[asterisk-users] Optus PRI via DSL

[mattf]

There was a bit of traffic on this list a while ago regarding OPTUS
multi line that comes in via a DSL box, and I am hoping some of those
people are still hanging around, and solved their problems.

We apparently have a 14B channel service with optus.

I have been trying to configure Asterisk, with a TE110p card with no
luck at this stage.

I have consoled onto the DSL box and have found the following:
The box is an ADTRAN 408582526 SHDSL box
The operation mode is NTU G.703 Unframed (Indep. Clk)


Now when I plug in an Rj-45 from the DSL box to the TE110p I get
nothing, and so I have made an E1 crossover, and I now get a combination
of green lights. One on the back of the TE110p, and the G.703 light on
the DSL modem comes on.

This would suggest to me that the physical link is working fine.

The following comes up at boot:
TE110P: Setting up global serial parameters for E1 FALC V1.2
TE110P: Successfully initialized serial bus for card
Found a Wildcard: Digium Wildcard TE110P T1/E1
Registered tone zone 1 (Australia)
TE110P: Span configured for CCS/HDB3/CRC4


my /etc/zaptel.conf is as follows:
span=1,1,0,ccs,hdb3,crc4
bchan=1-14
dchan=16
unused=15,17-31

loadzone = au
defaultzone = au


When I try and dial our inbound number, I just get a busy signal.

The fact that NTU G.703 is operating in Unframed mode on the DSL
looks to be the problem to me, however I am in over my head a little
with this stuff.

Unfortunately I do not have the management password to login and change
the framing on the HDSL modem.

Would I be correct to say if I use clear channel on the TE110p (ie not
use framing) it would work, and how do I set this on the card?

Anyone have any ideas of where I should go from here?

Thanks in advance

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[asterisk-users] Optus PRI via DSL

[mattf]

There was a bit of traffic on this list a while ago regarding OPTUS
multi line that comes in via a DSL box, and I am hoping some of those
people are still hanging around, and solved their problems.

We apparently have a 14B channel service with optus.

I have been trying to configure Asterisk, with a TE110p card with no
luck at this stage.

I have consoled onto the DSL box and have found the following:
The box is an ADTRAN 408582526 SHDSL box
The operation mode is NTU G.703 Unframed (Indep. Clk)


Now when I plug in an Rj-45 from the DSL box to the TE110p I get
nothing, and so I have made an E1 crossover, and I now get a combination
of green lights. One on the back of the TE110p, and the G.703 light on
the DSL modem comes on.

This would suggest to me that the physical link is working fine.

The following comes up at boot:
TE110P: Setting up global serial parameters for E1 FALC V1.2
TE110P: Successfully initialized serial bus for card
Found a Wildcard: Digium Wildcard TE110P T1/E1
Registered tone zone 1 (Australia)
TE110P: Span configured for CCS/HDB3/CRC4


my /etc/zaptel.conf is as follows:
span=1,1,0,ccs,hdb3,crc4
bchan=1-14
dchan=16
unused=15,17-31

loadzone = au
defaultzone = au


When I try and dial our inbound number, I just get a busy signal.

The fact that NTU G.703 is operating in Unframed mode on the DSL
looks to be the problem to me, however I am in over my head a little
with this stuff.

Unfortunately I do not have the management password to login and change
the framing on the HDSL modem.

Would I be correct to say if I use clear channel on the TE110p (ie not
use framing) it would work, and how do I set this on the card?

Anyone have any ideas of where I should go from here?

Thanks in advance

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RE: [Asterisk-Users] Two questions about Asterisk Call Center

[mattf]

Hello,

routing based on DNIS is dependant on what your telco sends you. Usually on
Robbed-bit T1s(RBS) they will send you ANI and DNIS together separated by
stars like this:
*7275551212*1234* 
(where 7275551212 is the ANI[callerID] and 1234 is the DNIS[last 4 digits of
the number dialed])
In Asterisk this shows up all as the exten and you need *NXXNXX*1234 in
your dialplan.

If you have PRI T1s then you can usually receive both the CallerID and the
full 10-digit number dialed from the carrier and you will get the full
number dialed as the extension, so 8881231234 in your dialplan.

Collecting wrapup codes is another thing. This means you need a database for
the calls coming in and in case of Asterisk that means tinkering with the
code. There are several add-ons that add this functionality to Asterisk and
some of them cost money, just do a search for queues and agents in Asterisk
on google.

Or you could go with a package like Aheeva or VICIDIAL that have GUI
interfaces and allow you a great deal more interoperability with other
systems and the ability for the agent to enter more info.

MATT---



-Original Message-
From: Tielin Xu [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 02, 2005 2:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Two questions about Asterisk Call Center


Hi:

I am new at Asterisk. Does anyone know how to define the call routing based
on DNIS as our conventional ACD to route a call in Asterisk? Second, how do
I collect Wrap-Up code for agents in Asterisk?

Many thanks.

Tielin

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RE: [Asterisk-Users] call center 20 seats

[mattf]

What kind of call center: inbound, outbound or both? 
how many lines per agent will you have? 
what kind of trunks will you be using?
do you need to tie into an existing database? 
do you want screen-pops?

MATT---

-Original Message-
From: Zeeshan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 02, 2005 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] call center 20 seats


hi,
 I am going to open up a call center starting with 5 and expanding to 20 
seats in 3 months. I have decided to use asterisk. I don't think I need 
FXO or any other card from digium.
If you have any document regarding setting up a call center with 
asterisk then please let me know.
What additional things I need to buy except the server (pentium 4 with 
1gb ram).
thanks in advance,
Zeeshan.

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RE: [Asterisk-Users] call center 20 seats

[mattf]

Hello,

You have several choices if you are doing almost all inbound, here's a
summary:
- Native Asterisk Agents and Queues (easy to setup but no screen pops
native. need add-ons for that, some are commercial)
- There are several companies that sell add-ons for Asterisk
queues/agents to extend functionality
- Aheeva Contact Center (all-in-one solution for in/out dialing with many
features, but it costs for install and maintenance)
- VICIDIAL (work with in/out and has screen pops, GPL and free, has many
features but not as well rounded as Aheeva)

As for trunks, I would recommend using IAX2 with something like a GSM or
G729 codec. There are many providers of IAX2 termination all over the world,
you should pick one that best fits your calling patterns and has the service
level you are looking for.
http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers

MATT---

-Original Message-
From: Zeeshan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 02, 2005 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center 20 seats


mattf wrote:

What kind of call center: inbound, outbound or both? 
  

It will be inbound 90%+ as I only need 2 seats for outgoing.

how many lines per agent will you have? 
  

one line per agent.

what kind of trunks will you be using?
  

Don't know yet. I am open for options and basically I don't want it to 
be bandwidth or process hungry.

do you need to tie into an existing database? 
  

No.

do you want screen-pops?

  

YES.

Let me know if you have any more questions.

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RE: [Asterisk-Users] GUI

[mattf]

astGUIclient is not a configuration tool, it is an end-user-interface that
extends the functionality of your phone through a web browser.

We recommend AMP if you need a web-based config utility.

MATT---

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, July 15, 2005 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GUI


Hi,

I was wondering which would be the best GUI to use for Asterisk management?

astGUIclient or AMP?

Thanks.


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RE: [Asterisk-Users] TDMoE and callerID

[mattf]

I don't notice it on my TDMoE that is configured as PRI either. Looks like
you need to post a bug to the tracker.

MATT---


-Original Message-
From: Weezey [mailto:[EMAIL PROTECTED]
Sent: Monday, July 11, 2005 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDMoE and callerID


I've been experimenting with the zaptel TDMoE stuff and I've got it all
working.  Calls go from one asterisk box to the other, with no issues,
except they don't bring the callerID along with them.  I tried the em
signalling from the wiki and I thought maybe that had something to do with
it, so I just changed it to half fxsks and fxoks and that didn't help me
any, I still don't get the callerID of the caller, even if I define it in
the outgoing end of the TDMoE in zapata.conf

So, is there a trick to it or does callerID information just not go across
TDMoE?

Thanks

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[Asterisk-Users] TDMoE bandwidth and load

[mattf]

Hello,

We've just started using TDMoE(local T1s connecting between Asterisk servers
in the same building over the LAN) to connect a few of our high-availability
servers instead of using crossover T1 cables. The 3 servers we have
connected to each other over TDMoE are running just fine and we have no
audio quality issues or bandwidth issues, but I'm considering using TDMoE to
connect 8 other servers to a main server and was wondering if a single
ethernet interface on the Main server can handle the load of 8 dynamic spans
connecting to it from other Asterisk servers.

Does anyone have any experience with using TDMoE to run 8 virtual T1s on a
single Ethernet port?

Thanks,

MATT---
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RE: [Asterisk-Users] Linux Distribution for Asterisk server use

[mattf]

Hello,

I would recommend Slackware mostly for it's streamlined, minimalist approach
and history of stable distro releases. But with that said, the most
important thing is building a custom streamlined Linux kernel no matter what
distro you use. This can save you bootup time as well as speeding up the
running of the machine. One other very important thing is to not install or
run Xwindows or any window environment(Gnome, KDE, ...) because it will
screw up Asterisk on a high-load machine if you have it running.

We have our 10 production Asterisk servers all running Slackware 10.1 with
custom Linux kernels and the high-volume servers each handle over 60,000
calls a day with no problems.

MATT---

-Original Message-
From: TWV [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 03, 2005 2:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Linux Distribution for Asterisk server use


Hello,
 
My question is about which Linux distribution to choose for Asterisk. (/me
holds breath)  OK, hopefully you're still reading, because whatever you were
thinking now, you're thinking wrong! ;)
First of all, I want to make clear that I have read EVERY message and reply
that I could possibly find about this topic, so that includes the dozens of
messages here on the Asterisk mailinglists, on the Digum forum, and even
Google search results!
 
Still, my question was not answered!  Mainly because the same answer always
came back: Use the one you are most comfortable with.  Well, I already
knew that (linux is linux), but it doesn't apply to my situation at all!
 
Let's make things clear and concrete now:
In my professional life, I work as a windows system  network administrator
and as a developer on the .NET platform, and have a long and extensive
experience with telecom and VoIP.  Working for a telecom company, I now have
accepted the challenge to extend our offering with Asterisk.  During the
past 6 months, I have learned as much as I possibly could about the Asterisk
PBX, successfully set up a complete test environment, developed IVR systems,
and now we feel the time is right to put these services into production.
BUT, before we can go live, one important problem remains: as I said, I am a
Windows guy, I have a VERY profound knowledge of Windows and manage almost
20 Windows 2003 Servers that run mission-critical applications on a 365/24/7
basis, and support a large number of Windows applications and Web services
(some of the applications I have developed are used by more than 25000 users
every day!)  Why am I telling this?  Well, because I want to make it clear
that I am perfectly happy with my platform/OS (windows), and have no intent
whatsoever to ever change servers or application platforms to linux (let
alone my workstation).  What's more, I have NEVER come in to contact with
linux/unix before, so I have never worked with ANY distribution.
 
