Re: [asterisk-users] Adit 600 password reset

[Darren Wright]

Are you trying ethernet or serial?  Have you tried the other?
 
 
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of C F
Sent: Thu 5/22/2008 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adit 600 password reset



In the manual it's mentioned that if the DIP switch marked RST is set
then it will reset CLI password.
I have not been successful in doing that. Has anyone tried it?
I bought one off eBay and can't get in because of username password
that I don't know. I am assuming local is set to off.

Thank you

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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Darren Wright]

I've used Adit600's almost exclusively for my installs.   All have worked great 
for me.
 
-D
 



From: [EMAIL PROTECTED] on behalf of Steve Totaro
Sent: Thu 4/3/2008 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks.



Just Google Quintum Tenor AX.  Well worth the money.

Thanks,
Steve Totaro

On Mon, Mar 31, 2008 at 10:03 PM, Al lists [EMAIL PROTECTED] wrote:
 Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
  Steve, what are my options for SIP to fxs?
  thank you!



  On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote:
   Don Pobanz wrote:
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
   
   
   
This does not sound right. If it is 2 PRIs then it should be 46 channels
   
   
  
   I may have the terminology incorrect. I don't have a D channel, so I
   guess this would be called a T1 then?
  
   Doug
  
  
   --
   Ben Franklin quote:
  
   Those who would give up Essential Liberty to purchase a little Temporary
   Safety, deserve neither Liberty nor Safety.
  
  
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Re: [asterisk-users] CentPBX mirror?

[Darren Wright]

CentPBX has bit the dust I believe.
 
-D



From: [EMAIL PROTECTED] on behalf of Chris Bagnall
Sent: Wed 4/2/2008 7:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CentPBX mirror?



Greetings list,

Not exclusively asterisk-related, but I've noticed the CentPBX site has been 
offline the last few days. Anyone know the reasoning behind that, and more 
importantly, is anyone mirroring it?

Thanks in advance.

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it http://www.minotaur.it/ 
This email is made from 100% recycled electrons





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Re: [asterisk-users] Had it with Dell Garbage - HP Question

[Darren Wright]

Yup.Trixbox.
 
-D
 



From: [EMAIL PROTECTED] on behalf of Al Baker
Sent: Sat 3/29/2008 2:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question



How did you chose Centos, versus Red Hat, Suse, Debian, ?
Was there some key feature it offered that the others didn't ?
Cost ?

Darren Wright wrote:
 Notifications can be done either thru SNMP traps or SMTP.  Insight
 Manager is free from HP, but any SNMP trapper can work with alerts.

 The recovery CD is just a build that reloads the majority of the system
 with a static ip.   We backup off site to one of our servers via FTP.

 ILO access is an integrated IP KVM.   So you can see the machine boot,
 get virtual media access, etc.

 O/S is CentOS.

 For smaller systems, RAID 1, and for larger DL380 based systems 0+1

 -D


  
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Thursday, March 27, 2008 8:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question

 How do you get notifications ?
 Is this thru one of the add on packages HP sells for the box ?  Which

 One
  
 ?
 Could you be more specific about what you mean by a recovery CD
 and hod do you get console access below multi used to do recovery ??

 What is integrated ILO BIOS Access sounds cool.

 What O/S you usin and what made you pick it ?

 What kind and how many RAIDS are you using. The HP site gave like 8
 different RAID controllers and like 20 CPUs to chose from.  How did

 you
  
 chose ?

 Thx for sharing !!!

 Darren Wright wrote:

 One of the major reasons we use DL320 / DL380's is the ease of
  
 swapping
  
 drives, and the integrated ILO BIOS level access.We can support

 remote
  
 sites with ease.

 If a drive dies we get a notification, a new one is sent and a non-
  
 techie can replace it with guidance.No onsite visit.   That is

 worth
  
 potentially thousands of dollars.

 We also leave a recovery CD there that can be inserted if we need to
  
 rebuild the system remotely.   Never had to, but it's worked in the

 lab.
  
 -D

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Re: [asterisk-users] Had it with Dell Garbage - HP Question

[Darren Wright]

Notifications can be done either thru SNMP traps or SMTP.  Insight
Manager is free from HP, but any SNMP trapper can work with alerts.

The recovery CD is just a build that reloads the majority of the system
with a static ip.   We backup off site to one of our servers via FTP.

ILO access is an integrated IP KVM.   So you can see the machine boot,
get virtual media access, etc.

O/S is CentOS.

For smaller systems, RAID 1, and for larger DL380 based systems 0+1

-D


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Thursday, March 27, 2008 8:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question
 
 How do you get notifications ?
 Is this thru one of the add on packages HP sells for the box ?  Which
One
 ?
 Could you be more specific about what you mean by a recovery CD
 and hod do you get console access below multi used to do recovery ??
 
 What is integrated ILO BIOS Access sounds cool.
 
 What O/S you usin and what made you pick it ?
 
 What kind and how many RAIDS are you using. The HP site gave like 8
 different RAID controllers and like 20 CPUs to chose from.  How did
you
 chose ?
 
 Thx for sharing !!!
 
 Darren Wright wrote:
  One of the major reasons we use DL320 / DL380's is the ease of
swapping
 drives, and the integrated ILO BIOS level access.We can support
remote
 sites with ease.
 
  If a drive dies we get a notification, a new one is sent and a non-
 techie can replace it with guidance.No onsite visit.   That is
worth
 potentially thousands of dollars.
 
  We also leave a recovery CD there that can be inserted if we need to
 rebuild the system remotely.   Never had to, but it's worked in the
lab.
 
  -D
 
  This message was sent from D2 Technology, INC.
 
 
 

 
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Re: [asterisk-users] Had it with Dell Garbage - HP Question

[Darren Wright]

One of the major reasons we use DL320 / DL380's is the ease of swapping drives, 
and the integrated ILO BIOS level access.We can support remote sites with 
ease.   
 
If a drive dies we get a notification, a new one is sent and a non-techie can 
replace it with guidance.No onsite visit.   That is worth potentially 
thousands of dollars. 
 
We also leave a recovery CD there that can be inserted if we need to rebuild 
the system remotely.   Never had to, but it's worked in the lab.
 
-D

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Re: [asterisk-users] DID T1 PRI

[Darren Wright]

That's not going to tell you anything about the digits in transit.   That's 
just telling you that your PRI is up.
 
you are going to need exten = 4DIGITS
 
 



From: [EMAIL PROTECTED] on behalf of broadband Voice
Sent: Sat 3/15/2008 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DID T1 PRI


Additional output 
 

[EMAIL PROTECTED] ~]# /sbin/ztcfg -vv

Zaptel Version: 1.4.9
Echo Canceller: MG2
Configuration
==

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: Clear channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: D-channel (Default) (Slaves: 24)

24 channels to configure.

 



 
On 3/15/08, broadband Voice [EMAIL PROTECTED] wrote: 

Can you share with me sample extensions.conf? This is what I have
 
exten = 215xxx,1,Dial(Zap/1)
 
in zapata.conf


[channels] 
context=external 
switchtype=ni1 
resetinterval=3600 
overlapdial=no 
priindication=outofband 
facilityenable=yes 
signalling=pri_cpe 
usecallerid=yes 
cidsignalling=bell 
hidecallerid=no 
restrictcid=no 
usecallingpres=yes 
echocancel=yes 
callerid=asreceived 
faxdetect=incoming 
nsf=sdn 
group=1 
channel=1-23 
 
zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24 
 

 
 


 
On 3/15/08, Darren Wright [EMAIL PROTECTED] wrote: 

Feel free to ping me off list.  I've setup quite a few Cavtel 
PRI's with *.the paperwork they asked you to setup?

Typically, they only send 4 digits.

Do you have the questionnare they asked you to fill out?

dwright at d2 - tech dot com.



From: [EMAIL PROTECTED] on behalf of broadband Voice
Sent: Fri 3/14/2008 9:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DID T1 PRI


Thanks. I am in Philly. I may have to configure the 
extensions.conf well to pass the incoming channels.


On 3/14/08, Steve Totaro [EMAIL PROTECTED] wrote:

   On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice
   [EMAIL PROTECTED] wrote:
I had Cavalier turn up a T1 PRI. How can I put in the 
DIDs to direct to
Asterisk. Here is a log
   
   
   
Zaptel Tool (C)2002 Linux Support Services, Inc.
 ⤠T2XXP (PCI) Card 0 Span 1
ââ[3;10Hâterfaces â 
â[3;37Hâ
â  â
   
 â 
   â
â
 âCurrent Alarms: No alarms.   
   â rd 0
Span 1   â  â
  âSync Source:T2XXP (PCI) Card 0 
Span 1   â rd
0 Span 2   â(R)  â
 âIRQ Misses:   0  
   â
â  â
  âBipolar Viol: 0 
â
â  â

Re: [asterisk-users] DID T1 PRI

[Darren Wright]

Feel free to ping me off list.  I've setup quite a few Cavtel PRI's with *.the 
paperwork they asked you to setup?
 
Typically, they only send 4 digits.
 
Do you have the questionnare they asked you to fill out?
 
dwright at d2 - tech dot com.



From: [EMAIL PROTECTED] on behalf of broadband Voice
Sent: Fri 3/14/2008 9:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DID T1 PRI


Thanks. I am in Philly. I may have to configure the extensions.conf well to 
pass the incoming channels. 


