs (Linksys).  I was able to get the SPA ATA to honor the ACME SBC settings, but as Scott says below, the registration values can't be more than 50% apart. 

On a related note...  I also found was that the SPA2102 and SPA8000 ATA software re-registers "3" sec before the timer expires; for example, if the registration timer set by ACME on the ATA is set to 205 seconds, the ATA will attempt to re-register every 202 seconds.  This is much different from the Cisco SPA phones, which register at 67% of the set reg timer value.  The "3" sec re-registration timer is NOT a configurable value as it is set in the software.  So, you will need to adjust your desired registration timer on the SBC for the ATAs to match what your Firewall UDP timers are expecting (if that is an issue for your CPE setup).  Otherwise, your SIP ports will be shut-down pre-maturely.   This was a problem for us.

-----Original Message-----
From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of voiceops-request at voiceops.org
Sent: Wednesday, February 01, 2012 8:51 AM
To: voiceops at voiceops.org
Subject: VoiceOps Digest, Vol 32, Issue 1

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Today's Topics:

   1. Linksys PAP2T (Mark Holloway)
   2. Re: Linksys PAP2T (Scott Berkman)
   3. Experiences with VoIP and 100+ seat sites (Darren Schreiber)
   4. Re: Experiences with VoIP and 100+ seat sites (Carlos Alvarez)
   5. Re: Experiences with VoIP and 100+ seat sites (Zak Rupas)
   6. Re: Experiences with VoIP and 100+ seat sites (Carlos Alvarez)

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Message: 1
Date: Tue, 31 Jan 2012 16:05:30 -0700
From: Mark Holloway <mh at markholloway.com>
To: VoiceOps <voiceops at voiceops.org>
Subject: [VoiceOps] Linksys PAP2T
Message-ID: <77FD9C82-4778-4889-A573-7810BF41B43A at markholloway.com>
Content-Type: text/plain; charset=us-ascii

Has anyone encountered an issue where the PAP2T isn't honoring the registration timer in the contact header sent by an Acme Packet SD to the PAP2T?  Any suggestions are appreciated.

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Message: 2
Date: Tue, 31 Jan 2012 20:31:17 -0500
From: "Scott Berkman" <scott at sberkman.net>
To: "'Mark Holloway'" <mh at markholloway.com>, "'VoiceOps'"
	<voiceops at voiceops.org>
Subject: Re: [VoiceOps] Linksys PAP2T
Message-ID: <033301cce081$33f1f640$9bd5e2c0$@sberkman.net>
Content-Type: text/plain;	charset="us-ascii"

What is the Acme trying to set the timer to, and how does that compare to the default registration interval in the Linksys?  I think they can't be more than 50% apart (so if the Linksys is set to 3600, and the Acme suggests 30, that's too much of a difference and you'd need to lower the interval on the Linksys).  There are also settings for min and max registration interval on the SIP settings page.

-----Original Message-----
From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org]
On Behalf Of Mark Holloway
Sent: Tuesday, January 31, 2012 6:06 PM
To: VoiceOps
Subject: [VoiceOps] Linksys PAP2T

Has anyone encountered an issue where the PAP2T isn't honoring the registration timer in the contact header sent by an Acme Packet SD to the PAP2T?  Any suggestions are appreciated.
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Message: 3
Date: Wed, 1 Feb 2012 08:25:27 -0800
From: Darren Schreiber <d at d-man.org>
To: "VoiceOps at voiceops.org" <VoiceOps at voiceops.org>
Subject: [VoiceOps] Experiences with VoIP and 100+ seat sites
Message-ID: <CB4EA777.482AB%d at d-man.org>
Content-Type: text/plain; charset="us-ascii"

Hi folks,
I'd love to hear some stories (good or bad) of hosted PBX VoIP installs on 100+ seat sites (single site). Specifically if you've done this with Broadsoft or another solidified switch. I have mixed opinions on how this type of scenario can be successful and now I'm being pressed by a client on a formal opinion. I figure having it based on experience from others on a similar product is worth hearing about.

Specifically curious about how you addressed call quality issues and ensured bandwidth and uplink were sufficient.

As much detail as you're willing would be great, on or off list.

Thanks,
Darren

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Message: 4
Date: Wed, 1 Feb 2012 09:41:20 -0700
From: Carlos Alvarez <carlos at televolve.com>
To: voiceops at voiceops.org
Subject: Re: [VoiceOps] Experiences with VoIP and 100+ seat sites
Message-ID:
	<CAFn1dUFB=ZgE-=ZzPHtYptyUPv_4MGGMNwsLHiCA=Kueu0RMoA at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Since I only use Asterisk, would my experience be useful with 100-seat sites?