Having explained all this, it should sound logical that I chose the
AsteriskWin32 version for learning Asterisk.  Of course, I realize that we
can't put any production system on AsteriskWin32...  So before we can go
live with Asterisk servers and services, this last issue remains to be
resolved: what Linux distribution should I choose (and learn)?
As I explained, I see Linux merely as a necessary evil (because of my lack
of knowledge) for running Asterisk.  So I'm asking about the best linux
distribution only to put up asterisk servers.  I'm NOT asking for the
easiest one or so (I always enjoy challenges and learning new things), I'm
asking for the best choice to build a carrier grade telecom system, having
to support thousands of users each and every day.  So it must be reliable
and easy to maintain and upgrade.  We are going to use Asterisk in our own
datacenters (supporting nation-wide services), as well as in servers that we
sell to corporations and callcenters for use as an advanced PBX/CTI system).
So it should be clear that I'm not talking about a hobby or home deployment
here.  Our central asterisk systems for example will have to manage DS3 or
(lots of) E1 trunks...
 
If you need more information, I am happy to supply it.  I appreciate your
time and am hoping for some good suggestions and arguments which will lead
me to the correct choice for now and for the years to come.
 
Thanks!
 
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RE: [Asterisk-Users] play message to callee before connect toinco mingcall

[mattf]

You can send both paties to a meetme conference with Manager Redirect. Or if
you are feeling more adventurous you could load the Manager Bridge patch
that I posted to the bugtracker two months ago. It allows bridging of any
two existing channels together through a manager action:
http://bugs.digium.com/view.php?id=4297

MATT---

-Original Message-
From: Roland Zagler [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] play message to callee before connect
toincomingcall


sorry for the misunderstanding, robert!

the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played the caller should be
connected
to the sip phone 100.

the behaviour is just like MoH, but the problem is, that the caller has
to hear a
soundfile from the beginning, and MoH does not do this as the sound
files inside
the directory specified in musiconhold.conf are started when asterisk is
started.

an idea i had was to write an agi script which is called in dialplan
before i issue a
playback command to the caller, which is producing a callfile to call
sip phone 100,
but how can i connect these 2 calls afterwards? is there a possibility
to connect
the answered sip phone 100 to the incoming call (that is still listening
to the playback
command)?

roland

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Saturday, July 02, 2005 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] play message to callee before connect
toincomingcall


On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:

 Thank you, Robert!

 The announcementfile plays well, but at Dial-option m i have to
 specify a MoH class,
 that is something i cannot use (as i wrote in my post).

Noted, which is why I offered option two.


 Background command waits for a user input, but the caller should be
 connected to
 SIP Phone 100 after it has answered and the announcement has been
 played.

But it doesn't REQUIRE input. Background completes when then sound file 
ends. Are you saying you want to move on to announcing the call to the 
callee as soon as it comes in while the caller is listening to the 
soundfile?

I was following your sequential steps in your post, but if you intend 
to fork the process and be doing two things at once, then it's more 
complex.


 Before
 connecting to SIP Phone 100 the caller should hear a soundfile...

 wiki says nothing about an Dial-option to play a soundfile to the 
 caller
 ;-(

Sure it does... BACKGROUND.

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RE: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card near NH??

[mattf]

Hello,

Either Digium or Sangoma can overnight a card to you. As the car drives you
could go to Toronto and pickup a card from Sangoma if you needed if a few
hours before Overnight would deliver it.

There are also a lot of resellers that can overnight to you as well.

MATT---

-Original Message-
From: Tom Rymes [mailto:[EMAIL PROTECTED]
Sent: Friday, July 01, 2005 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
near NH??


I need a Digium or Sangoma T1 card that has at least 2 spans on it
fairly quickly. Does anyone know of a vendor for either of these in NH
or Northern MA?

Please let me know!

Tom



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RE: [Asterisk-Users] Sangoma and quad card hang up problems

[mattf]

Hello,

Need to give some more info here:
- What kind of hardware?
- What distro/version of Linux?
- What version of Asterisk?
- What wanpipe driver version for Sangoma card?
- What firmware version for Sangoma a104 card?
- What are your zaptel/zapata settings for this machine and your other
Asterisk server?
- Have you contacted Sangoma technical support yet?

MATT---


-Original Message-
From: mobilpete [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 29, 2005 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sangoma and quad card hang up problems


need help trying to figure out why calls hang when using multple ports on
Sangoma card.
we have 1 quad card with 3 T1 ports configured, Port1 acts as connection to
teleco (to our T1 PRI)
port 2 connects second system and routes calls to port1
port 3 is Asterisk pbx
calls all go in and out properly but sometimes we get a call hang on when
both sides hangup. this causes all calls to fail until we restart * with
restart now cmd. Which taks approx 10-20 seconds to complete.
see log files form a call with show channels at bottom.


This was an incoming call that was answered and completed   
 
== Spawn extension (pri-g3, 17087492476, 1) exited non-zero on 'Zap/47-1'
-- Hungup 'Zap/47-1'
-- Executing Dial(Zap/1-1, Zap/G3/7732997763|120) in new stack
-- Called G3/7732997763
-- Accepting call from '7082975266' to '7732997763' on channel 0/1, span
1
-- Zap/47-1 is ringing
-- Zap/47-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/47-1
-- Hungup 'Zap/47-1'
  == Spawn extension (default, 7732997763, 1) exited non-zero on 'Zap/1-1'
-- Call accepted by 64.4.200.98 (format unknown)
-- Channel 0/1, span 1 got hangup
-- Channel 0/1, span 1 got hangup
 
Why 2 of the items in red?
 
Notice we got both hang up request but below is what show channels states -
this is after we both hung up on call.
 
NPS-816-Bwyn-Sw1*CLI show channels
Channel  (ContextExtensionPri )   State Appl. Data
Zap/1-1  (default7732997763   1   )  Up (None)(None)
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[Asterisk-Users] New astGUIclient version released 1.1.4

[mattf]

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.4

http://astguiclient.sf.net/

The client suite runs on Windows, UNIX and Mac, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or Zap
phones and Zaptel, IAX or SIP trunks.

For this revision, in addition to adapting the code to the 'Local/' channel
changes made in Asterisk release 1.0.8 and CVS_HEAD, we have added the
ability to use SIP trunks for outbound and inbound lines to the package, as
well as adding an autodial IVR survey example script to VICIDIAL.

We have also created a graph showing possible hardware configurations for
systems running astGUIclient to better  understand where astGUIclient fits
in and what it needs to run: 
http://astguiclient.sf.net/images/sample_physical_setup.gif

Let me know what you think.

Thanks,

MATT---
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RE: [Asterisk-Users] Asterisk server with remote monitoring capab ilities

[mattf]

We have two Baytech RPC3 remote power switches(8 outlets each), they are
great, you can telnet into them and reset ports as needed. I even setup one
of them to be controlled by an AGI script on our Asterisk servers to cycle
power over the phone. Saved countless hours of driving. APC makes them too
although they are more expensive.

MATT---

-Original Message-
From: beonice [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 23, 2005 7:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk server with remote monitoring
capabilities


Hello, all.

I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.) 

I've been looking at the fancy xeon-based systems
listed on ebay for a couple of hundred dollars, in the
hope that some of them have remote reboot
capabilities, but most of the sellers don't mention
this ability, and by the time I send out email, the
item is already taken anyway. :)

So, to cut the long story short, has anyone used one
of these server-class machines with remote reboot
capability, and does it really help? Are there any
particular configurations to stay away from? 

The wiki doesn't talk specifically about issues
regarding dual-CPU machines, but in following the chat
here on asterisk-users, it seems there are definitely
issues there ... can anyone elaborate? I don't want to
spend money on a fancy system that turns out to be
useless for my purposes.

Thanks for any insight!

Cheers,
Maya




 
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RE: [Asterisk-Users] MCI vs. XO/Allegiance

[mattf]

Let me throw another complaint against XO on the table. They actually shut
off the wrong T1 and they transferred all of the DIDs to the T1 they shut
off! how screwed up is that? We are now about 2 years later and their
billing department still calls us every month for nonpayment of the T1 that
they turned off.

We also have 6 T1s through MCI, all long distance. They have been much
better to deal with.

MATT---

-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Monday, June 13, 2005 7:34 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] MCI vs. XO/Allegiance


Well, the fact that two negatives for XO and a positive for MCI all came
at once says a lot to me.

Interestingly enough their SLA reads...

*   24/7/365 Network Monitoring and Service. If for some reason your
network is having problems, the chances are XO will know about it before
you do and respond before any problems become critical.

Thanks!
Wiley
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Traue, Jr.
Sent: Monday, June 13, 2005 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance

I'm using an XO pri, and as long as you never change anything on the pri
XO's not bad.  Our experience is that if you chance anything on the PRI
configuration they'll screw it up somehow (YMMV).  One thing we have
learned is that XO doesn't monitor our voice circuits, so if one of our
PRI's goes down, we have to notify them almost immediately or they
decommission it so the alert goes away.

Paul

Wiley Siler wrote:
 Hello All,
 
 Anyone out there using ISDN PRI from either MCI or XO/Allegiance? 
 Gotta make the choice today and the difference per month is only about

 $25 in favor of MCI.
 
 Billing is pretty much the same between the two so I have pretty much
no 
 point of reference on which to choose.
 Any thoughts from anyone experienced with these two compnies would be 
 greatly appreciated!
 
 Thanks,
 Wiley
 
 


 
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RE: [Asterisk-Users] Opinions of Sphinx?

[mattf]

Hello,

Very true, we actually convert the 8k 8bit wav files to 16 bit files and
normalize the audio through SoX for sphinx.  We spent a couple weeks
developing a custom limited vocabulary and we do the recognition in 2
phases, the first phase is the quick and dirty, we then analyze the results
of that and if our internal score is high enough to recognize the words we
finalize that recording(about 90%), if not we put the non-recognized ones in
a second batch to be analyzed  at a higher level of analysis by sphinx. we
then analyze and score those(about 50% finalize). so we are left with about
95% of our total recordings that finalize correctly with 5% that need to be
manually listened to for confirmation. This was a rather complicated process
to build with many stumbles along the way. The hardest part is the building
of the vocabulary and the tuning of the analysis of the conversions. 

Here's the batch launching script, although that is a very small part of the
whole that you need for this all to work:
http://astguiclient.sourceforge.net/experimental_code/sphinx2_pltest.pl


MATT---


-Original Message-
From: Race Vanderdecken [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 11, 2005 9:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Opinions of Sphinx?


Curious, which codec are you using with Sphinx?

The smaller the bandwidth, generally, the harder it is to do
recognition.

Sphinx4 is a tool built on JAVA (one moment while I clear my throat and
spit to get the coffee taste out of my mouth.)

Write me offline; I am curious about doing batch reco for several
projects, if you don't want to answer here.

Race the tyrant Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Saturday, June 11, 2005 9:04 PM
To: 'Brian Roy'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Opinions of Sphinx?

We use batch sphinx to analyze recordings at night. We attempted
real-time
sphinx, but it is way too slow and resource-intensive to use for
realtime on
Asterisk with more than a couple lines at once(and that's at the poor
quality settings). We have not tried sphinx4, but I wouldn't imagine
that it
would be that much improved in speed over version 3.

MATT---


-Original Message-
From: Brian Roy [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 11, 2005 6:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Opinions of Sphinx?


On 5/31/05, Alistair Cunningham [EMAIL PROTECTED] wrote:

 
 Has anyone tried Asterisk and Sphinx (bonus points if in a production
 environment)? If so, what's your opinion on quality of recognition,
 stability, resource usage, etc?