On 3/14/08, Steve Totaro [EMAIL PROTECTED] wrote: 

On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice
[EMAIL PROTECTED] wrote:
 I had Cavalier turn up a T1 PRI. How can I put in the DIDs to direct 
to
 Asterisk. Here is a log



 Zaptel Tool (C)2002 Linux Support Services, Inc.
  ⤠T2XXP (PCI) Card 0 Span 1
 ââ[3;10Hâterfaces â â[3;37Hâ
 â  â

  â
â
 â
  âCurrent Alarms: No alarms.  
â rd 0
 Span 1   â  â
   âSync Source:T2XXP (PCI) Card 0 Span 1  
 â rd
 0 Span 2   â(R)  â
  âIRQ Misses:   0 
â
 â  â
   âBipolar Viol: 0
 â
 â  â
  âTx/Rx Levels: 0/  0 
â
 â(R) â
   âTotal/Conf/Act:  24/ 24/  0
 â
 â  â
  â 112   â
 â  â
   â123456789012345678901234â Back â   
 â
 â  â
  âTxA 
â
 â  â
   âTxB    
 â
 â  â
  âTxC 
â
 â
   âTxD    
 â
 â
  â   
â14Câ
  âRxA â Loop â
â
 â Quit â  â
   âRxB    
 â
   â
  âRxC 
â
 â
   âRxD    
 â
 ââ
  â
â
  
ââ

 T2XXP (PCI) Card 0 Span 1
 F10=Back


 I need to add 215-xxx- etc to come in to the Asterisk box.



Do you have DIDs already?  When you call a DID and watch the Asterisk
console with a little verbose, you should see the call come and how
many digits the telco is sending.

Then you need to make matching entries for those DIDs either in the
form of exact matches or pattern matches to do pretty much whatever
you can imagine.

Are you in Philly?

Thanks,
Steve Totaro

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Re: [asterisk-users] Aastra phones and park/pickup feature

[Darren Wright]

You'll want to use the XML park and pickup with the aastras.

 

Feel free to ping me off list if you need help.

 

-Darren

Dwright at d2-tech dot com

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG
Technical Support
Sent: Monday, March 03, 2008 2:45 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Aastra phones and park/pickup feature

 

We are installing Aastra phones (480's and 57i's) into a fairly simple
asterisk setup.  Although call park  pickup work fine using xfer to 700
(to park), dial 701 (to pickup), we are unable to make the park/pickup
softkey feature work on the aastra's.

 

Although we've programmed the softkeys per the manuals, they seem to
have no effect (just dead).  For example, our 57i is setup like this:

 

softkey4 type: park
softkey4 label: Park
softkey4 value: asterisk;70
softkey4 line: 1
softkey4 states: connected

 

softkey4 type: pickup
softkey4 label: Pickup
softkey4 value: asterisk;70
softkey4 value: 1
softkey4 states: idle, outgoing

(we also tried asterisk;700 with the same result).  Has anyone got the
softkey park/pickup working on aastra?

 

Thanks

Michelle


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Re: [asterisk-users] Had it with Dell Garbage

[Darren Wright]

I've used lots of Digium T1 cards on DL380 / DL320's without a hiccup.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Joshua Kinard
Sent: Tue 2/26/2008 5:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Had it with Dell Garbage


Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very 
likely, 380's as well).  I just learned this the hard way.
 
--J

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Norman 
Franke
Sent: Tuesday, February 26, 2008 5:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Had it with Dell Garbage


On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote:


On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote:

I've had it with Dell server garbage.They seem to 
change RAID

controllers as much as I change socks, and then the 
controllers don't work

with Linux, unless you load a new driver.They sell 
servers with a PCI-e

slot in them, but then you get it and find out the RAID 
controller is using

the PCI-e slot!   Their sales folks are dumber than 
rocks, and they change

them more often than I change underwear.

 [end rant].




Can anyone recommend an IBM or Gateway server that you 
have used with

Asterisk and are happy with, and which will support 
RAID-1 or RAID-5 and has

room for one or two PCI-express interface cards?







HP DL380 is my baby.




Thanks,

Steve Totaro


Ditto. We've been using HPs for a while without problem. I'm currently 
using a DL380 (a recent quad processor one) and it screams. 

-Norman



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Re: [asterisk-users] Suggestions for reliable DID providerforCanada, USA and Europe

[Darren Wright]

Yup, SIP is working ok as well, except for the cross-country 100ms round trip.  
 
Their answer was to upgrade to 1.4
 
Not an option for me. 
 
Please ping me off list so we can further discuss.
 
dwright at d2 - tech dot com
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of John Faubion
Sent: Sun 2/24/2008 1:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Suggestions for reliable DID 
providerforCanada,USA and Europe



 I've had some serious issues with Teliax as of late with their new
Denver server.  DTMF issues, IAX2 connection issues,  and major
 latency issues.  They are blaming it on 1.2 vs 1.4 asterisk compatibility
issues.   I have had zero problems with their old servers.

Interesting... I've got several lines on Teliax that have been in place for
several months and the service has been very good. Recently we connected a
new system to Teliax and I've been fighting the same issues you mention.
I've been told the problem is with my software since SIP seems to work
fairly well but not IAX. I also found out that my system is one of the first
20 systems to connect to their new Denver server. Now I'm curious about how
many others are having the same problem.

John



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Re: [asterisk-users] Suggestions for reliable DID provider forCanada, USA and Europe

[Darren Wright]

I've had some serious issues with Teliax as of late with their new
Denver server.  DTMF issues, IAX2 connection issues, and major latency
issues.  They are blaming it on 1.2 vs 1.4 asterisk compatibility
issues.   I have had zero problems with their old servers. 

 

Voicepulse has been WAY better, but no flat charges, no 729.

 

Frankly, even my broadvoice (yikes!) connection has been significantly
better, no 729. 

 

For a full Virutal PRI, I'd look at a provider that can give you the
port and SIP connections, like XO.  I've had good success with XO's
product.

 

-Darren

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: Saturday, February 23, 2008 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Suggestions for reliable DID provider
forCanada, USA and Europe

 

I used TelIAX for a while and was happy with the service.  I used it for
testing before we connected to our PRI...

 

http://www.teliax.com

 

 

On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote:





I posted the same question on asterisk-biz mailing list but didn't have
much response. So I am posting it here now.

I need a good, reliable and stable DID provider for USA, Canada and
Europe. I prefer to have fixed monthly rates for incoming and outgoing
calls and not per minute charges.

Features I need to get with DIDs are:

1. my own caller ID and caller name on outbound calls
2. multiple channels per DID
3. g729 coded
4. canreinvite=yes option
5. IAX protocol

Those who are already in this business, please advise me whom to go
with. Is getting a virtual PRI a good solution? From their websites,
they all look good so its hard to decide who is really good and will not
disappear like Allo, or start giving voice quality issues.

Thanks,
-- 
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Re: [asterisk-users] Which echo-can for Digium B410P ?

[Darren Wright]

The HWEC, not software.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Olivier
Sent: Thu 2/21/2008 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which echo-can for Digium B410P ?


Hi,

Which echo-canceler shall I pick for Digium B410P ? 

Is HPEC relevant ? Reading from its datasheet, it seems related to analog cards.
Regards


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Re: [asterisk-users] Single T1 with DIDs

[Darren Wright]

I'm surprised that Cavtel has not gotten back to you?   I use both XO and 
Cavtel in DC / Philly. Cavtel is almost unbeatable at pricing.   Paetech is 
also in the area, but I've had nothing but problems with them.  USLEC is 
another option.  
 
I have direct contact with Cavtel agents.   
 
Feel free to ping me off list to discuss.
 
-Darren
D2 Technology, INC.
 
dwright at d2 - tech dot com
 
 
 



From: [EMAIL PROTECTED] on behalf of broadband Voice
Sent: Sun 2/10/2008 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Single T1 with DIDs


Steve,
 
Can you recommend a T1 provider for me? I tried Cavalier but have no response 
and the other provider I am waiting for quotes is Broadviewnet. Thanks.

 
On 1/17/08, broadband Voice [EMAIL PROTECTED] wrote: 

Steve,
 
That is very helpful, How much are we talking about in terms of the 
loop and minute charges.  If you want it offline I can send you a private my 
with my phone number. 

 
On 1/17/08, Steve Totaro [EMAIL PROTECTED] wrote: 




On Jan 17, 2008 5:23 AM, broadband Voice [EMAIL PROTECTED]  
wrote:


Can anyone share their experience with me? I am looking 
for a provider that delivers Dialtone over T1 to terminate to my asterisk box 
and also provide DIDs. Does the DIDs come with the T1 services or those are 
purchased/charged seperately. Any help greatly appreciated. My target markets 
are Philadelphia and Washington DC Metro areas. 



I would be glad to help you out with this as I have T1s in both 
PA and MD and have been through all the paces with all of the big players in 
the area from T1s to T3s. 

I pay $.65 per DID per month on top of the loop and minute 
charges. 