On Wed, Feb 1, 2012 at 9:25 AM, Darren Schreiber <d at d-man.org> wrote:

> Hi folks,
> I'd love to hear some stories (good or bad) of hosted PBX VoIP 
> installs on
> 100+ seat sites (single site). Specifically if you've done this with
> Broadsoft or another solidified switch. I have mixed opinions on how 
> this type of scenario can be successful and now I'm being pressed by a 
> client on a formal opinion. I figure having it based on experience 
> from others on a similar product is worth hearing about.
>
> Specifically curious about how you addressed call quality issues and 
> ensured bandwidth and uplink were sufficient.
>
> As much detail as you're willing would be great, on or off list.
>
> Thanks,
> Darren
>
>
> _______________________________________________
> VoiceOps mailing list
> VoiceOps at voiceops.org
> https://puck.nether.net/mailman/listinfo/voiceops
>
>

--
Carlos Alvarez
TelEvolve
602-889-3003
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Message: 5
Date: Wed, 1 Feb 2012 09:49:21 -0700
From: Zak Rupas <zak at simplesignal.com>
To: Carlos Alvarez <carlos at televolve.com>, voiceops at voiceops.org
Subject: Re: [VoiceOps] Experiences with VoIP and 100+ seat sites
Message-ID: <9f608f078e619af3a0749676f7006115 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

*I have done and my company has done 100 seat installs using Hosted phones with Broadsoft. Let me know what type of questions you have*

* *

*Thanks-*

Zak

*From:* voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org]
*On Behalf Of *Carlos Alvarez
*Sent:* Wednesday, February 01, 2012 9:41 AM
*To:* voiceops at voiceops.org
*Subject:* Re: [VoiceOps] Experiences with VoIP and 100+ seat sites

Since I only use Asterisk, would my experience be useful with 100-seat sites?

On Wed, Feb 1, 2012 at 9:25 AM, Darren Schreiber <d at d-man.org> wrote:

Hi folks,

I'd love to hear some stories (good or bad) of hosted PBX VoIP installs on
100+ seat sites (single site). Specifically if you've done this with
Broadsoft or another solidified switch. I have mixed opinions on how this type of scenario can be successful and now I'm being pressed by a client on a formal opinion. I figure having it based on experience from others on a similar product is worth hearing about.

Specifically curious about how you addressed call quality issues and ensured bandwidth and uplink were sufficient.

As much detail as you're willing would be great, on or off list.

Thanks,

Darren

_______________________________________________
VoiceOps mailing list
VoiceOps at voiceops.org
https://puck.nether.net/mailman/listinfo/voiceops

-- 

Carlos Alvarez

TelEvolve

602-889-3003
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Message: 6
Date: Wed, 1 Feb 2012 09:51:46 -0700
From: Carlos Alvarez <carlos at televolve.com>
To: "VoiceOps at voiceops.org" <VoiceOps at voiceops.org>
Subject: Re: [VoiceOps] Experiences with VoIP and 100+ seat sites
Message-ID:
	<CAFn1dUHCTwifUJf_faiAR2pDynKkkqq5XLVLx3hPetkvTCCuyQ at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

On Wed, Feb 1, 2012 at 9:25 AM, Darren Schreiber <d at d-man.org> wrote:

> Hi folks,
> I'd love to hear some stories (good or bad) of hosted PBX VoIP 
> installs on
> 100+ seat sites (single site). Specifically if you've done this with
> Broadsoft or another solidified switch. I have mixed opinions on how 
> this type of scenario can be successful and now I'm being pressed by a 
> client on a formal opinion. I figure having it based on experience 
> from others on a similar product is worth hearing about.
>

I've done a couple in this range.  I don't think 100 is a lot, and I don't think it's much of a challenge.  The "things to do" are pretty straightforward and there are lots of sources for best practices.  For example, if you do separate cabling and switches for the phones, then you can simply ignore LAN QoS (CoS) and you'll have a nice separate network for troubleshooting purposes.  This is what I do.

> Specifically curious about how you addressed call quality issues and 
> ensured bandwidth and uplink were sufficient.
>

You need a circuit that either has QoS from the ISP, is direct to the VoIP carrier, or a separate circuit with guaranteed bandwidth to the carrier(s) if you want a guarantee.  That said, I've done 70+ concurrent calls over wild internet without major issues by simply selecting an ISP with excellent upstream connectivity.  I don't know your level of understanding of internet routing, so it's hard to know where to go with those details.

When we deploy to our larger local customers (which to us is 25 or more), we use a local ISP who has a city-wide WiMAX network.  They deliver a VLAN from the customer site directly to our presence in the same facilities they are in.  This takes the guessing out of it.  You can do the same with most any carrier with the right engineering.  The first question would be who is the SIP provider?

This isn't black magic.  100 phones really is pretty simple and the ability to give them very high levels of service is well set.

--
Carlos Alvarez
TelEvolve
602-889-3003
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