Alistair,

Well, it's been a couple weeks and no answers on the list. That isn't
encouraging, but I'm hoping to accomplish some of the same thing. Have
you made any progress on your own with this? Let me(us) know...

Thanks,

-Brian
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RE: [Asterisk-Users] Voice quality of Softphones vs. IP Phones an d Gateways.

[mattf]

In our experience, the total cost of softphones(money, reduced sound quality
and lower reliability) in a large call center environment is actually
greater over time than the cost of a channelbank and cheap analog
headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2
kinds of SIP analog adapters and we've tried channelbanks over the last 3
years. Right now we are half done with our conversion at all of our
in/outbound telemarketing rooms to channelbanks. The first 2 we installed a
year ago have never gone down. which is a much better track record than any
of the other VOIP devices we used.

I will note however that the second most cost-effective and reliable
solution was Sipura SIP Analog adapters, partially because they use cheap
analog phones and you can hide them under a desk where they will not get
ruined when an agent spills their half gallon of Mountain Dew all over.

MATT---


-Original Message-
From: Cenk Yabas [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 11, 2005 10:03 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice quality of Softphones vs. IP Phones and
Gateways.


I've tried almost any softphone available on the market with many different
PC, soundcard, headphones combinations.
None of them prooved production reliable in a call center environment. 
I've also tested many IP Phones and Gateways. Even the cheapest one supplies
much better quality. 
Is this a fact or am I missing a point. 
I would certanly prefer a softphone because of cost and simplicity in CTI
applications.
Cenk.
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RE: [Asterisk-Users] Opinions of Sphinx?

[mattf]

We use batch sphinx to analyze recordings at night. We attempted real-time
sphinx, but it is way too slow and resource-intensive to use for realtime on
Asterisk with more than a couple lines at once(and that's at the poor
quality settings). We have not tried sphinx4, but I wouldn't imagine that it
would be that much improved in speed over version 3.

MATT---


-Original Message-
From: Brian Roy [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 11, 2005 6:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Opinions of Sphinx?


On 5/31/05, Alistair Cunningham [EMAIL PROTECTED] wrote:

 
 Has anyone tried Asterisk and Sphinx (bonus points if in a production
 environment)? If so, what's your opinion on quality of recognition,
 stability, resource usage, etc?

Alistair,

Well, it's been a couple weeks and no answers on the list. That isn't
encouraging, but I'm hoping to accomplish some of the same thing. Have
you made any progress on your own with this? Let me(us) know...

Thanks,

-Brian
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RE: [Asterisk-Users] astGUIclient installation problem

[mattf]

Hello,

This issue was just handled Monday on the astguiclient-users list:
http://sourceforge.net/mailarchive/forum.php?thread_id=7448401forum_id=4358
6

You just need to use OLD_PASSWORD in the SET PASSWORD for your mysql server
to get the auth method for that account back to the pre 4.1.12 version
default method of login authentication.

Also, consider joining the astguiclient-users list, a lot of tweeks and
fixes come up on there that don't make it into the documentation right away.

MATT---


-Original Message-
From: kritikus Araklidas [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 09, 2005 12:11 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] astGUIclient installation problem


Hi everyone:

I try to install astGUIclient for my call center. I'm interesting to put in 
work the monitoring client, i follow step by step the installation from 
scratch but when i try to run the application from my Windows XP 
astGUIclient i got the follow error:

Client does not support authentication protocol requested by server; 
consider up
grading MySQL client at astGUIclient_1.1.0.pl line 4704

Any idea will be appreciated.

Regards.

Kritikus.

_
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[Asterisk-Users] Digium vs. Sangoma: Performance

[mattf]

Hello,

Several people asked to get a hold of the stats I used to determine that the
Sangoma T1/E1 boards performed better in our real-world tests than the
Digium boards. We've decided to post our results after confirming them over
the past month of operations. 

Here is a link to the last two reviews I wrote about Sangoma with Asterisk:
http://astguiclient.sourceforge.net/Sangoma_experience.txt

Here is a summary of the production environment that these servers were
evaluated in and the hardware used.

HARDWARE:
Each server was constructed in house with the following components:
- 4U rackmount server chassis
- Asus P5AD2 motherboard
- Intel P4 3.2GHz Prescott processor socket 775 with 1 MB L2 cache
- 2GB Kingston DDR2-533 RAM
- 3 x 160GB Maxtor DiamondMax Plus 9 SATA 7200RPM hard drives
- Enermax 550W power supply

SOFTWARE:
- Slackware 10.1 with custom kernel 2.4.29
- Asterisk 1.0.6
   the configurations in zaptel.conf, zapata.conf and extensions.conf were
the same
- astGUIclient suite
Note: Sangoma machine uses wanpipe-beta8a-2.3.3 drivers

The cards compared are the Digium TE405P in T1 mode and the Sangoma A104.

Each server is setup the same with 3 telco T1s connected(one of them
connected on the other end to another Asterisk server) and a channelbank
connected to the fourth T1 port.

The live application used to evaluate performance and gather the stats is
the astGUIclient package with VICIDIAL for outbound dialing.
http://astguiclient.sourceforge.net/

Here are links to screenshots of our performance monitoring web pages for
each server:
http://astguiclient.sourceforge.net/VDreports/performance_Digium_TE405P.gif
http://astguiclient.sourceforge.net/VDreports/performance_Sangoma_A104.gif


The data was gathered every 5 seconds from both servers.

You can see that even though the Sangoma server ran at slightly higher call
capacity(11% more calls with 4% more talk time), that it still had a
significantly lower average load(27% lower) and peak load(56% lower). The
periodic tests that we have done over the last month confirmed these
results, that all things being equal, the Sangoma server performed 30-50%
more efficiently at doing the same job.

I am currently awaiting a beta test version of the Digium TE406P card(with
the echo-canceller daughterboard) and I am awaiting word from Sangoma on
testing their newly announced A104d card(with on-board echo-canceller) to
see how they both fare in our real-world tests.

MATT---
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RE: [Asterisk-Users] Gnudialer

[mattf]

Hello,

I'm the lead developer of astGUIclient(with VICIDIAL) and I tried GnuDialer
a little while ago. It is different in several ways from how VICIDIAL
operates:
- Gnudialer is partially compiled into Asterisk and uses Asterisk agents
while VICIDIAL operates entirely on top of a stock Asterisk installation and
uses it's own agent/queue system.
- Gnudialer can use any kind of trunk(I think, we only tested it with Zap)
while VICIDIAL is limited to IAX2 or Zap trunks
  NOTE: It is much more efficient to use IAX2 trunks than SIP trunks
especially if you have many concurrent connections to the same provider,
that's one major reason to use IAX2 trunks instead of SIP trunks.
- VICIDIAL is more involved to get installed although there is a very large
and comprehensive installation documentation section while Gnudialer is
easier to install if you know what you're doing.
- VICIDIAL allows multiple campaigns across multiple servers meaning you can
have 8 Asterisk servers with 150 agents all dialing the same campaign at the
same time.
- VICIDIAL allows for several options not available to Gnudialer like
unlimited campaign-specific disposition codes, a custom dial-timeout for
outbound calls, non-connected remote-agent capacity, easily configurable
3-way calling/DTMF macros/blind-transfer, fronter to closer capability,
agent click-to-record and easy manager call monitoring and barge-in.
- VICIDIAL has very detailed web-based administration with real-time agent,
campaign and performance reports.

The documention for Gnudialer is not very developed, but it has only been
around for 5 months so far and I'm sure that the documentation will grow as
it grows and has a larger user-base.

VICIDIAL does require a perl/TK app to run on the agent computer to
facilitate the display of customer information and to allow for
call-termination/recording/park/etc... We are currently developing a
real-time web-only VICIDIAL interface(much like the astGUIclient web-app
that we released last week) that should be ready by the end of August. We
also plan on having SIP-trunk compatibility by the end of August.

If one of the Gnudialer developers is monitoring this I would love to learn
about some more of the new features coming to Gnudialer.

Hope this helps,

MATT---


-Original Message-
From: Jesus Mogollon [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 07, 2005 7:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Gnudialer


Hi there!


Is there anyone out there using gnudialer? I tried vicidial but couldn't get
it to work (does vicidial support SIP trunks anyways?). Gnudialer seems to
be simpler, though their web interface needs a little work (version 2.0
seems like a step in the right direction but it isn't out yet). How do you
register an agent? The documentation is lacking...

Jesus Mogollon
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[Asterisk-Users] New astGUIclient version released 1.1.1

[mattf]

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.1

http://astguiclient.sf.net/

Screen shots: http://astguiclient.sourceforge.net/screenshots.html

The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or Zap
phones and Zaptel or IAX trunks.

In addition to many bug fixes and new features, we've added a web-based
version of the astGUIclient that only requires a web browser to use and
allows you to extend the functionality of your phone wherever you are.

Here are some of the features of the astGUIclient web client:
- Grabs live call info from a DB updated every second
- Displays live status of users phones and Zap/IAX/SIP/Local channels
- Allows calls to be placed from GUI and directed to phone
- Allows intrasystem calls at the click of a button
- Allows call recording at the click of a button
- Allows conference calling through GUI
- Administrative Hangup of any live Zap/IAX/SIP/Local channel
- Administrative Hijack of any live Zap/IAX/SIP/Local channel
- Administrative switch user function
- Call Parking sends calls to park ext and then redirects to phone ext
- CallerID popup with links to open custom web pages
- Voicemail display and button to go right to check voicemail
- Allows Blind transfers of calls to specific voicemail boxes
- Allows Blind transfers of calls to intrasystem extensions
- Allows Blind transfers of calls to external numbers
- Send to Voicemail directly from the inbound call popup window without
answering
- All client phone connections are shown not just the first
- Allows transfers to conferences
- Call parking with callerID 


To upgrade from 1.1.0 you need to update all of the server apps and web
pages and run the update sql script.

Let me know what you think.

Thanks,

MATT---
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RE: [Asterisk-Users] choice of processors

[mattf]

Need to provide a little more info:

What's the bus speed?
What kind of motherboard would you use with each?
What kind of RAM at what speed?
What cache size are on the CPUs?

Also, what price are these as equals? I've seen two Xeon 2.8GHz 800MHz
processors for about US$450 and a single P4 at the same price would be a
3.6GHz 2MB cache at several resellers. which resellers do you buy through or
are you buying a prebuilt system?

MATT---



-Original Message-
From: Steven Langley [mailto:[EMAIL PROTECTED]
Sent: Monday, May 30, 2005 11:03 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] choice of processors


Hi there

I am moving into a production environment. I will mostly be using Meetme,
with Ztdummy for timing. I have a question on which of 2 processor setups is
favourable.

I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4
3.06GHz Processor. These will cost me exactly the same amount.

Would one of these processor setups be favourable, both in terms of
performance and running Asterisk?

Many thanks

Steven


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RE: [Asterisk-Users] CRM integration (was RE: CallerID)

[mattf]

Hello,

We use astGUIclient, it does have server side apps that have to be installed
on your Asterisk server, but it does have callerID popups that allow you to
search a customizable web page when a call comes in. We are also releasing a
new version of the astGUIclient app next week that is entirely web-based and
easier to configure the client side.
http://astguiclient.sf.net/

MATT---

-Original Message-
From: Iassen Hristov [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 25, 2005 10:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CRM integration (was RE: CallerID)


I am also very interested in CRM integration. Anything I can do to help?