Thanks,
Steve Totaro
 


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Re: [asterisk-users] IAX2 trunks unreliable becoming UNREACHABLE aftera time

[Darren Wright]

I'm having the same exact problem..2 sites connected via the internet, 2 
remote sites are unreachable, but the home site finds and can make calls just 
fine to the 2 remotes.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Royce Souther
Sent: Sun 2/10/2008 2:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] IAX2 trunks unreliable becoming UNREACHABLE aftera 
time


I have a network of offices using Asterisk that are connected via IAX2 trunks. 
The trunks work great for a day or two then for no reason at all one end of the 
trunk will become UNREACHABLE while the other end is still connected. The oving 
nly way to fix the problem is to shutdown Asterisk completly then start it 
backup again. The end that dies is not always the same, some times it is server 
A and some times it is server B. Never have I seen that both ends die, just 
one. The side that is still connected can make calls to the end that died but 
not the other way. If you call from the server with the dead IAX2 trunk you 
here All circuts are busy now. All networks have static IP addresses and 
their firewalls are setup to allow UDP 4569 to come in to the Asterisk systems.

I have been doing a lot of research into this problem. I found this bug tracker 
http://bugs.digium.com/view.php?id=5912 that talks about it being an old 
problem with  version 1.2.1 using rand() and it not being thread safe. This I 
can understand. The thread proposed using rand_r() or ast_random() in place of 
rand(), that sounds like a good idea. So when I look at my newer 1.2.18 version 
I find that it is still using rand() and the bug tracker continues to be opened 
and closed and reopened again and again.

Do I dare ask if anyone has a reliable IAX2 trunk? If so how? Should I avoid 
using IAX2 all together? I know SIP trunking is an option but it becomes a real 
management problem with trying to deal with all the many ports that need to be 
open through the firewalls, IAX2 seems like a better way to go if only it was 
reliable.

-- 
Open Source: To innovate then create
Proprietary: To imitate then litigate 

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Re: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO?

[Darren Wright]

Danger Wil Robinson!
 
Don't do it.  If the cost is negligible, you are going to give up a huge 
control / reliability factor.   Unless you dedicated a T1 to just voice, you'll 
not be able to guarantee quality.   
 
I've had a few small companies use VOIP trunks with POTS backup, but I wouldn't 
even consider switching a PRI to SIP.
 

 


From: [EMAIL PROTECTED] on behalf of Jim Canfield
Sent: Fri 10/5/2007 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO?



I've been considering replacing a PRI with SIP or IAX trunks.  The
monthly cost difference is marginal, but it would save a bit on the
hardware side and soft trunks would be easier to manage. I can't help
but wonder what I would be giving up?  I'd like to hear some lessons
learned from those who are doing it or decided, for whatever reason,
it's a bad idea.








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Re: [asterisk-users] Best config for 12 FXO system?

[Darren Wright]

None are great options.   I'd use a T1 card and a channel bank.  

At minimum I'd do the single 2400P.   IRQ problems are going to be a
bear with multiple cards.

-Darren



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Tony Mountifield
 Sent: Wednesday, October 03, 2007 12:02 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Best config for 12 FXO system?
 
 I have a client who wants a Meetme box with 12 FXO ports, to connect
 to Analogue lines coming from an Ericsson PBX.
 
 It looks like I could do this with four different hardware
configurations:
 
 a) three TDM04B cards (based on TDM400P)
 b) one TDM04B and one TDM808B
 c) one TDM804B (or TDM854B?) and one TDP808B
 d) one TDM2403B (half filled TDM2400P)
 
 Apart from considerations of cost and PCI slot availability, are there
any
 technical reasons to choose one of the above configurations over the
 others?
 
 Cheers
 Tony
 --
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
 
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Re: [asterisk-users] ADIT 600 CMG = Asterisk question

[Darren Wright]

Are you talking about PRI's?   The ADIT's can't handle termination of PRI's, 
only DI. I use them all the time to breakout FXS/FXO's for incoming and 
outgoing analog lines, but they have a tendency to introduce lots of 
echo.I've had to use HWEC every time I use the 600.
 
-D
 



From: [EMAIL PROTECTED] on behalf of Barton Fisher
Sent: Mon 9/3/2007 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ADIT 600  CMG = Asterisk question



I've searched but can't find an answer as to how many MGCP paths can a
single ADIT/CMG card support? It appears it's only 24 ports, maybe 48.
What I'd like to do is install 6 Telco T1's into a single (or more) Adit
600 and route inbound calls towards asterisk. Can I have more than one
CMG in a single chassis?

Or maybe you know of a better way to connect T1's to asterisk without
zaptel cards using SIP Trunks?

Thanks

Bart

--

Barton Fisher
Innovative Communications
714-228-5400 Ext 5410
http://www.icpage.com



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Re: [asterisk-users] FW: The trixbox Revolution Continues! Sign upforthe Webinar.

[Darren Wright]

Looks like it's time to fork..

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Lenz
 Sent: Monday, August 13, 2007 7:28 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] FW: The trixbox Revolution Continues!
Sign
 upforthe Webinar.
 
 
 I see that the TrixBox Pro website is available now:
 http://www.trixbox.com/products/trixbox-pro/
  From what I'm reading, there is a free version available, plus two
other
 versions, one at $9.99/per user/per month and the other at $19.99/per
 user/per month.
 They have a centralized architecture for monitoring and access with
local
 iron on customer premises. (Hmm... I seem to remember Signate had a
 similar, though on a smaller scale - approach, and it was not very
 successful - but maybe the market is more mature now).
 
 Anyway, TrixBox Pro is available for download, so I guess it's time to
 give it a go :-)
 l.
 
 
 
 
 On Sat, 11 Aug 2007 14:55:36 +0200, Steve Totaro
 [EMAIL PROTECTED] wrote:
 
  Let's fork Digium's GUI.
 
  Zeeshan Zakaria wrote:
  Why don't they say FreePBX. After all trixbox is all about FreePBX.
If
  they remove FreePBX from Trixbox, nothing is left in it. A half
  working HUD, and another small little things don't make any major
  difference after all.
 
  So are the FreePBX developers with the Fonality team or with the
open
  source community? I think now is the time for someone to come up
with
  another similar product to compete with them. Are there any such
  people around who agree with me and can spend time and expertise to
  develop such a thing? Lets start our own FreePBX type project to
keep
  the beauty of open source telephony available to all in case in a
year
  or two they all go commercial..
 
 --
 Loway Research - Home of QueueMetrics
 http://queuemetrics.com
 
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Re: [asterisk-users] North American voice BRI - Informal survey

[Darren Wright]

I wonder if this is issue is largely limited to to Canada.  (thus
limiting the market)  In the states I think you can get PRI for around
$250.  Am I right?  In Canada, you have to have about 9 or 10 lines to
justify a PRI.  At $250, the cost and added features could justify PRI
at around 4 lines.  Mind you, that still leaves a whole tonne of systems
at the 4 lines and under mark.  




 

No way.message rates lines hover at $350, and flat rate's run
$450-$500 or so.

 

 

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RE: [asterisk-users] Dry Copper Pair

[Darren Wright]

How far is the run?   I'm wondering what you mean by $0 for hardware?   I 
typically use Ethernet extenders,  but it has been a crapshoot on the quality 
from Verizon.
 
What is a BANA circuit?
 
Finding someone who will even sell it to you has been somewhat of a game as 
well.
 



From: [EMAIL PROTECTED] on behalf of Smith, Rick
Sent: Fri 5/11/2007 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dry Copper Pair



Let me see.  Dry pair, $40 for the circuit.

Hardware for each end, $0.

Not paying verizon for DSL or PTP T-1 service?  Priceless.

It's a BANA circuit, btw, in Verizon territory.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Friday, May 11, 2007 3:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dry Copper Pair


You might be able to try ordering it from a CLEC that can provision it
over UNE and sell it for considerably less.  Depending on your area,
their interconnection agreement, tariffs, etc.  So, your mileage may
vary.

On Fri, 11 May 2007, Matt said something to this effect:

 Hi,
 Does anyone know of a way to get a dry copper pair (also known as an
alarm
 line) from Verizon for less than $20/end?   I know we have been able
to get
 them, but they come out to $40/month for a circuit.. and there's no
 dial-tone over it


--
Alex Balashov   [EMAIL PROTECTED]
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[asterisk-users] Passing PRI traffic to remote * over IAX

[Darren Wright]

We are moving our office, but our PRI isn't moving for a while yet.

 

I'd like to setup a box at the old office to receive -ALL-- PRI traffic
and send it over an IAX trunk to another Trixbox install at the new
office.  Everything should go, period.  

 

Any ideas on a simple dialplan to make this happen?

 

Thanks,

 

-Darren

 

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RE: [Asterisk-Users] An FXO version of IAXy?

[Darren Wright]

I'd be very interested to know how your Audiocodes install goes.
 
My experience was not good at all with an MP-108.  It was very inconsistent, 
and extremely hard to configure.  I paid $275 for support from ABP, the 
Audiocodes USA support provider, which was a waste.  Thier answer was mostly 
oh your lines are different, or  it must be something on your asterisk box. 
 I've gotten lots of Digium, Sangoma, and Sipura external boxed working.the 
Audioocodes was impossible
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of David Rahn
Sent: Fri 3/31/2006 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] An FXO version of IAXy?



I believe that Avaya is rebranding this device for use with there new
system ( actually they bought NIMCAT) the phones are adhoc networked -
no server- ( anyway this may be why it is a hard device to purchase.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr.
Michael J. Chudobiak
Sent: Wednesday, March 22, 2006 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] An FXO version of IAXy?