One thing I don't understand is how is the browser being launched on the
person's PC. Or is it not launched automatically?

Anyone know of a simple app running on the desktop to do this? I looked
into IPSwithcBoard and it appears like it should be able to do the job,
but: 
a) It is pretty heavy - it does a lot of other things that are not
necessary in this scenario
b) it did not work for me. It gives me a bunch of exceptions when
connecting to my Asterisk box.

---
Iassen Hristov

 --
 
 Message: 3
 Date: Tue, 24 May 2005 22:03:48 -0500
 From: Anton Krall [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] CallerID
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=us-ascii
 
 Hi Michiel.
 
 Could We do that? Could you help me disect your apis so I can better
 understand what you are doing? I code in php so mustly what We need to do
 is understand what you are doing to your DB, etc.
 
 Let me know what to do.
 
 Thx! And I sure think this is going to be a great addon to the list for
 everybody looking to use php and integrate CTI into apps like SugarCRM,
 salesforce.com, etc. 
 
| -Original Message-
| From: [EMAIL PROTECTED] 
| [mailto:[EMAIL PROTECTED] On Behalf Of 
| Michiel van Baak
| Sent: Martes, 24 de Mayo de 2005 04:45 p.m.
| To: asterisk-users@lists.digium.com
| Subject: Re: [Asterisk-Users] CallerID
| 
| On 23:44, Tue 24 May 05, Anton Krall wrote:
| I like you scenario Michiel!
| 
| Ok, I have sugarcrm installed on the same box as 
| asterisk...What do I 
| need to do in order to, for example, start my softphone or hardphone 
| and when an incoming calls comes in, make a window popup (web window 
| that is) and show the contact window for that particular 
| callerid on sugarcrm..
| 
| Any ideas? 
| 
| Anton,
| 
| We have this setup working in our webbased php app.
| The process is like this:
| When a call comes in, we launch a php agi script. This agi 
| script does a SQL query in our contacts database to see if one 
| of our customers has a phone number record that matches 
| ${CALLERID}. If so, it will put this customers database id in 
| a temporary table. In our application we have a php script 
| that turns this table into a xml file, and that xml file is 
| read by a little javascript thing in our php webbased app. 
| 
| As you see it doesn't matter what phone/software you have on 
| the client, all is done in asterisk dialplan and the webbased 
| app. I think it should be possible to set the calleridname on 
| the asterisk channel so the phone will show it too, but we 
| didn't care about that cause some of the phones are budgetones 
| (yeah yeah I know, they are getting replaced by GXP-2000's 
| next week) and those wont show it correctly
| 
| Have another look at the files in the tgz file on my website. 
| If you want we can have a look at it together and see how we 
| can get this to work with sugarCRM.
| I'm not familiar with sugarCRM but I can have a look at it 
| with you if you want.
| If it is possible in sugarCRM to get a customers overview 
| screen using an url with parameters it should be very easy to 
| modify our scripts so it works with sugarCRM. If it works with 
| html post data it will need some work, but a day should be 
| enuf to fix that I think.
| 
| Right now we are not generating alerts of calls with a 
| callerid that does not match one of our customers, but it can 
| be done so it will just show you the phone nr and not the 
| customer name. 
| 
| If you want to contact me off-list that's ok.
| But I think this is interesting enough for the list so other 
| ppl and vendors can watch and start working on their projects ;)
| 
| Sorry for the long story, but this is an issue that, in my 
| opinion, really makes asterisk a HUGE added value product for 
| business life.
| 
| Cheers,
| 
| 
| | -Original Message-
| | From: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] On Behalf 
| Of Michiel 
| | van Baak
| | Sent: Martes, 24 de Mayo de 2005 02:14 p.m.
| | To: asterisk-users@lists.digium.com
| | Subject: Re: [Asterisk-Users] CallerID
| | 
| | On 11:18, Tue 24 May 05, Anton Krall wrote:
| | Anyway that works would be nice :) What I would 

RE: [Asterisk-Users] CRM integration (was RE: CallerID)

[mattf]

The method we use for web popups on incoming calls in the astGUIclient
client app that we are working on for release next week is to use
AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't
require any META refreshes. We've been using this internally for the last
month and it works great.

MATT---


-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 25, 2005 10:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID)


As one of the members said. The idea behind this is not to have to install
any app on the computer but base everything on web apps. Im still trying to
find out how to mix everything together but it's a mixture of Asterisk
Manager usage and some PHP coding for example.

You can probably just open a web page, and it will show you the callerid of
whos calling you, provided you first told the web page your extension number
or technology, like SIP/myextension so it can check the manager and see when
a Dial is executed and extract the callerid.

After that, it can probable just open a new browser automatically and insert
the callerid or whatever into a URL for your CRM.

It doesn't seem to be complicated but for example, the things that bother me
are refreshes, I don't want to use meta refreshes for this monitoring
webpage every X seconds, rather, use something more realtime... Any ideas? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Iassen Hristov
|Sent: Miércoles, 25 de Mayo de 2005 09:15 a.m.
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] CRM integration (was RE: CallerID)
|
|I am also very interested in CRM integration. Anything I can 
|do to help?
|
|One thing I don't understand is how is the browser being 
|launched on the person's PC. Or is it not launched automatically?
|
|Anyone know of a simple app running on the desktop to do this? 
|I looked into IPSwithcBoard and it appears like it should be 
|able to do the job,
|but: 
|a) It is pretty heavy - it does a lot of other things that are 
|not necessary in this scenario
|b) it did not work for me. It gives me a bunch of exceptions 
|when connecting to my Asterisk box.
|
|---
|Iassen Hristov
|
| --
| 
| Message: 3
| Date: Tue, 24 May 2005 22:03:48 -0500
| From: Anton Krall [EMAIL PROTECTED]
| Subject: RE: [Asterisk-Users] CallerID
| To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|  asterisk-users@lists.digium.com
| Message-ID: [EMAIL PROTECTED]
| Content-Type: text/plain;charset=us-ascii
| 
| Hi Michiel.
| 
| Could We do that? Could you help me disect your apis so I can better 
| understand what you are doing? I code in php so mustly what 
|We need to 
| do is understand what you are doing to your DB, etc.
| 
| Let me know what to do.
| 
| Thx! And I sure think this is going to be a great addon to the list 
| for everybody looking to use php and integrate CTI into apps like 
| SugarCRM, salesforce.com, etc.
| 
|| -Original Message-
|| From: [EMAIL PROTECTED]
|| [mailto:[EMAIL PROTECTED] On Behalf 
|Of Michiel 
|| van Baak
|| Sent: Martes, 24 de Mayo de 2005 04:45 p.m.
|| To: asterisk-users@lists.digium.com
|| Subject: Re: [Asterisk-Users] CallerID
|| 
|| On 23:44, Tue 24 May 05, Anton Krall wrote:
|| I like you scenario Michiel!
|| 
|| Ok, I have sugarcrm installed on the same box as
|| asterisk...What do I
|| need to do in order to, for example, start my softphone or 
|hardphone 
|| and when an incoming calls comes in, make a window popup 
|(web window 
|| that is) and show the contact window for that particular
|| callerid on sugarcrm..
|| 
|| Any ideas? 
|| 
|| Anton,
|| 
|| We have this setup working in our webbased php app.
|| The process is like this:
|| When a call comes in, we launch a php agi script. This agi script 
|| does a SQL query in our contacts database to see if one of our 
|| customers has a phone number record that matches 
|${CALLERID}. If so, 
|| it will put this customers database id in a temporary table. In our 
|| application we have a php script that turns this table into a xml 
|| file, and that xml file is read by a little javascript thing in our 
|| php webbased app.
|| 
|| As you see it doesn't matter what phone/software you have on the 
|| client, all is done in asterisk dialplan and the webbased app. I 
|| think it should be possible to set the calleridname on the asterisk 
|| channel so the phone will show it too, but we didn't care 
|about that 
|| cause some of the phones are budgetones (yeah yeah I know, they are 
|| getting replaced by GXP-2000's next week) and those wont show it 
|| correctly
|| 
|| Have another look at the files in the tgz file on my website. 
|| If you want we can have a look at it together and see how 
|we can get 
|| this to work with sugarCRM.
|| I'm not familiar with sugarCRM but I can have a look at it with you 
|| 

RE: [Asterisk-Users] What does Asterisk need in the way of a GUI?

[mattf]

Are you talking about an Asterisk configuration GUI that would modify
Asterisk settings or an end-user GUI that would compliment a regular user's
phone?

MATT---

-Original Message-
From: Mitchel Constantin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 25, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] What does Asterisk need in the way of a GUI?


We are two programmers who are passionate for Asterisk and we will be
dedicating the next three months towards programming for Asterisk and
would like to get some input from everyone on what they feel Asterisk
is lacking or needs based on what is not currently a part of it or
available through third parties. Hopefully, by asking up front we
won't be wasting our time on something nobody wants or needs.

Specifically I am asking in the way of GUI's (web-based or not), not
in backend programming as Mark and others have that well under
control!

Thank you for your suggestions,
Mitchel  Tom
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RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur.  You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
to set this up and I think you will probably have to go through some
trial-and-error before you find the perfect system layout for your
operations.

I would first try setting up machines that would just have the T1s on them
and take the calls in(or out) and record them. Then have those
connect(through IAX or T1 crossover) to the servers that have your queues
and phones set up on them. You will also need some really big archiving
mechanism if you want to keep those recordings around to reference in the
future. audio recordings can take up a lot of space if you need to keep them
for 3 years like we do.

SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they
can hit the PCI-bus bottleneck and have issues. You may see the
ast_channel_walk_locked warning in Asterisk when this happens.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues


We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come in on toll
free numbers via T1's, and there are 3 of them.  I want to use a media
gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
gateway), and use asterisk as the PBX having all incoming and outgoing
channels as SIP.  Almost all dialplans will be using Queues, and there will
likely be no more than 10 queues, with (currently) about 80 incoming
toll-free numbers.  There are approximatley 30 agents, but as of right now
there are no more than 15 agents logged in at a time.  We need to be able

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

We have several different setups, but on a couple servers we are doing upto
50 concurrent conversations of recording. We ran into the 50-60 recording
ceiling about a year ago and it's mostly the hard drive that limits it to
that number, really it's a lot if you think about it, Asterisk is having the
hard drive write 100-120 audio files(-in and -out for each conversation)
several times a second. It is also important to note that we mix them with
sox after hours to reduce load on the system and load on the drives.
Although this does mean that the recordings are not available until the next
day. We also have setup 2 systems to copy the in and out files off to
another machine to be mixed more closely to realtime so that is an
alternative.

MATT---


-Original Message-
From: Ilan Rabinovitch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Inbound call center - reliability \
scalabil ity with queues


Matt,

Are you doing any call recording / monitoring?  What percentage?  

Ilan

On 5/23/05, mattf [EMAIL PROTECTED] wrote:
 For an inbound call center with 4 T1s and 30-50 agents on you would do
just
 fine with a single, one-processor machine. We have handled more than this
on
 a single P4 server although we use astGUIclient instead of Asterisk
queues,
 but the load is very similar. I would recommend a Sangoma Quad T1 card
 because they are about 30% more efficient than Digium T1 cards.
 