 D-Link has a 4 port FXO device on their site.
 http://www.dlink.com/products/?sec=2pid=451

 Apparently it hasn't shipped yet and costs $500.00

I've been testing a AudioCodes MP104-FXO-C3S (around $1000) 4-port FXO
box. It works, but the number of configuration options are staggering,
complex, and inter-related, and the documentation  support just aren't
good enough to make installation easy.

The D-link DVG-3004S is pretty much impossible to get.

There is also the Mediatrix 1104 (also around $500), but it is reputed
to be hard to configure (no web interface - just snmp!).

Slapping a Sangoma A200 into a computer (and configuring it through
Zaptel/Asterisk) is much, much simpler than trying to make the
appliance gateways work, at least in my experience.


- Mike
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[Asterisk-Users] 7940/60 SIP Call Park Button

[Darren Wright]

Anyone figure out a way to add a call park
button, either on the bottom or on the sides during the call?










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[Asterisk-Users] More Voicemail prompts

[Darren Wright]

Can Comedian Mail handle more than just an away and busy message?   I've got a 
client that would like even more of them.
 
I can write an app to replace messages externally, but I was wondering of 
comedian could handle it internally.
 
 
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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

[Darren Wright]

ThanksI've got the SEPMAC files that I use successfully with SCCP.

-D


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: Tuesday, March 07, 2006 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco
7970

On Mon, 2006-03-06 at 22:11 -0500, Darren Wright wrote:
 InterestingI've upgraded the 7970 to SIP, but it is still saying
 unprovisioned.  I've got a SIPMAC file, but it is still looking for
the
 SEPMAC file...
 

That's correct - the CCM5 loads only look for SEP files.  Even when you
give it one, it will not register with Asterisk.  If you need a fully
formatted SEPxml file, I will email you one off line for a 70.

 
 Anyone got this working yet?

Nope :(

 -D
 
 
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RE: [Asterisk-Users] HW Echo Cancellers

[Darren Wright]

Forget the orion.lots of DTMF problemstech support is not
Terribly well spoken.

Look for ANY of the 257* series...

Just ebay for t1 echo

-D


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, March 08, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HW Echo Cancellers

  Tellabs looks a little too up-scale for what I need :). $1k for a
  single port orion unit might be worth considering for really
stubborn
  installs though.
 

 Why? they go for around $100.00 on eBay.

What goes for $100 on eBay?  Tellabs?  or Orion?  I can't find any
Orion equipment on eBay.  What model Tellabs am I looking for?
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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

[Darren Wright]

OK.

 I've got the COP SIP filehow do we use this thing on the 7970?

-Darren


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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

[Darren Wright]

InterestingI've upgraded the 7970 to SIP, but it is still saying
unprovisioned.  I've got a SIPMAC file, but it is still looking for the
SEPMAC file...


Anyone got this working yet?

-D


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RE: [Asterisk-Users] Echo Cancelation on TE110P

[Darren Wright]

only for the whole cardthe tx and rx gain affect all 24 channels.
 
-D
 



From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Fri 3/3/2006 11:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo Cancelation on TE110P


On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few 
users are complaiining about echo. According to the users, the echo seems to be 
phone number dependant. They claim that certain phone numbers have echo while 
others dont. Are there any tuning parametes like there is for a TDM400 card? 
 
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
http://www.techdatapros.com http://www.techdatapros.com/  
 
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RE: [Asterisk-Users] Software E.C. Along with Tellabs

[Darren Wright]

You may want to turn the Rx gain down a bit..

-Darren


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Joseph Tanner
 Sent: Wednesday, February 15, 2006 10:35 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Software E.C. Along with Tellabs
 
 Shouldn't hurt, I'd give it a try.  But first you may want to fiddle
 with the Tellabs configuration some more.  This has some good
 information:  http://www.voip-
 info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
 
 Joseph Tanner
 
 On 2/15/06, Doug Lytle [EMAIL PROTECTED] wrote:
  Since putting my Tellabs EC into place around 2 weeks ago, the echo
  problem has almost been eliminated.  Reports of some very faint
echo,
  but everybody is happy.
 
  My question is, if I were to also turn on the Asterisk Software EC,
  would this remove any residual echo that may make it past the
Tellabs
  Hardware EC.
 
  Thanks,
 
  Doug
 
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RE: [Asterisk-Users] TE411P Really Bad Echo ORION

[Darren Wright]

The Orion echo canceller is just ok.     The
Tellabs units work just as well if you dont mind 10 mins of soldering.



I have the orion running with an adit 600
and a TE110P.   Echo cancel is fairly good, but I have loads of problems with
DTMF digits.



-Darren

    













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Rob Lith
Sent: Saturday, February 11, 2006
8:10 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
TE411P Really Bad Echo





TE406P/411P and if you
need to go dedicated to hanlde all possible look at an external dedicated
canceller like www.oriontelecom.com
VCL-E1 ECHO CANCELLER (1U Version) ± $1295

Rob



On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


On Sat, 11 Feb 2006, Rob Lith wrote:
 It claims to have carrier-grade algorithms - don't glibly
translate that 
 to carrier grade hardware, it's a PCI card...

What echo canceller hardware do you recommend for an asterisk PC?

-Dan
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RE: [Asterisk-Users] TE411P Really Bad Echo

[Darren Wright]

Eh.   Not for $1000 more, and I've got both in production.  Customer service 
was an issue.  

-Darren


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
 Sent: Saturday, February 11, 2006 10:19 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] TE411P Really Bad Echo
 
 On Sat, 11 Feb 2006, Rob Lith wrote:
  TE406P/411P and if you need to go dedicated to hanlde all possible look
 at
  an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
  CANCELLER (1U Version) ± $1295
 
 Is the orion echo canceller a higher quality EC than tellabs?
 
 -Dan

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RE: [Asterisk-Users] TE411P Really Bad Echo

[Darren Wright]

WELL!

The Orion guys agreed to send me one as a demo for 30 days.  I'm doing 1 
install / week now, so it was a good business opportunity for them. I had 
issues with DTMF during the test phase, and the tech guys were not terribly 
helpful.  3 weeks into the test (a week early) collections calls me and asks 
why I haven't paid yet !?!?!?!?  I fought them for another 2 weeks before I 
figured out 90% of the DTMF issues, and then paid.

-D




 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C F
 Sent: Sunday, February 12, 2006 12:28 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] TE411P Really Bad Echo
 
 Darren, how was customer service an issue? I mean once you got one to
 work, it just plug and forget.
 
 On 2/12/06, Darren Wright [EMAIL PROTECTED] wrote:
  Eh.   Not for $1000 more, and I've got both in production.  Customer
 service was an issue.
 
  -Darren
 
 
   -Original Message-
   From: [EMAIL PROTECTED] [mailto:asterisk-users-
   [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
   Sent: Saturday, February 11, 2006 10:19 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [Asterisk-Users] TE411P Really Bad Echo
  
   On Sat, 11 Feb 2006, Rob Lith wrote:
TE406P/411P and if you need to go dedicated to hanlde all possible
 look
   at
an external dedicated canceller like www.oriontelecom.com VCL-E1
 ECHO
CANCELLER (1U Version) ± $1295
  
   Is the orion echo canceller a higher quality EC than tellabs?
  
   -Dan
 
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RE: [Asterisk-Users] Tellabs 2572 EC Photos here.

[Darren Wright]

HAHAHA!

That's EXATCLY the same setup I'm running...even down to the cards in
the 600


Working like a champ.

-d


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Doug Lytle
 Sent: Saturday, January 21, 2006 8:06 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Tellabs 2572 EC Photos here.
 
 [EMAIL PROTECTED] wrote:
  Hello Dan,
 
  Have a look at this link:
 
  http://www.adcomcorp.com/asterisk/tellabs
 
  I got those pictures up there, may be of help.  In essence, 1 pair
is
  either a tx pair or an rx pair.
 
 
 Very cool!  I've got two coming and this will be a big help.
 
 Thanks,
 
 Doug
 
 --
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a little
Temporary
 Safety, deserve neither Liberty nor Safety.
 
 
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RE: [Asterisk-Users] HW Echo Cancellers

[Darren Wright]

I have used the orion...you can buy right from them.  However, I was not
impressed with their sales teamI have one on a beta test, and they
threatened to call a collection agency in when I refused paybent before
the beta expired.

I had some weird DTMF issues with the Orion, otherwise ok.

-D


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Friday, December 16, 2005 5:52 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] HW Echo Cancellers

Hi,

To solve echo problems, I'm considering 2
alternatives.
1 Sangoma A104d
   - I can't find support for asterisk 1.2.1
2 Desktop echo canceller
   -
http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html
   - I want to know where to buy and price.

Any suggestion is appreciated.

Thanks.
Jason.

p.s. : asterisk cli command reload can change
rx_gain and tx_gain?

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RE: [Asterisk-Users] HW Echo Cancellers

[Darren Wright]

$1k for a single port T1
 
 
I've gone down the Tellabs route, and am infinitely more happy.thanks C F 
for the docs..
 
-D
 



From: [EMAIL PROTECTED] on behalf of Steve Davies
Sent: Fri 12/16/2005 10:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HW Echo Cancellers



On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote:
 I have used the orion...you can buy right from them.  However, I was not
 impressed with their sales teamI have one on a beta test, and they
 threatened to call a collection agency in when I refused paybent before
 the beta expired.


Can you give an indication of price for their units? I've tried
mailing a couple of times, but received no answer. I am just
interested to know what price range we'd be looking at.

Thanks,
Steve
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RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install

[Darren Wright]

What about telnet access?  If you don't know the Ethernet IP use a
packet sniffer to detect it and then telnet to it.   It may not be
password protected.

-D


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, November 23, 2005 9:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk
install

Thanks Jerry,

I have called Carrier Access and they can reset the password but for a
considerable fee.   We have serial access but after it boots it
immediately
asks for a username and password.  We have the username but the password
is
not what it is suppose to be.   There's a reset switch on the faceplate
but
I think the LOCAL SET is OFF and that is why it doesn't respond.  Their
manual says the Reset switch is not under the control of LOCAL SET, yet
it
doesn't seem to work.  Well, we might not know the proper boot sequence.
It
contains flash memory and there is a timing that important to that reset
procedure.  Anyone's help is much appreciated.

--Jim   

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Wednesday, November 23, 2005 7:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk
install

Not sure but are you connecting via serial or ehternet? Seems to be the
serial had a way to do this easily on bootup. Otherwise I would be
interested for future reference. Carrier Access does have a good support
team, just need to know your serial number.

On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

 Looking for a way to hard reset a ADIT 600 just purchased used.   
 But it
 seems to have a master password already set.  We've tried the front 
 reset but maybe we don't have the right sequence of boot order.  Any 
 help would be much appreciated?  - Jim



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RE: [Asterisk-Users] Linksys SPA941

[Darren Wright]

I agree, this phone has now filled a large void in my client base, especially 
for small systems in the 4 Trunk x 8 phone range.   The Ciscos were just way 
too expensive to make it viable. 
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Julian Lyndon-Smith
Sent: Mon 11/21/2005 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Linksys SPA941



Just picked up two of these puppies from my parcelforce depot.

Man, they are smart phones. They look the business. I installed one
within seconds, fantastic web configuration - much like the SPA3000 box.

Speakerphone sounds good, handset feels and sounds good.

I'll be using this heavily over the next couple of days, and I'll let
you all know how we find it.

And nearly half the price of a second-hand 7940 it's a real steal.

Julian.
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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[Darren Wright]

I've done some very interesting testing recently:

The 64ms cards are working wonderful.  $19.00 a pop is a steal.   They
work great with your KB1 canceller, but any others cause HORRIBLE echo.
I am facing the tail end AWAY from the asterisk boxso the echo is
definitely coming from somewhere between the TE110P and the Adit 600.  

Interesting hunh?

I have not gotten my hands on a VX2 card yet, but the 64's are working
so well I'm not sure there is a reason too.  


The Orion canceller is very nice as well, but $1000.

-Darren


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Boutilier
Sent: Wednesday, October 19, 2005 12:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Darren
Wright
 Sent: Tuesday, October 18, 2005 7:42 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600
 
 8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual 
 Xeon running 1.0.9 and 1.2 (tried both)
 
 The echo is insurmountable.  I have tried everything, and the 
 pots lines are clean.  If I go from an FXO on the Adit 600 straight to
an FXS, I
 get no echo from an analog phone.  

No echo that you can hear - remeber that echo relies on two things, a
reflected signal and a delay between the transmission and the reception
of the signal long enough for the brain to perceive it. Looping the
channel bank will not introduce any delays. Passing through Asterisk
will, by design.

 I put an 128ms T1 echo canceller in between the adit and the 
 TE110P, and the echo was still horrible.  
 
 I finally disabled the Zapata echo cancellerand WHAMMO!  It's
 perfect now.  
 

It sounds like something is confusing the zaptel canceller causing it to
distort the signal. It seems to be very sensitive to signals that are
too 'hot' (ie. too loud). Try lowering the gain on the signal going out
of the channel bank into the T1. If it's too quiet try increasing the RX
gain on the Zaptel side to compensate.

{clip}
 Any ideas so I don't have to spend $1000 on an echo canceller?
 

I provided the patches to 1.2 that formed the basis for the kb1 echo
canceller, which is a derivative of the mark2 used in v1.0, and I still
use a 64ms Tellabs hardware echo can as well as the zaptel echo
canceller. Note that, in my case at least, the zaptel tends to handle
those echos that leak through the Tellabs gear - such as acoustic room
echos from speaker phones or cheap cordless handsets. 

If you need the echo issue resolved, stick with hardware cancellation.
If you don't want to spend $1k, take a look at
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers.
It's not everyones cup of tea, but it works fine for me which is why I
shared it.

The Zaptel echo can will be fixed so it performs predictably for
everyone eventually, but until then go with 3rd party T1 gear if you
want it reliably avoided.

Hope that helps.

Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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RE: [Re] Re: [Asterisk-Users] Echo canceller on TE406 Asterisk

[Darren Wright]

I have given up totally on Digium based echo cancel, hardware or
software.  The KB1 is the best so far, but still unacceptable.  I
installed a hardware echocan FACING the T1 card in the asterisk box, and
all is perfect.   No complaints from any of my clients since taking that
leap.

-Darren


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[Asterisk-Users] Terrible echo with Te110P and Adit 600

[Darren Wright]

8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual Xeon running
1.0.9 and 1.2 (tried both)


The echo is insurmountable.  I have tried everything, and the pots lines
are clean.  If I go from an FXO on the Adit 600 straight to an FXS, I
get no echo from an analog phone.  

I put an 128ms T1 echo canceller in between the adit and the TE110P, and
the echo was still horrible.  

I finally disabled the Zapata echo cancellerand WHAMMO!  It's
perfect now.  

The TE110P is on it's own IRQ.. and the machine has PLENTY of
horsepower.

Any ideas so I don't have to spend $1000 on an echo canceller?

-Darren




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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[Darren Wright]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Boutilier
Sent: Wednesday, October 19, 2005 12:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Darren
Wright
 Sent: Tuesday, October 18, 2005 7:42 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600
 
 8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual 
 Xeon running 1.0.9 and 1.2 (tried both)
 
 The echo is insurmountable.  I have tried everything, and the 
 pots lines are clean.  If I go from an FXO on the Adit 600 straight to
an FXS, I
 get no echo from an analog phone.  

No echo that you can hear - remeber that echo relies on two things, a
reflected signal and a delay between the transmission and the reception
of the signal long enough for the brain to perceive it. Looping the
channel bank will not introduce any delays. Passing through Asterisk
will, by design.

---
I meant that if I take an incoming POTS line to the FXO port, map that
to the FXS, and then make a call from the analog phone to the same
person that I tried calling on the Cisco 7960, the echo on the 7960 is
terrible, and the FXS port is just fine.  
--- 

 I put an 128ms T1 echo canceller in between the adit and the 
 TE110P, and the echo was still horrible.  
 
 I finally disabled the Zapata echo cancellerand WHAMMO!  It's
 perfect now.  
 

It sounds like something is confusing the zaptel canceller causing it to
distort the signal. It seems to be very sensitive to signals that are
too 'hot' (ie. too loud). Try lowering the gain on the signal going out
of the channel bank into the T1. If it's too quiet try increasing the RX
gain on the Zaptel side to compensate.


Agreed.  I've tried them lowered to the point that DTMF becomes an
issue, and they have the volume pegged on the 7960's to even hear the
callers.  I cannot adjust the TX/RX on the T1 coming out of the Adit
600.  I can adjust the FXO ports, as well as the TE110P.  

-



{clip}
 Any ideas so I don't have to spend $1000 on an echo canceller?
 

I provided the patches to 1.2 that formed the basis for the kb1 echo
canceller, which is a derivative of the mark2 used in v1.0, and I still
use a 64ms Tellabs hardware echo can as well as the zaptel echo
canceller. Note that, in my case at least, the zaptel tends to handle
those echos that leak through the Tellabs gear - such as acoustic room
echos from speaker phones or cheap cordless handsets. 

If you need the echo issue resolved, stick with hardware cancellation.
If you don't want to spend $1k, take a look at
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers.
It's not everyones cup of tea, but it works fine for me which is why I
shared it.

The Zaptel echo can will be fixed so it performs predictably for
everyone eventually, but until then go with 3rd party T1 gear if you
want it reliably avoided.


Yupgot one running at home thanks to your WIKI.  But for clients
moving forward, I need something a bit more mainstream.  I'm
disappointed that the TE110P + adit 600 has been an issue on multiple
systems now, and that the software echo canceller has been a major
failure.  

It makes that solution WAY to expensive with the echo
cancellerthat's well into the 2k range, and a good FXO - SIP gateway
with echo canceling is significantly less than that.  

Thanks for your help

-Darren
 

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RE: [Asterisk-Users] Adit 600 FXO card sound quality

[Darren Wright]

Sounds like no comfort noise.do you see this as well on VOIP trunks?  sign 
up for goiax or something like that to testI tend to think it is between 
asterisk and the polycoms.  
 
You could also test with a Sipura SP3000 as a replacement to see if you have 
the same issue, or if you have an extra FXS card, cross connect one of the FXS 
to the FXO channels.
 
My FXO's in the adit 600 are fantastic.
 