 When you say that you need to scale to 100s of consecutive calls, is that
 closer to 200 or 900? and what timeframe is that planned for?
 
 We have a distributed in/outbound call center environment across 4
 geographic locations with over 20 T1s connected so it is possible for
 Asterisk to handle over 1000 consecutive calls across the system if you
 design it right. One of the reasons we don't use Asterisk queues, other
than
 the difficulty in customizing the code to work with the ManagerAPI and
 client apps, is that it was hard to scale across multiple servers. That's
 why we use the astGUIclient suite which is more customizable and scalable
 across multiple servers, although (and it pains me to say this because we
 developed it) it is not as easy to install and setup than just creating an
 Asterisk queue.
 
 We use a combination of SIP, IAX and Zap client phones depending on the
 system and the user and yes 711 is always best to use when you can. And if
 you have many remote phones using a codec like GSM it may actually be
better
 to have a dedicated machine doing nothing but the transcoding from
GSM-711
 and then just using IAX or a crossover Zap T1 to the inbound server to
 reduce processor load.
 
 In any case it is always advisable to have a backup server that is fully
 ready to jump in production with a minimum of reconfiguration.
 
 A couple more questions, will you do much recording? and what kind of
disks
 do you plan on using?
 
 Hope this helps,
 
 MATT---
 
 -Original Message-
 From: Warren Smith [mailto:[EMAIL PROTECTED]
 Sent: Monday, May 23, 2005 8:12 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
 with queues
 
 
 We are wanting to move off of our legacy inter-tel phoneswitch and move to
 VoIP and asterisk.  We are looking for a new PBX because the inter-tel
 switch is too difficult to integrate our existing (and new) software into.
 
 We are a technical support center.  All our calls currently come in on
toll
 free numbers via T1's, and there are 3 of them.  I want to use a media
 gateway to convert the T1's into SIP VOIP (I want reliable hardware for
the
 gateway), and use asterisk as the PBX having all incoming and outgoing
 channels as SIP.  Almost all dialplans will be using Queues, and there
will
 likely be no more than 10 queues, with (currently) about 80 incoming
 toll-free numbers.  There are approximatley 30 agents, but as of right now
 there are no more than 15 agents logged in at a time.  We need to be able
to
 support 60-70 simultaneous calls initially and we have to be able to do
this
 reliably.  We also need to be able to scale into the 100's of simultanous
 calls range.
 
 What would be the best option, to have 2 powerful machines (dual
 powersupply, ) with one as a hotswappable backup or have multiple machines
 with a sort of load balancer setup?  Having multiple machines could
possibly
 cost less, but I'm not sure how the queues and agents would be managed
 across multiple machines.  I.e. how would the agent 'login' to each
asterisk
 machine so that the calls could be handed to it, and how would the calls
get
 handed to an available agent by 4 seperate asterisk machines?  I've read
 through the wiki, but I'm not sure how much overhead queues would put into
 the system.  I want to have all the codecs the same, so the asterisk
 machines doesn't do any transcoding, and have all channels as SIP

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

Well, that really changes things then. I'm not really sure what to tell you
because we've never done it that way. The ciscos are limited in how you can
have them send calls to different servers based upon specific parameters so
you will be limited there somewhat. Is there a specific reason you're not
going to use Asterisk servers for the T1-SIP conversion? 

It would allow you to do the recording up front and give you more control
over call handling, and I can't imagine that you can get a new 3 x T1 Cisco
VOIP machine for less than $5000 like you can with Asterisk.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 2:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


The asterisk machines will not have anything to do with the T1's, when they
receive the call it will be SIP VOIP.  There will be media gateways (i.e.
cisco media gateways) to change all T1 signals to VOIP before it reaches the
PBX.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Tuesday, May 24, 2005 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur.  You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
to set this up and I think you will probably have to go through some
trial-and-error before you find the perfect system layout for your
operations.

I would first try setting up machines that would just have the T1s on them
and take the calls in(or out) and record them. Then have those
connect(through IAX or T1 crossover) to the servers that have your queues
and phones set up on them. You will also need some really big archiving
mechanism if you want to keep those recordings around to reference in the
future. audio recordings can take up a lot of space if you need to keep them
for 3 years like we do.

SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they
can hit the PCI-bus bottleneck and have issues. You may see the
ast_channel_walk_locked warning in Asterisk when this happens.

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 24, 2005 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues


I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap

RE: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues

[mattf]

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM-711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-Original Message-
From: Warren Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues


We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come in on toll
free numbers via T1's, and there are 3 of them.  I want to use a media
gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
gateway), and use asterisk as the PBX having all incoming and outgoing
channels as SIP.  Almost all dialplans will be using Queues, and there will
likely be no more than 10 queues, with (currently) about 80 incoming
toll-free numbers.  There are approximatley 30 agents, but as of right now
there are no more than 15 agents logged in at a time.  We need to be able to
support 60-70 simultaneous calls initially and we have to be able to do this
reliably.  We also need to be able to scale into the 100's of simultanous
calls range.

What would be the best option, to have 2 powerful machines (dual
powersupply, ) with one as a hotswappable backup or have multiple machines
with a sort of load balancer setup?  Having multiple machines could possibly
cost less, but I'm not sure how the queues and agents would be managed
across multiple machines.  I.e. how would the agent 'login' to each asterisk
machine so that the calls could be handed to it, and how would the calls get
handed to an available agent by 4 seperate asterisk machines?  I've read
through the wiki, but I'm not sure how much overhead queues would put into
the system.  I want to have all the codecs the same, so the asterisk
machines doesn't do any transcoding, and have all channels as SIP.  There
will be music on hold.  Would a dual 2.8 ghz xeon in this config be able to
handle 80 simultaneous incoming calls?  Would using the 711 codec make a
difference in available processing power?

I'm sorry if this has been answered a million times already, I just didn't
see many configs close to what we're trying to do to compare to.  Thanks for
any input you may have.
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RE: [Asterisk-Users] Redirect two channels to each other?

[mattf]

This may be somewhat of a cross-post with -dev, but I have Manager Bridge
Action working under CVS_HEAD and releases 1.0.6-7.
http://bugs.digium.com/view.php?id=4297

MATT---

-Original Message-
From: Josiah Bryan [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Redirect two channels to each other?


On Wednesday 27 April 2005 11:38 am, Alexander Lopez wrote:
 As ny 10 year old step-daugher says I don't get it..

 Can't you just do a redirect if you specify the channels, * doesn't care
 if they are bridged together or not.  You may end up with zombie
 channels if the other leg does not drop, but you could do a soft hangup
 and take care of that..


 Or am I missing something


I dunno...maybe _im_ missing something..

IIRC (http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Redirect), 
Manager action 'Redirect' only takes Channel, ExtraChannel, Exten, Context, 
and Priority as parameters.  
 
Example (transferring existing 2 party call to a meetme room): 
 
Action: Redirect 
Channel: Zap/73-1 
ExtraChannel: SIP/199testphone-1f3c 
Exten: 8600029 
Context: default 
Priority: 1 

The problem is that you cant redirect to an existing _channel_.  I dont know

of any channel 'hack' like there is for Local extensions (e.g. to make an 
extension look like a channel, use Local/[EMAIL PROTECTED], etc - is there the 
inverse of that? Make a channel look like an extension?) 

It almost sounds like there needs to me a new manager action:

Action: Bridge
ChannelA: SIP/199testfone-1f3c
ChannelB: Zap/6-1


It sounds like the intrinsic functionality for 'bridging' is already there
in 
Asterisk (duh!), it just needs to be encapsulated in a manager action.

Any takers? Maybe a bounty is needed...?
 
-josiah

-- 
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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RE: [Asterisk-Users] Predictive Dialers

[mattf]

What exactly are you looking for?

There are basically 3 commercial solutions: Aheeva, DACX and Sinedialer
and there are 2 open-source solutions: ShadyDial and VICIDIAL

What features do you need that are not addressed by one of these?

MATT---


-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 11, 2005 1:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Predictive Dialers


I took a look but was wondering if there are any other options out there? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Nuno Viegas
|Sent: Miércoles, 11 de Mayo de 2005 04:27 a.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Predictive Dialers
|
|Hi Anton,
|
|Start by having a look at this:
|
| http://www.voip-info.org/wiki-Predictive+dialer 
|
|N
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: 11 May 2005 10:19
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Predictive Dialers
|
|Guys.
|
|Anybody know of any predictive dialers for Asterisk and Windows? 
|
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RE: [Asterisk-Users] Predictive Dialers

[mattf]

Could you let us know what you would consider a 'friendlier user interface'?

MATT---

-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 11, 2005 4:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Predictive Dialers


I like vicidial's features but Im looking for a friendlier user interface.. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of mattf
|Sent: Miércoles, 11 de Mayo de 2005 01:22 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Predictive Dialers
|
|What exactly are you looking for?
|
|There are basically 3 commercial solutions: Aheeva, DACX and 
|Sinedialer and there are 2 open-source solutions: ShadyDial 
|and VICIDIAL
|
|What features do you need that are not addressed by one of these?
|
|MATT---
|
|
|-Original Message-
|From: Anton Krall [mailto:[EMAIL PROTECTED]
|Sent: Wednesday, May 11, 2005 1:09 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Predictive Dialers
|
|
|I took a look but was wondering if there are any other options 
|out there? 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Nuno 
||Viegas
||Sent: Miércoles, 11 de Mayo de 2005 04:27 a.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Predictive Dialers
||
||Hi Anton,
||
||Start by having a look at this:
||
|| http://www.voip-info.org/wiki-Predictive+dialer
||
||N
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: 11 May 2005 10:19
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: [Asterisk-Users] Predictive Dialers
||
||Guys.
||
||Anybody know of any predictive dialers for Asterisk and Windows? 
||
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||
|
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RE: [Asterisk-Users] Predictive Dialers

[mattf]

Thanks for the suggestion, we are planning a complete overhaul of the user
interface later this summer. We definitely want to add more color and
buttons and make it look less like a depressing grey utility box.

MATT---

-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 11, 2005 10:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Predictive Dialers


For example, vicidial has great features but the screenshoots for windows
show too much info on the screen for end users to learn to like.. The screen
is more admin oriented. End users would want more buttons, etc.. For
example, take a look at Altigen.com interfaces.. Very end user oriented..
But Im looking for something open source or more accesible. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of mattf
|Sent: Miércoles, 11 de Mayo de 2005 03:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Predictive Dialers
|
|Could you let us know what you would consider a 'friendlier 
|user interface'?
|
|MATT---
|
|-Original Message-
|From: Anton Krall [mailto:[EMAIL PROTECTED]
|Sent: Wednesday, May 11, 2005 4:26 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Predictive Dialers
|
|
|I like vicidial's features but Im looking for a friendlier 
|user interface.. 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of mattf
||Sent: Miércoles, 11 de Mayo de 2005 01:22 p.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Predictive Dialers
||
||What exactly are you looking for?
||
||There are basically 3 commercial solutions: Aheeva, DACX and 
|Sinedialer 
||and there are 2 open-source solutions: ShadyDial and VICIDIAL
||
||What features do you need that are not addressed by one of these?
||
||MATT---
||
||
||-Original Message-
||From: Anton Krall [mailto:[EMAIL PROTECTED]
||Sent: Wednesday, May 11, 2005 1:09 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Predictive Dialers
||
||
||I took a look but was wondering if there are any other options out 
||there?
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Nuno 
|||Viegas
|||Sent: Miércoles, 11 de Mayo de 2005 04:27 a.m.
|||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|||Subject: RE: [Asterisk-Users] Predictive Dialers
|||
|||Hi Anton,
|||
|||Start by having a look at this:
|||
||| http://www.voip-info.org/wiki-Predictive+dialer
|||
|||N
|||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|||Krall
|||Sent: 11 May 2005 10:19
|||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|||Subject: [Asterisk-Users] Predictive Dialers
|||
|||Guys.
|||
|||Anybody know of any predictive dialers for Asterisk and Windows? 
|||
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||
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RE: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED

[mattf]

Hello,

Have you tried the wanpipe-beta8c-2.3.3.tgz release in the custom/2.3.3 dir
on their FTP site? Also, have you contacted Sangoma for support? They are
very responsive.