-D
 



From: [EMAIL PROTECTED] on behalf of C F
Sent: Sun 10/2/2005 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Adit 600 FXO card sound quality



I have an adit 600 with one fxo card connected to a Digium single span T1 card.
CallerID, disconnect supervision work perfect, however the users
complain that they have some sound quality issues, after testing it I
realized that whenever one is in a phone call they get like silence
between the sounds coming from the other party, almost like a cell
phone, in other words if there is no sound coming from the other party
it sounds like they have hung up, which is very annoying.
The usres all use Polycom IP 501s connected to Asterisk which is
running a a TE110, here is the configs from the Adit:
-
-Adit 600 configuration file
-Created on 01/04/2002 at 14:26:40 for root
-This file is valid for the following configuration only:
-
-CardType
-
-SLOT A   T1x2 SW Version:  9.0.0
-SLOT 1   FXOx8
-SLOT 2   FXOx8
-SLOT 3   FXSx8
-SLOT 4   FXSx8
-SLOT 5   FXSx8
-SLOT 6   RTRx1
-NOTES:
-1. It is necessary to issue the commands 'restore defaults'
-   and 'reset' BEFORE downloading the configuration file to
-   ensure proper configuration.
-2. Lines beginning with '-' will be ignored as comments
-   by the CLI.  Before downloading, review the sections of
-   the configuration file delimited by these comments and
-   delete the commands that are not needed (e.g. 'set ip
-   address' and 'add user' are likely candidates for
-   deletion).
-3. While downloading, a character delay of 5 ms and a line
-   delay of 300 ms is recommended.
-

-Turning off verification messages.

set verification off

-Setting local off.

set local off

-Disconnecting all connections.

disconnect a
disconnect 1
disconnect 2
disconnect 3
disconnect 4
disconnect 5
disconnect 6

-Setting IP addresses.

set ethernet ip address 192.168.1.51 255.255.255.0
set ip gateway 192.168.1.1

-Setting the SNMP MIB-II System Group objects.

set snmp getcom public
set snmp setcom public
set snmp trapcom public
set snmp trapauth enable
set snmp trapevent all



-Setting slot a.

set a:1 up
set a:1 fdl none
set a:1 lbo 1
set a:1 framing esf
set a:1 id toasterisk
set a:1 linecode b8zs
set a:1 loopdetect csu
set a:1:1-24 side drop
set a:1:1-24 type voice
set a:1:1-24 signal ls
set a:2 down
set a:2 fdl none
set a:2 lbo 1
set a:2 framing esf
set a:2 id CAC DS1# A:2
set a:2 linecode b8zs
set a:2 loopdetect csu
set a:2:1-24 side drop
set a:2:1-24 type voice
set a:2:1-24 signal ls

-Setting slot 1.

set 1:1-8 signal lscpd
set 1:1-8 txgain -3
set 1:1-8 rxgain -6

-Setting slot 2.

set 2:1 signal lscpd
set 2:1 txgain -3
set 2:1 rxgain -6
set 2:2-8 signal ls
set 2:2-8 txgain -3
set 2:2-8 rxgain -6

-Setting slot 3.

set 3:1-8 signal ls
set 3:1-8 txgain -3
set 3:1-8 rxgain -6
set 3:1-8 linelength short

-Setting slot 4.

set 4:1-8 signal ls
set 4:1-8 txgain -3
set 4:1-8 rxgain -6
set 4:1-8 linelength short

-Setting slot 5.

set 5:1-8 signal ls
set 5:1-8 txgain -3
set 5:1-8 rxgain -6
set 5:1-8 linelength short

-Setting slot 6.

set 6 proxy disable

-Setting users.

add user root

-Setting network id.

set id channelbank

-Setting primary and secondary clock sources.

set clock1 a:1
set clock2 internal

-Making connections.

connect a:1:1-8 1:1-8

-Turning verification on.

set verification on

==
If anybody got this working perfectly please let me know.
Thank You
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RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice

[Darren Wright]

I am also a long time client, and have no incoming BV today.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Jason Schafer
Sent: Mon 9/26/2005 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice



 Does asterisk says something in the verbose console?

I'm not sure what the verbose console is, but I can run sip debug and
post the output when I make an inbound call.

 please post your sip.conf relevant entries for BroadVoice.

[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
pedantic=no ; added for Broadvoice support 8/3/05 EK
externip=216.xxx.xxx.xxx
localnet=172.xxx.xxx.0/255.255.255.0


I have just
 cancelled with BroadVoice (too much latency for the places i wanted to
 call), so i never used the incoming number. But im glad to help if i can.

I have outbound setup on VOIPJet, my intent with the Broadvoice is to
setup a forward on busy with my landline to roll over to the BV number.

Here's the output from sip debug

m=audio 14008 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

13 headers, 12 lines
Using latest request as basis request
Sending to 147.135.0.128 : 5060 (non-NAT)
Found no matching peer or user for '147.135.0.128:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.0.128:14008
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for s in from-sip-external
list_route: hop:
sip:[EMAIL PROTECTED]:5060;ep=147.135.0.129;transport=udp
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: Schafer Trish
sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179
To: Jason Schafersip:[EMAIL PROTECTED];user=phone
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


  to 147.135.0.128:5060
 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack
 -- Executing Goto(SIP/147.135.0.129-095da350, from-pstn|s|1) in
new stack
 -- Goto (from-pstn,s,1)
 -- Executing GotoIf(SIP/147.135.0.129-095da350,
1?from-pstn-reghours|s|1:) in new stack
 -- Goto (from-pstn-reghours,s,1)
 -- Executing GotoIf(SIP/147.135.0.129-095da350,
0?from-pstn-reghours-nofax|s|1:2) in new stack
 -- Goto (from-pstn-reghours,s,2)
 -- Executing Answer(SIP/147.135.0.129-095da350, ) in new stack
We're at 216.xxx.xxx.xxx port x
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: Schafer Trish
sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179
To: Jason
Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1782 1782 IN IP4 216.xxx.xxx.xxx
s=session
c=IN IP4 216.xxx.xxx.xxx
t=0 0
m=audio 14138 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

  to 147.135.0.128:5060
 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack
asterisk1*CLI

Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr
From: Schafer Trish
sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179
To: Jason
Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 ACK
Contact: sip:[EMAIL PROTECTED]:5060;transport=udp
Max-Forwards: 69
Content-Length: 0


9 headers, 0 lines
 -- Executing SetVar(SIP/147.135.0.129-095da350, intype=aa_2) in
new stack
 -- Executing Cut(SIP/147.135.0.129-095da350, intype=intype|-|1)
in new stack

RE: [Asterisk-Users] Carrier Access - Access Bank I config

[Darren Wright]

I tried for weeks with an AB I, and never got anywhere...I could not get the T1 
to sync properly.  I switched exclusively to ADIT 600's and have had no issues 
since.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Time Bandit
Sent: Mon 9/26/2005 2:25 PM
To: asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Carrier Access - Access Bank I config



Hi,

Is there somebody using an Access Bank I with Asterisk that could
share the secret ingredients needed to make it work ?

I've searched around and found some info, I tryed almost every
configuration possible but I can't seem to find the right combination.
If someone could provide me with the config needed on Asterisk as well
as the dip-switch settings on the channel bank part, I would be really
greatfull.

Thanks
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RE: [Asterisk-Users] Orinoco Injectors

[Darren Wright]

RightoI was asking if anyone knew if the orinoco's were standard
802.3af
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corey S.
McFadden
Sent: Friday, September 23, 2005 9:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Orinoco Injectors


  Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12)

  to work with the Cisco 79* series phones?
 
  I'm not sure if the are the statndard POE or not
 
 Cisco's phones are not standard POE.  They reversed the polarity, and 
 I think they run the power hot all the time.  Can't remember
specifically.


Cisco phones will work with any 802.3af standard 48V PoE midspan
injector or PoE switch.  You just need a patch cable made to the correct
spec.



*
This message has been scanned for viruses and
dangerous content, and is believed to be clean.

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[Asterisk-Users] Orinoco Injectors

[Darren Wright]

Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12) to
work with the Cisco 79* series phones?

I'm not sure if the are the statndard POE or not

-Darren

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[Asterisk-Users] Adding DNIS digits

[Darren Wright]

Is there a way to add DNIS digits, and pass it onto another line?

My provider will noy supply DNIS digits over analog lines, so I'd like
to take a call on a trunk, add some DNIS digits, and pass it to another
asterisk system...

-Darren

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RE: [Asterisk-Users] Adding DNIS digits

[Darren Wright]

Because I am trying to simulate DID lines which my telco cannot provide
over analog. I'd like to build it around DID's so that the numbers are
portable if they ever convert to VOIP trunks.  

-Darren


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, September 12, 2005 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Adding DNIS digits

Whats wrong with dial(zap/channel/${EXTEN}addeddnisdigitsgoeshere)

On 9/12/05, Darren Wright [EMAIL PROTECTED] wrote:
 Is there a way to add DNIS digits, and pass it onto another line?
 
 My provider will noy supply DNIS digits over analog lines, so I'd like
 to take a call on a trunk, add some DNIS digits, and pass it to
another
 asterisk system...
 
 -Darren
 
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RE: [Asterisk-Users] Adding DNIS digits

[Darren Wright]

Actually, I have:

8 VOIP  FXS (provided by Telco, new product)\

Adit600 --- TE110P
8 POTS -/

The plan it to move totally to VOIP when they support MGCP directly, and
they will be able to move the POTS lines to VOIP when ready.


I'm going to have to group the channels, because there are 3 companies
running off this one systemeach one with its own hours and
autoattendant.  

The context and goto solution still requires management on the main
system..I'd rather have another little box that can be removed that adds
the DNIS digits and passes them to the main system.