I am using wanpipe-beta8a-2.3.3.tgz and it's been working great on my A104
for a week now.

MATT---


-Original Message-
From: Dmitry Zhukovski [mailto:[EMAIL PROTECTED]
Sent: Monday, May 09, 2005 5:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: SV: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED


Ok, I have tested with almost all versions both in 2.3.2--*-stable and
2.3.3-*-beta - I am getting same messages: 

May  9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels
available!  Using Primary on channel anyway 16!

and same Down state

pb01*CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0

Can anybody help me?

br,
dmitry

Dmitry Zhukovski
System developer


ComX Networks A/S
Naverland 31, 2
DK-2600 Glostrup
Denmark

Phone: +45 70 25 74 74
Fax: +45 70 25 73 74
Web: www.comx.dk

Dmitry Zhukovski
Direct: +45 32 87 73 90
E-mail: [EMAIL PROTECTED]

  





Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af Chris Mason
(Lists)
Sendt: 06 May 2005 14:10
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: RE: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED


 I have got same problem with Sangoma A102 and didn't get how did you fix a
problem? Can anybody explain?

I have been working with Sangoma on a A101 installation and the fixes they
came up with will be rolled into a new beta they are releasing in the next
day or two. One thing I found out was you must stop the Zaptel modules
loading, wanrouter will load zaptel at startup.
Chris Mason
www.anguillaguide.com
Tel:  (305) 704-7249 Fax: (815)301-9759 
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RE: [Asterisk-Users] Sangoma card !

[mattf]

Hello,

Have you contacted Sangoma about this? What version of wanpipe and what
version of zaptel/asterisk are you using?

MATT---

-Original Message-
From: Nguyen Trung Tin [mailto:[EMAIL PROTECTED]
Sent: Monday, May 09, 2005 1:25 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sangoma card !


Hello All !

i'm purchased sangoma card A-101. i connect to E1 with MF/R2 signalling. but
card don't work. negotiation with E1 fail.
please help me to correct it. i dont' know some parameters such as:
MTU, BAUDRATE

Thanks

Tin Trung
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[Asterisk-Users] My Sangoma Experience in Asterisk: Followup

[mattf]

My Sangoma Experience in Asterisk: Followup
2005-05-06
  original review can be found at:
  http://astguiclient.sourceforge.net/Sangoma_experience.txt


It's now been a month since I finished my last round of tests with the
Sangoma A104u board in Asterisk. I have had a lot of conversations with the
guys at Sangoma about how my system runs so that they could set up their
test systems to try to replicate my problems. They seem to take it very
personally that their card did not work well for me and have been working on
all sorts of fixes and improvements to the wanpipe drivers and the firmware
for the boards in the last several weeks.  Here's a summary of what they've
done: 
 - D4/AMI circuits now work fully with Sangoma cards
 - RBS EM Wink start circuits now work under full load
 - Created a new Hardware HDLC PRI D-channel implementation that runs more
efficiently
 - For the A104 cards they have created a TDMV driver to streamline the
voice data path to offer better scalability

 The first two made me very happy because I was now able to fully run the
Sangoma card under full production load with D4/AMI and Wink start T1s. To
upgrade to the new software I had to first upgrade the firmware on the A104
card. Sangoma includes the firmware as well as a loader with the wanpipe
drivers. Upgrading the board firmware was actually very easy, just start the
firmware loader script(wan_aftup), select the board, pick the firmware file
to update to and it's done in a couple minutes, and you don't even have to
reboot.

 As for the driver software, Sangoma had changed a few things in the last
month(including a new installation README for asterisk). First, for the
optimized HDLC to work you now have to re-compile zaptel after you finish
your wanpipe installation(not a big deal). Second, you now have to configure
your wanpipe spans differently depending on whether it is a PRI circuit or a
RBS circuit(by selecting the DCHAN in the wancfg utility per span). Third,
now the wancfg utility can setup the startup order of the spans, taking away
an extra step that you previously had to do.

 After finishing the installation and getting Asterisk back up and running I
tried both a D4/AMI circuit and an EM Wink start circuit and found they
both worked well. Then I put some test traffic through all four T1 ports and
again everything went well. Now it was time to put the server into
production and unlike last time, it ran without any problems all day. The
performance results ended up as I expected: For our production environment a
30-50% reduced system load leading to higher capacity on the server than was
possible with the Digium TE405P board.

 Another piece of news that came out in the last month was that Digium will
be shipping the TE406p in May 2005(a 405 with an echo-canceller daughtercard
that will retail for $2,195). I would very much like to get my hands on one
of these to test it's performance against the Sangoma A104 to see what kind
of impact off-loading the echo-canceller and DTMF detection really has for
one of these cards.

 In the last month I have learned a little more about Sangoma and Asterisk.
Sangoma has given money and has been donating code to the Asterisk community
for some time now. They also have given money to other fledgling telephony
projects such as Yate (http://yate.null.ro/pmwiki/).

 In light of the improvements made in the last month and the reliability
I've seen in the system I've been running for the last month with a Sangoma
card in it, I would now recommend Sangoma cards for just about anyone except
for Linux/Asterisk novices. The configuration might be a bit confusing for a
newbie especially when compared to the ease of popping in a Digium single T1
board and going right to the Asterisk install. Other than that, Sangoma
boards are now to the point where they can be used by just about any T1/E1
user in all types of environments.

 This has been another step forward by Sangoma in the battle of the Asterisk
telco boards, but Digium has also been busy on another front recently,
developing their new channelized DS3 card. It will be interesting to see how
this card and the new TE406p perform as well as to see what new cards come
out from Sangoma in the next year. And as always I say, competition is a
good thing.


MATT---
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RE: [Asterisk-Users] Polycom Images

[mattf]

They would also be competing indirectly with themselves(Cisco and Avaya
license their technology in their phones). But the real reason comes down to
support. I've been following the Polycom/Asterisk thing for 2 years now and
it really comes down to the fact that they do not want Asterisk users
calling them. They want them to call the person that sold them the phone,
but usually those resellers have no idea where to get the firmware or they
aren't authorized by Polycom to give it out. There are a few resellers that
do have a clue and can get you the firmware. You can probably post to the
BIZ list to get a hold of a reseller that will help you.

MATT---


-Original Message-
From: Charlie Watts [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom Images


Manjit Riat wrote:
 Out of curiosity what's the reason? Why would they not sell phones to
 asterisk users? Do they not trust asterisk or their phones to work
 with each other?  

My guess: They don't want to compete with the folks that OEM Polycom
hardware. Lots of commercial phone system vendors just re-brand Polycom
phones, and Polycom doesn't want to hurt their relationship with those
businesses.

I bet that at some point one of the Asterisk-using Polycom vendors gets
the momentum to get a better Polycom-Asterisk user relationship going.
The trouble is that most of the Asterisk-using hardware vendors are
hardware agnostic, and don't want sign anything that says We'll only
sell Polycom equipment.
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RE: [Asterisk-Users] Polycom Images

[mattf]

Quote from the site when you click on the download SIP software link:

SIP
Software  - Certified VoIP Resellers can download from the Polycom Resource
Center
NOTE: At this time, end-user customers can not download software. Please
work directly with the Polycom Certified VoIP Reseller you purchased the
products from to obtain the appropriate software. 


So, no you cannot download the software unless you have the correct kind of
reseller account, and if you are a reseller you have to get your Polycom
agent to turn on this capability for you.


MATT---


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 6:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Images


mattf [EMAIL PROTECTED] writes:

 They would also be competing indirectly with themselves(Cisco and Avaya
 license their technology in their phones). But the real reason comes down
to
 support. I've been following the Polycom/Asterisk thing for 2 years now
and
 it really comes down to the fact that they do not want Asterisk users
 calling them. They want them to call the person that sold them the phone,
 but usually those resellers have no idea where to get the firmware or they
 aren't authorized by Polycom to give it out. There are a few resellers
that
 do have a clue and can get you the firmware. You can probably post to the
 BIZ list to get a hold of a reseller that will help you.

Hem, tell me if I'm wrong, but can't the firmware be downloaded directly
from polycom's web site ?

couic

-- 
Rémi
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RE: [Asterisk-Users] MEETME core uses ulaw?

[mattf]

Look at the app_conference description on the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_conference

I believe it does what you want to do, but I really don't know if it works
with CVS_HEAD or stable releases. I'd be curious to hear how it affects
performance as well.

MATT---

-Original Message-
From: Dan Morin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 04, 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] MEETME core uses ulaw?


So no one has any ideas about how to get MeetMe to work with a codec other
than ulaw?
 
Is anyone successfully doing it?
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] MOH Core uses ulaw...
 
I'm trying to get Asterisk setup as a conference bridge.  When I originally
tried MeetMe, I was using GSM and as the conference got longer, the delay
got worse and worse.  From my research, I assumed that it was because MeetMe
uses ulaw at its core, so everything is getting transcoded twice and each
instant adds more and more delay to the cycle.  To test this, I changed all
of my connections to ulaw and now I get very minimal delay.
 
However, this is not acceptable for me.  I'm anticipating most of my meeting
attendees to come in over my VoIP connection and if this voip line is using
ulaw, it will significantly reduce the number of simultaneous users that my
internet connection can handle.
 
So, it seems to me that I need to change the core codec of MeetMe to
something like GSM so that I can get OK call quality, while getting the most
out of my Internet connection.  Does anyone know how to do this?  Am I on
the right track or way off with this one?
 
Is anyone using MeetMe with GSM or any other non ulaw codec and not having a
problem? 
 
Also (sorry so many questions) I'm not thrilled with GSM or iLBC.  I know
there are a lot of people who like G.729...what are the costs involved with
using this one?
 
Thanks in advance.
Dan
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RE: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation

[mattf]

Hello,

is this how you are starting up:

1. modprobe zaptel
2. wanrouter start
3. ztcfg -v
4. asterisk -vvvgc

Also, what version of the wanroute driver software are you using?

MATT---

-Original Message-
From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 30, 2005 5:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation


I have installed Asterisk on a CentOS4 box and then installed Asterisk from
CVS.
I installed a Sangoma A101 and connected it to a Adtran 600 using a T1
Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces.

I ran through the wanpipe install instructions and configured it, now I can
run 

[EMAIL PROTECTED] asterisk]#  wanrouter hwprobe

---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A101u  : SLOT=1 : BUS=1 : IRQ=209 : CPU=A : PORT=PRI

Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0 

So I know the card is there OK.

My /etc/zaptel.conf looks like:
span=1,1,0,esf,b8zs
loadzone = us
defaultzone=us
fxsls=1-12

I am only trying to get half to load for now to make it simple.

[EMAIL PROTECTED] asterisk]# ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

When I run service zaptel restart I get:

Waiting for zap to come online...Error: missing /dev/zap!

Wha am I doing wrong?


Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (646)722-0001 Fax: (815)301-9759 
(305) 704-7249
Yahoo IM: [EMAIL PROTECTED] 
Skype ID: netconcepts

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RE: [Asterisk-Users] Asterisk Hardware Recommendation

[mattf]

If price would truly not an option just get one of the Signate Telephony
5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and
allow you to have upto 5000 SIP streams go through it. You could have that
be your gateway and do the SIP-IAX through that machine and scale upto 100
T1s if you want.

But that is a bit steep. So on to your choices. I would really say that the
setup you choose will depend on what kind of users you have as well as how
often you need to change/add users to the system and how the users are using
the system at what times. Any of them that you listed could work depending
on how they are used, but in some cases you may not want to use some of the
scenarios listed because they would either be incapable of meeting your
needs or overly complex to manage.

The easiest and cheapest one would actually not be listed:
Scenario 6:
Direct SIP-Zap on two separate servers half SIP users on each server
PSTN --2xT1-- A1  SIP_Agents
PSTN --2xT1-- A2  SIP_Agents

There is really no reason to have another 2 servers running IAX to the T1
servers, and this is simple and easy to set up and involves only 2 machines.

The next setup I would recommend would be Scenario 4, although you will have
to get a machine with a fast/wide BUS(like an Apple G5) to handle ever
increasing numbers of SIP-IAX streams as the system would grow.

If you can explain more about what kind of use this system will have I can
give a better recommendation.

MATT---


-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 10:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation


This is great information. I have the following questions based on a 
hypothetical scenario and some assumptions:

Based on the price of these configurations, I wouldn't even mind 
putting two servers each with 2 T1s just so that I could get all calls 
recorded and distribute the risk of failure.

Now, I don't know if it would make a difference or not, but here it 
goes:

Assuming the cost of the systems is of no importance for a moment 
(actually looking for the most scalable and reliable solution), which 
would be a better approach to solve the issue of activating 4 T1s which 
will be constantly taxed with load and be able to record all 
conversations:

Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register. 
Call recording in A1.
PSTN --4xT1-- A1  SIP_Agents

Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. 
Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents 
register (IAX to SIP transcoding). Call recording in A1 or A2.
PSTN --4xT1-- A1  A2  SIP_Agents

Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. 
Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP 
agents register to, and the other half would register in A1. Call 
recording in A1 and/or A2.
PSTN --4xT1-- A1  SIP_Agents
A1 --IAX-- A2  SIP_Agents

Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX 
transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP 
Agents register at A2 (IAX to SIP transcoding). Call recording in [A1 
and A3] or A2.
PSTN --2xT1-- A1  A2  SIP_Agents
PSTN --2xT1-- A3  A2  SIP_Agents

Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX 
transcoding. Asterisks (A2 and A4) will connect to A1 and A3 
respectively via IAX. Half SIP Agents register in A2 and other half in 
A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and 
A4].
PSTN --2xT1-- A1  A2  SIP_Agents
PSTN --2xT1-- A3  A4  SIP_Agents

Hopefully you're all able to understand my 5 scenarios. I guess, my 
questions would be:

1) Is there a load limiting factor in terms of whether you do the 
Monitoring of the calls when you're doing TDM-IAX transcoding or 
IAX-SIP transcoding?
2) Will a single CPU machine handle the 4 T1s to do TDM-IAX 
transcoding, if another machine is doing the actual recording (IAX-SIP 
transconding) (Scenarios 2,3,4,5). Basically, just setup cheap 
Asterisk boxes to act as VoIP gateways and the distribute the load 
and/or intelligence on other Asterisk boxes to connect SIP agents and 
all dialing rules, etc?

Thanks,
Daniel

On Apr 28, 2005, at 9:17 PM, mattf wrote:

 You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA 
 HD
 for about $600. One of those can easily handle a Sangoma dual T1 
 card($900)
 or a Digium quad T1 card($1400). For that you can have a system for 
 about
 $1500-$2000 that will be able to fully record 2 T1s(48 channels) worth 
 of
 Zap-SIP conversations. Putting two of those together with a nice big
 fileserver will give you a lot of flexibility, as well as only a 
 reduction
 in capacity if one of the servers go down instead of a total outage, 
 for
 about the same overall price of a single high-end Dual Xeon server. 
 Building
 your system

RE: [Asterisk-Users] Redirect two channels to each other?

[mattf]

I would suggest opening up a bug on the tracker, if it hasn't been done
already, for all of these discussions to be logged to.

From my glancing at the code, I think there would be a little more cleanup
involved in it than just the ast_bridge_chan function. But either way being
part of the Manager this is going to have to be clean enough to pass the
Mark test to make it into CVS. Any master Asterisk developers listening out
there who would like to tackle this one?

MATT---


-Original Message-
From: Josiah Bryan [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 1:41 PM
To: Nicolás Gudiño; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Redirect two channels to each other?


On Thursday 28 April 2005 12:57 pm, Nicolás Gudiño wrote:
  It almost sounds like there needs to me a new manager action:
 
  Action: Bridge
  ChannelA: SIP/199testfone-1f3c
  ChannelB: Zap/6-1
 
  It sounds like the intrinsic functionality for 'bridging' is already
  there in Asterisk (duh!), it just needs to be encapsulated in a manager
  action.

 Yes, we need that action on the manager! (but replace ChannelA and
 ChannelB to Channel1 and Channel2 as on the link event).

Fine by me...but does anybody know the relevant subs in any of the asterisk 
source that actually does the bridging? I mean, is it possible that its as 
easy as: 'void ast_bridge_chan( ast_chan * a, ast_chan *b )' and we just
need 
to package it as a manager action? Anybody have any pointers on how to 
proceed?

-josiah



-- 
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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RE: [Asterisk-Users] Asterisk Hardware Recommendation

[mattf]

I have never been able to do more than 50 concurrent recordings with Zap -
SIP phone calls without the audio skipping and/or breaking up. Also, if you
are using Digium TE4XXP and want to do a lot of recording I would recommend
against a SCSI RAID card because of the interrupt conflicts that you will
run into over time. I would recommend a couple of cheaper Asterisk servers
with a dual T1 or Quad T1 board in them and SATA drives, with a nice big
archive server that the audio will be copied to several times a day. Also,
do not record(Monitor) with the 'm' flag on because this will also lead to
more disk read-write while you are already trying to write another 100 or so
streams. Offload the -in and -out files to the archive server and let it
soxmix them together instead. This is the method that we have settled on for
our 12 Asterisk servers and it works rather well for us.

MATT---


-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk Hardware Recommendation


Hi,

I've been reading on the wiki as well as on this list, different 
suggestions of what to look for when designing an asterisk server with 
a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, 
that will be hit to full capacity (96 simultaneous calls). This box 
will also deliver these calls to SIP users and record all their 
conversations via Monitor.

I've heard that it's not necessarily a matter of memory (RAM) nor the 
need to have a multi-processor machine. But what really matters is that 
the motherboard (architecture) is designed to handle such a high amount 
of interrupts generated by the TE4XXP, the NIC, the storage array 
(whether it's SCSI or IDE or SATA).

Does anyone have experience with particular brands of either 
motherboards they recommend are capable to handle this or complete 
systems (e.g. Dell  or whichever brands), that are ready for this?

Thanks,
Daniel

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RE: [Asterisk-Users] Asterisk Hardware Recommendation

[mattf]

You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA HD
for about $600. One of those can easily handle a Sangoma dual T1 card($900)
or a Digium quad T1 card($1400). For that you can have a system for about
$1500-$2000 that will be able to fully record 2 T1s(48 channels) worth of
Zap-SIP conversations. Putting two of those together with a nice big
fileserver will give you a lot of flexibility, as well as only a reduction
in capacity if one of the servers go down instead of a total outage, for
about the same overall price of a single high-end Dual Xeon server. Building
your system this way from the start will also allow it to scale much more
easily than using just a single very expensive server. You can just add
another 2 T1s of capacity at any time for just $1500.

I recommend only 50 or less recordings concurrently because that is the
ceiling that we discovered while trying Zap-SIP recording on both Dual
Processor server-class systems and single processor cheaper commodity
computers as well as on SCSI, IDE and SATA drives.

If anyone out the has reliabily done recording of more than 50 conversations
I would like to know the hardware architecture of your setup.

Thanks,

MATT---


-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 6:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation


Thank you again. I will definitely do that. By cheaper asterisk 
servers, do you mean single-CPU machines that can handle Quad T1s and 
still do the call monitoring?

BTW, I tried the monitoring without the 'm' option and mounted the 
audio directory via NFS. Big NO NO for everyone. Just do what Matt 
says: copy the -in and -out to archive server separately several times 
a day :) - don't record to NFS mounted drive.

Thanks,
Daniel

On Apr 28, 2005, at 6:42 PM, mattf wrote:

 I have never been able to do more than 50 concurrent recordings with 
 Zap -
 SIP phone calls without the audio skipping and/or breaking up. Also, 
 if you
 are using Digium TE4XXP and want to do a lot of recording I would 
 recommend
 against a SCSI RAID card because of the interrupt conflicts that you 
 will
 run into over time. I would recommend a couple of cheaper Asterisk 
 servers
 with a dual T1 or Quad T1 board in them and SATA drives, with a nice 
 big
 archive server that the audio will be copied to several times a day. 
 Also,
 do not record(Monitor) with the 'm' flag on because this will also 
 lead to
 more disk read-write while you are already trying to write another 100 
 or so
 streams. Offload the -in and -out files to the archive server and let 
 it
 soxmix them together instead. This is the method that we have settled 
 on for
 our 12 Asterisk servers and it works rather well for us.

 MATT---


 -Original Message-
 From: Daniel Salama [mailto:[EMAIL PROTECTED]
 Sent: Thursday, April 28, 2005 5:56 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Asterisk Hardware Recommendation


 Hi,

 I've been reading on the wiki as well as on this list, different
 suggestions of what to look for when designing an asterisk server with
 a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP,
 that will be hit to full capacity (96 simultaneous calls). This box
 will also deliver these calls to SIP users and record all their
 conversations via Monitor.

 I've heard that it's not necessarily a matter of memory (RAM) nor the
 need to have a multi-processor machine. But what really matters is that
 the motherboard (architecture) is designed to handle such a high amount
 of interrupts generated by the TE4XXP, the NIC, the storage array
 (whether it's SCSI or IDE or SATA).

 Does anyone have experience with particular brands of either
 motherboards they recommend are capable to handle this or complete
 systems (e.g. Dell  or whichever brands), that are ready for this?

 Thanks,
 Daniel

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RE: [Asterisk-Users] Redirect two channels to each other?

[mattf]

Maybe this would best be explained in a diagram:

1).  person A --- music on hold  and  person B --- music on hold

2).  *some manager API action*

3).  person A --- person B


This is what I think he's asking about, how do you take two parties on
different conversations and put them together without using a meetme
conference?


MATT---


-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Redirect two channels to each other?


 
As ny 10 year old step-daugher says I don't get it..