-Original Message-
From: C F [mailto:[EMAIL PROTECTED] 
Sent: Monday, September 12, 2005 4:10 PM
To: Darren Wright; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Adding DNIS digits

In other words you telling me you have something like the following:

TDM40B in an Asterisk box, or a T1 card to channel bank, configured in
zapata.conf to go to context = incoming
in extensions.conf you have:
[incoming]
whatever extensions.

What you want to make sure is that if/when you switch the TDM40B to a
PRI with DIDs that you don't have to rewrite incoming context.
This is how you would do it:
[incoming]
exten = s,1,Goto(1234,1) ;just jump to the future DID which is 1234

exten = 1234,1,Noop() ;here is my future did

The only problem is that you don't realy know what DID numbers you
will get from your provider, so you are not saving anything.

If I misuderstood you please clarify.

On 9/12/05, Darren Wright [EMAIL PROTECTED] wrote:
 Because I am trying to simulate DID lines which my telco cannot
provide
 over analog. I'd like to build it around DID's so that the numbers are
 portable if they ever convert to VOIP trunks.
 
 -Darren
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Monday, September 12, 2005 2:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Adding DNIS digits
 
 Whats wrong with dial(zap/channel/${EXTEN}addeddnisdigitsgoeshere)
 
 On 9/12/05, Darren Wright [EMAIL PROTECTED] wrote:
  Is there a way to add DNIS digits, and pass it onto another line?
 
  My provider will noy supply DNIS digits over analog lines, so I'd
like
  to take a call on a trunk, add some DNIS digits, and pass it to
 another
  asterisk system...
 
  -Darren
 
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RE: [Asterisk-Users] Adding DNIS digits

[Darren Wright]

ABSOLUTELY.

-D
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Monday, September 12, 2005 4:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Adding DNIS digits

Oh...I missed a post...

You have 8 lines coming in, connected to a channel bank. You have 3
companies on a single asterisk server. 

You need to populate the DNIS based on which pots line the call came in
on and then route it as needed?

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Monday, September 12, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Adding DNIS digits

EhI don't get it

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wright
Sent: Monday, September 12, 2005 4:10 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Adding DNIS digits

Because I am trying to simulate DID lines which my telco cannot provide
over analog. I'd like to build it around DID's so that the numbers are
portable if they ever convert to VOIP trunks.  

-Darren


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, September 12, 2005 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Adding DNIS digits

Whats wrong with dial(zap/channel/${EXTEN}addeddnisdigitsgoeshere)

On 9/12/05, Darren Wright [EMAIL PROTECTED] wrote:
 Is there a way to add DNIS digits, and pass it onto another line?
 
 My provider will noy supply DNIS digits over analog lines, so I'd like

 to take a call on a trunk, add some DNIS digits, and pass it to
another
 asterisk system...
 
 -Darren
 
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[Asterisk-Users] adding DNIS digits

[Darren Wright]

Situation:

8 POTS lines, 3 companies, 1 system.  Channel banking the POTS lines
onto a T1 thru an ADIT 600.

The only way our carrier will provide DNIS is thru Analog DID #'s.

Anyone know of a piece of hardware that can add DNIS digits to a
particular line?

-Darren



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RE: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection

[Darren Wright]

I've bought 2 Adit600's on ebay now for less than $500, for exactly the
install you are talking about.

I'm wondering why they wouldn't be a T1 though...should be less than
12-16 FXO's.

-Darren 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: Thursday, September 08, 2005 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk
connection

Hi, today a customer asked how to use asterisk with 12 to 16 FXO ports.
I can use a channel bank with 16 FXO ports and connect the channel bank
with a T1 cable to a T1 card in the Asterisk Server.
Asterisk will then send the calls to the Voip provider over the
internet.

However a 16 fxo port channel bank is about USD 1500 + a t1 card USD 500
+ a USD 1000 computer = 3 thousand us dollars + my installation fees
(life isn't free).

Sounds expensive for such a small install.

Suggestions?


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!) Panama, Republic of Panama
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[Asterisk-Users] T400P vs TE405P

[Darren Wright]

Anyone care to elaborate on the
differences between the T400P and the TE405P?



-Darren








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RE: [Asterisk-Users] How to connect many analog lines to Asterisk?

[Darren Wright]

Wow, first of all, if you have a hundred analog lines, you are doing
yourself a disservice.a 4 T1's would be much much cheaper, and much
easier to manage.

Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk
box.

-Darren
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005 5:20 AM
To: [EMAIL PROTECTED]: [Asterisk-Users] How to
connect many analog lines to Asterisk?

Hello!

If I have more than a hundred analog telephones (analog lines) that need

to be connected to Asterisk PBX, what kind of hardware do I need, and 
where can I buy it?

Thanks in advance!
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[Asterisk-Users] No DID on ZAP

[Darren Wright]

I can't seem to get any ZAP trunks on my TE110P to match any extensions
for incoming DID.


I've even used the exten = _X.,1And it still will not match that.
All I get is:


 -- Starting simple switch on 'Zap/1-1'
  == Starting Zap/1-1 at zap-custom,s,1 failed so falling back to exten
's'
  == Starting Zap/1-1 at zap-custom,s,1 still failed so falling back to
context 'default'

The only think it will match is exten = s,1

And then it works fine...all Callerid is perfect.

Any ideas?

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RE: [Asterisk-Users] Adtran TSU 600

[Darren Wright]

Yup...gonna need a T1 card for the server.  Hope the TSU600 came with
the TDM controller...it should have

-D

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Edwards
Sent: Thursday, August 11, 2005 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Adtran TSU 600

I got a good deal on one of these channel banks loaded with 24 FXS
ports.  I know 24 seems pretty overkill for a home user, but I got this
shipped cheaper than I could have gotten a TDM400P w/ 1 FXS port.  I've
read that these are compatible w/ asterisk, but can they be used w/o a
T1?? (I'm not really sure how * is connected to the channel bank). 

Would I have to have a T100P (whatever the new model is.. T1/E1
selectable.. blah blah) and a T1 xover cable?  (If so, suddenly the deal
just got more expensive)
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RE: [Asterisk-Users] call load balancing

[Darren Wright]

---
An ever better way is get some kind of SLA with guaranteed uptime and 
bandwith, a symetrical link, and do some traffic shaping to ensure that 
VoIP has priority. Part of the point of VoIP is to save money by 
collapsing voice and data networks onto one (presumably robust) network,

so having 2 shabby separate DSL connections kinds of defeats the
purpose.
--


How do you traffic shape incoming packets though  Without your ISP
to provide QoS for downstream voice traffic, quality can still be an
issue


-Darren


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[Asterisk-Users] Press # to continue / Findme

[Darren Wright]

I have implemented a simple findme solution based on DID's.  

In the findme context, after trying each respective number (at s,5 and
s,6), I would like a voice saying The person was not available, press
pound to try the next number. Otherwise, it hangs up after 20 seconds
without dialing the next number.

Any ideas?  Using background dosen;t work, because you hit # and it
hangs up.




[default]
exten = _8134712509,1,Goto(columbia,s|1)
exten = _8134712510,1,Goto(constitution,s|1)



[columbia]
exten = s,1,setvar(GSMNUM=xx)
exten = s,2,setvar(IRINUM=xx)
exten = s,3,setvar(F55NUM=xx)
exten = s,4,macro(findme,${GSMNUM},${IRINUM},${F55NUM})

[constitution]
exten = s,1,setvar(GSMNUM=xxx)
exten = s,2,setvar(IRINUM=xxx)
exten = s,3,setvar(F55NUM=xxx)
exten = s,4,macro(findme,${GSMNUM},${IRINUM},${F55NUM})



[macro-findme]
exten = s,1,Answer
exten = s,2,Wait,2
exten = s,3,BackGround(pls-wait-connect-call)
exten = s,4,Dial(SIP/[EMAIL PROTECTED],20,m)
exten = s,5,Background(gsm)
exten = s,6,Background(silence/5)
exten = s,7,Dial(Zap/1/${ARG2},15,m)
exten = s,8,Background(iridium)
exten = s,9,Background(silence/5)
exten = s,10,Dial(${ARG3}/sip.broadvoice.com,10,m)
exten = s,11,Hangup
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RE: [Asterisk-Users] FXO gateways / Audiocodes MP-108

[Darren Wright]

My experience with an MP-108 was similar.  Incredibly complex to setup,
and very little help from MFR, or even ABPTECH, the main US reseller.
We just couldn't get it working properly.

Ended up with a TE110P with an Adit 600 channel bank, which ROCKS.
Unbelieveably easy to setup.  No echo whatsoever.

-Darren


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[Asterisk-Users] ADIT 600 Expert needed

[Darren Wright]

Doing some funky stuff with an adit 600 that is...above my head, cross
connecting T1 channels, etc.  

Need an Adit 600 expert..paid time.

Ping me off list.

dwright (at) d2-tech (dot) com 

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RE: [Asterisk-Users] 7970 SIP

[Darren Wright]

There is DEFINITELY 7970 SIP firmware out there...maybe Betabut it's
out there.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 03, 2005 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 7970 SIP

Nkm [EMAIL PROTECTED]  :

 On 8/2/05, Darren Wright 
 wrote:
   Can anyone point me to the location of the 7970 SIP image?  I'm
logged

There's no SIP firmware for 7970, only SCCP firmware.

Am I right?

Sergio


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RE: [Asterisk-Users] 7970 SCCP configs?

[Darren Wright]

 Ok I've got SCCP running I have my 7970 firmware files.

Can anyone send an XMLdefault config and an SEP config file?

There are a bunch of sbn files in the package...not sure what needs to
be loaded.

-Darren

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[Asterisk-Users] 7970 SIP

[Darren Wright]

 Can anyone point me to the location of the 7970 SIP image?  I'm logged
in thru my CCO acount with my smartnet contract and cannot find it
anywhere.


I know that a bunch of people have it at this point...how'd they get it?

-Darren

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[Asterisk-Users] Audiocodes MP-108 FXO to Asterisk HELP

[Darren Wright]

 
Does anyone have configs on the MP-108 FXO to asterisk setup?  I'm doing
my best with the limited docs, but having very little success.

Thanks,

-Darren
 
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[Asterisk-Users] 12 FXO ports into Asterisk

[Darren Wright]

I have a client that has 10 POTS lines incoming.  There is no other
option for lines here.

I have 3 options I can see:

1. 3 TDM400 cards
2. An external SIP/FXO gateway
3. A T1 card plus a channel bank.


Does anyone have any thoughts on these 3 or suggestions on keeping the
cost down?

-Darren


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RE: [Asterisk-Users] 12 FXO ports into Asterisk

[Darren Wright]

I agree about the TDM.nuff said.

There looks to be a nice gateway for $1000 or so.  

The problem I see with channel banks is that any of them with FXO cards,
even on ebay are $$$. FXS are a non-issue.

-Darren


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, June 23, 2005 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 12 FXO ports into Asterisk


 I have a client that has 10 POTS lines incoming.  There is no other
 option for lines here.
 
 I have 3 options I can see:
 
 1. 3 TDM400 cards
 2. An external SIP/FXO gateway
 3. A T1 card plus a channel bank.
 
 
 Does anyone have any thoughts on these 3 or suggestions on keeping the
 cost down?

You might have a 4th choice by subscribing to itsp services (DID)
and either call forward the existing numbers to the DID's, or
perm transfer those numbers to the itsp.

Having used the TDM card since it came out, I'd be very hesitant to
use them in this type environment. Very likely to be a high
maintenance/support item for you. They are sort of okay in small soho
environments, but if you dig through the last year's worth of postings,
you'll see lots of quality and audio level issues with the TDM.

The gateway approach will work, but you'll probably spend a fair amount
of up-front time finding one that works well and getting it set up to
be reliable. They are rather expensive as well (comparatively speaking).

The T1 card with channel bank will likely be your least cost approach
after considering setup time, ongoing support costs, etc. Generally
considered the most solid approach with *.



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RE: [Asterisk-Users] 12 FXO ports into Asterisk

[Darren Wright]

I'm looking at $1500 for the bank plus $500 for the T1 for a 10-port FXO
solution.

3 TDM cards are significantly less than that.

Any other ideas?



-Original Message-
From: Garrett Smith [mailto:[EMAIL PROTECTED] 
Sent: Thursday, June 23, 2005 6:52 PM
To: Darren Wright
Subject: RE: [Asterisk-Users] 12 FXO ports into Asterisk

I would go the T1 card plus a channel bank. Rhino channel banks are
great,
plus they are the most economical ones out there.

Thanks,

Garrett Smith
[EMAIL PROTECTED]
716-250-3408 Direct
716-903-9495 Cell


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wright
Sent: Thursday, June 23, 2005 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 12 FXO ports into Asterisk

I have a client that has 10 POTS lines incoming.  There is no other
option for lines here.

I have 3 options I can see:

1. 3 TDM400 cards
2. An external SIP/FXO gateway
3. A T1 card plus a channel bank.


Does anyone have any thoughts on these 3 or suggestions on keeping the
cost down?

-Darren


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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Darren Wright]

You find me a reliable Teir 1 ISP T1 in New Hope, PA
for $300 to $400 





and I'll give you the amount I save over the next quarter. NPA-NXX is 





215-862. Good luck. 









Ive got a Full T1 from a rather
large Mid-Atlantic CLEC for $291. Ive got about  dozen of them from DC
to Trenton, NJ.



-Darren










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RE: [Asterisk-Users] IAX.CC/SixTel

[Darren Wright]

I also ordered from them, and called immediately to follow up.  They got the 
DID done in 5 mines.  But this was in March or so.
 
I noticed that they must be using the same carrier as Voice pulseall the 
numbers were the same.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Thu 5/12/2005 11:44 AM
To: 'Alfredo Manrique'; 'Asterisk Users Mailing List - Non-Commercial 
Discussion'
Subject: RE: [Asterisk-Users] IAX.CC/SixTel



I ordered two weeks ago, they charged my CC, still nothing, trouble tickets
never answered, emails never answered, phone is never answered, money has
not been refunded yet.
-Kerry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alfredo
Manrique
Sent: Thursday, May 12, 2005 6:17 AM
To: BJ Weschke; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX.CC/SixTel

Same thing happend to me. I order a 954-XXX-XXX DID on 04-06-2005 and I'm
still waiting. My order status also says pending.

On 5/11/05, BJ Weschke [EMAIL PROTECTED] wrote:
 I ordered a 973-XXX- and 585-XXX- DID from them on 2/3 and 2/7
 of this year respectively.

 Their customer service portal still lists these orders as pending
 though they told me back when I ordered them that provisioning would
 happen within 1 business day.


 On 5/11/05, Wiley Siler [EMAIL PROTECTED] wrote:
 
 
  Anyone have an opinion about these guys and their recent performance?
 
  I need some local DIDs and they provide for my area code..
 
  Thanks,
  Wiley
 
 
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RE: [Asterisk-Users] Broadvoice Down?

[Darren Wright]

Yup..major broadvoice issues here as well.  I can dial-in, but no dial out. 
 This is the first problem for me in 2 months.
 
-D
 



From: [EMAIL PROTECTED] on behalf of Sean Kennedy
Sent: Mon 4/25/2005 3:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Broadvoice Down?



JD Austin wrote:

 I tried calling Broadvoice support.. on hold for 1/2 hour then it hung
 up on me with a reorder (fast busy), so I tried again.
 Just got through to a rep- they said it's a 'carrier issue' that their
 'partner carrier' was having issues and that it would be up soon.
 Makes me wonder if I should be signing up with their 'partner carrier'
 instead.

 JD


Actually, with all the threads I've seen on the mailing list, I'm weary
of anything having to do with broadvoice.

Personally.  Maybe it's just that they have such a large user base on
linux.  Who knows.

Voicepulse gets my business tho.  :)

Sean
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RE: [Asterisk-Users] Broadvoice gateways!

[Darren Wright]

thru my cavalier T1 I am getting 20ms  ping times to dca.broadvoice.com...I 
switched all my voicepulse and sixtel to BV.
 
 



From: [EMAIL PROTECTED] on behalf of trixter http://www.0xdecafbad.com
Sent: Thu 4/21/2005 4:25 PM
To: Gerard Marcel; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice gateways!



On Thu, 2005-04-21 at 15:35 -0400, Gerard Marcel wrote:
 How many gateways does broadvoice have?  Does anyone know?  I know
 about sip.broadvoice.com.  Are there other ones?

sip.broadvoice.com is a generic placeholder (techincally it points to
proxy.mia.broadvoice.com, but if you follow their directions and
edit /etc/hosts to the fastest gateway then its a placeholder name)

At least:
proxy.mia.broadvoice.com (miami)
proxy.lax.broadvoice.com (la)
proxy.chi.broadvoice.com (chicago)
proxy.dca.broadvoice.com (washington dc - prolly baltimore in reality)

the recommended thing is to to:

#!/bin/sh
for i in mia lax chi dca; do
ping -c 10 -q proxy.${i}.broadvoice.com
done

and use the fastest one to where your server is.

--
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] SPA-841 Call waiting?

[Darren Wright]

I was the one who originally asked the question..upgrading to @home
.7 fixed it for me.  I hadn't dug into the dialplan, but all is well
now.


Too bad it was all for naught.  The SPA-841 speakerphone sucks ROYALLY.

-Darren
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Clark
Sent: Wednesday, March 30, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA-841 Call waiting?

Paul Dugas wrote:
 On Wed, March 30, 2005 4:22 pm, Steve Clark said:
 
I'll go thru the config one more time if it still doesn't work I'll 
then try having it register with asterisk not that I can see where 
that should make a difference.
 
 
 Huh?  Then what are you registering with?  My 841 basically does 
 nothing until after it registers with the server.  Perhaps I'm missing

 something here...
 
There is an option under admin-ext that you can set to allow making
calls with out reg. and ans calls without reg.

But I just changed it to register and have the gotten the same results.
I call form phone a thru asterisk to the phone B and answer it. I then
use  phone C to
  call phone B and I get a sip 486 busy here and asterisk goes into vm.

I have on admin-phone page both ext set to 1. I have admin-ext1
enabled and registering with asterisk. I have admin-ext2 disabled. I
get both lines active and can call out from either one.

Steve
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[Asterisk-Users] SPA-841 Call waiting?

[Darren Wright]

 Has anyone gotten call-waiting to function on the 4 line SPA-841?
I've seen some documents that say it can do it, some say no way.

If yes, can you share configs / SPA-841 settings?

If no, did you work around it?


I can call out just fine on all 4 lines, however, if I am on the line,
another call coming in does not ring the 2nd line...it just goes to busy
/ VM.

-Darren



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