Can't you just do a redirect if you specify the channels, * doesn't care
if they are bridged together or not.  You may end up with zombie
channels if the other leg does not drop, but you could do a soft hangup
and take care of that..


Or am I missing something


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Wednesday, April 27, 2005 10:41 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Redirect two channels to each other?

I've been scratching my head trying to think of a way to do this, but
without success yet.

I'm using the Manager API. If I have two channels linked to each other
(i.e. direct connection), or even if they are independent channels, I
can transfer them both to the same extension by using Action: Redirect
and using Channel: for one and ExtraChannel: for the other. This is most
useful for putting them both in the same Meetme conference.

What I want to do is find a way to take two unrelated existing channels
(which for the sake of argument might be sitting in MusicOnHold,
separate conferences, the same conference or whatever), and link them
together into a direct call rather than having them talk via their own
Meetme conference.

Does anyone have any ideas if this can be done?

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] Re: Redirect two channels to each other?

[mattf]

I haven't looked through the code yet, but no, there is no easy way that
I've discovered to be able to do this, although it seems like there should
be. 

All that we really need to be able to do is somehow execute a 'Link' action
between two channels that would bridge the two desired channels and unlink
whatever those two channels were originally connected to. I'll take a look
at the manager API code later this week and see if there's any easy way of
patching it to accomplish this.

Thanks,

MATT---

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 12:44 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Redirect two channels to each other?


In article
[EMAIL PROTECTED],
mattf [EMAIL PROTECTED] wrote:
 Maybe this would best be explained in a diagram:
 
 1).  person A --- music on hold  and  person B --- music on
hold
 
 2).  *some manager API action*
 
 3).  person A --- person B
 
 
 This is what I think he's asking about, how do you take two parties on
 different conversations and put them together without using a meetme
 conference?

Thanks Matt, that is exactly what I am asking. I assume you haven't found
a way either, otherwise you would have mentioned it! :-)

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] tonezone in tunisia

[mattf]

Tunisia should be +1 from GMT

http://www.freedomphones.net/phone_codes_GMT.txt


MATT---


-Original Message-
From: Samuel T. Cossette [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 1:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] tonezone in tunisia


Hi,

Is there someone here who know which tonezone I should use in Tunisia?

The only information I have it's the ringtone: 425/1200, 0/4600.

thanks,

Samuel T. Cossette
[EMAIL PROTECTED]

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RE: [Asterisk-Users] Monitor via Manager question

[mattf]

Doesn't work that way, you have to know the exact channel that you want to
Monitor on SIP. What you can do is a Command Show Channels in the manager
and parse through the output to find the first channel then do your Monitor
command.

MATT---

-Original Message-
From: Dana Olson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 20, 2005 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Monitor via Manager question


Hello. I checked in the wiki and read a bunch of old threads from this
mailing list but haven't found what I'm looking for.

I'm using a simple PHP script, and here is the relevant portion:

fputs($socket, Action: Monitor\r\n);
fputs($socket, Channel: Zap/1-1\r\n\r\n);

That works fine. As does this:

fputs($socket, Action: Monitor\r\n);
fputs($socket, Channel: SIP/8000-h4d8\r\n\r\n);

But what I need to be able to do is this:

fputs($socket, Action: Monitor\r\n);
fputs($socket, Channel: SIP/8000\r\n\r\n);

And have it record either the first call that is up on SIP/8000, or
the last, or whatever (doesn't matter, only one call at a time will be
up on this line).

However, if I try this, it always comes back to me with:

Response: Error
Message: No such channel

Because I am not specifying the actual call that is up. Is there any
way to do this? Or can I somehow easily look up what Zap channel is
used by SIP/8000 and pass that?

The other twist is that SIP/8000 will be specified by a variable
passed through a form. Basically, I want a web form with two buttons
and a text box: Start Rec., Stop Rec., and User Ext.. I didn't start
out that complex though, just right now it's a simple PHP script, and
it was taken from the Wiki. I need to get the core functionality
working properly before I add the buttons and whatever.

Thanks in advance for any advice.

--
Dana
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RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds

[mattf]

Have they improved in 3 years? I used to have a half dozen Compaq servers
and had all sorts of problems with them and I got very tired of calling my
authorized Compaq repair professional. That's when I swore off brand-name
servers and their high price tags and started building my own in-house.
Also, the high-end server boards seem to be more expensive for a slower
option than what is available from the cutting-edge desktop motherboards. I
had bought a ServerWorks motherboard a year ago for a DB server, and it
works great, but the Asus DB system that I put together for half the money
at the same time is 30% faster and both are still running strong.

MATT---


-Original Message-
From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 20, 2005 11:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P
car ds


Seems odd, though I would suspect the boards.

Have you tried higher end boards, like compaq proliant servers?

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, April 18, 2005 11:25 AM
To: 'Andrew Latham'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P
car ds

For this particular server all telco equipment is in a climate
controlled room kept at 66 degrees F and they are all on APC SmartUPS
rackmount power battery backups, Also all of these connections had
previously been connected to other Digium cards in the last year with no
issues.

MATT---

-Original Message-
From: Andrew Latham [mailto:[EMAIL PROTECTED]
Sent: Monday, April 18, 2005 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Motherboard failure with 2 Digium TE405P
cards


Check the telco equipment you are plugging into (PBXes) with the
crossovers.. Unless they are all on the same power grid and protected I
would blame them. my two cents...

On 4/18/05, mattf [EMAIL PROTECTED] wrote:
 Hello,
 
 I have spend a long time trying to figure out exactly what is the 
 problem with one of my Asterisk servers, it is the only one at any of 
 our
locations
 that has two Digium quad T1 cards in it with 7 T1s connected to it. 
 Most
of
 the rest of our Asterisk servers run identical hardware except that 
 they only have a single TE405P board in them. Here's what seems to 
 happen to
this
 system starting 6 months ago:
 
 Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or 
 SCSI drives and two TE405P Digium quad T1 boards. Hook up one local 
 and one
long
 distance T1, hook up 4 crossover PRIs to other telco equipment, hook 
 up
one
 channelbank.
 
 The system will run perfectly for about 5 weeks, then randomly the 
 channel bank users will notice a weird audio cracking sound and the 
 system will crash. Upon investigation the second TE405P card will have

 it's lights all off and on reboot they will not go back on again. 
 After frantically switching the PCI slot that the lights-out card was 
 in to a free slot the card works again and everything is happy again, 
 but now no digium card
will
 work in the other slot again. Another 5 weeks or so passes and again 
 one
of
 the Digium quad cards stops working. At this point I swap out the 
 entire system(including quad cards) with another system that has been 
 running for
6
 months with no problem and put the malfunctioning system in production
with
 a single quad card(which now has been running fine 4 months later) and
after
 6 weeks it happens to the new system. The whole process repeats itself

 and
I
 am now on my 3rd set of completely different components serving in 
 this role(even with different brands of components) and my first PCI 
 slot just failed last week. We need to have the capability to handle 7

 T1s on this machine and it is not over-heated or overloaded from a 
 system load standpoint. We also have $200 550W Enermax power supplies 
 in these servers that have never failed us before.
 
 So here's the question, do two Digium TE405P boards draw too much 
 power or do something else that would harm a brand new motherboard
over time?
 
 Does anyone else out there run two quad board in production? if so 
 what hardware do you use?
 
 I'm just looking for some user feedback before I contact Digium 
 hardware support on this.
 
 Thanks,
 
 MATT---
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--
Andrew Latham

http://www.lathama.com
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Vici Dialer

[mattf]

Title: Vici Dialer



It's 
quite easy to do multiple campaigns on VICIDIAL(just create a new campaign, load 
leads, assign the leads to that campaign and have the agent log into the new 
campaign), you might want to post you question on the astGUIclient-users mailing 
list though:

https://lists.sourceforge.net/lists/listinfo/astguiclient-users

MATT---

  -Original Message-From: TOBY 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, April 19, 2005 
  1:23 AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] Vici Dialer
  Anyone out there with any Vici Dialer experience? 
  I'm using it as the dialer for Asterisk. I need to configure it for multiple 
  campaigns.
  Toby [EMAIL PROTECTED] 
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[Asterisk-Users] Motherboard failure with 2 Digium TE405P cards

[mattf]

Hello,

I have spend a long time trying to figure out exactly what is the problem
with one of my Asterisk servers, it is the only one at any of our locations
that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of
the rest of our Asterisk servers run identical hardware except that they
only have a single TE405P board in them. Here's what seems to happen to this
system starting 6 months ago:

Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI
drives and two TE405P Digium quad T1 boards. Hook up one local and one long
distance T1, hook up 4 crossover PRIs to other telco equipment, hook up one
channelbank.

The system will run perfectly for about 5 weeks, then randomly the channel
bank users will notice a weird audio cracking sound and the system will
crash. Upon investigation the second TE405P card will have it's lights all
off and on reboot they will not go back on again. After frantically
switching the PCI slot that the lights-out card was in to a free slot the
card works again and everything is happy again, but now no digium card will
work in the other slot again. Another 5 weeks or so passes and again one of
the Digium quad cards stops working. At this point I swap out the entire
system(including quad cards) with another system that has been running for 6
months with no problem and put the malfunctioning system in production with
a single quad card(which now has been running fine 4 months later) and after
6 weeks it happens to the new system. The whole process repeats itself and I
am now on my 3rd set of completely different components serving in this
role(even with different brands of components) and my first PCI slot just
failed last week. We need to have the capability to handle 7 T1s on this
machine and it is not over-heated or overloaded from a system load
standpoint. We also have $200 550W Enermax power supplies in these servers
that have never failed us before.

So here's the question, do two Digium TE405P boards draw too much power or
do something else that would harm a brand new motherboard over time?

Does anyone else out there run two quad board in production? if so what
hardware do you use?

I'm just looking for some user feedback before I contact Digium hardware
support on this.

Thanks,

MATT---
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RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds

[mattf]

For this particular server all telco equipment is in a climate controlled
room kept at 66 degrees F and they are all on APC SmartUPS rackmount power
battery backups, Also all of these connections had previously been connected
to other Digium cards in the last year with no issues.

MATT---

-Original Message-
From: Andrew Latham [mailto:[EMAIL PROTECTED]
Sent: Monday, April 18, 2005 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Motherboard failure with 2 Digium TE405P
cards


Check the telco equipment you are plugging into (PBXes) with the
crossovers.. Unless they are all on the same power grid and protected
I would blame them. my two cents...

On 4/18/05, mattf [EMAIL PROTECTED] wrote:
 Hello,
 
 I have spend a long time trying to figure out exactly what is the problem
 with one of my Asterisk servers, it is the only one at any of our
locations
 that has two Digium quad T1 cards in it with 7 T1s connected to it. Most
of
 the rest of our Asterisk servers run identical hardware except that they
 only have a single TE405P board in them. Here's what seems to happen to
this
 system starting 6 months ago:
 
 Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI
 drives and two TE405P Digium quad T1 boards. Hook up one local and one
long
 distance T1, hook up 4 crossover PRIs to other telco equipment, hook up
one
 channelbank.
 
 The system will run perfectly for about 5 weeks, then randomly the channel
 bank users will notice a weird audio cracking sound and the system will
 crash. Upon investigation the second TE405P card will have it's lights all
 off and on reboot they will not go back on again. After frantically
 switching the PCI slot that the lights-out card was in to a free slot the
 card works again and everything is happy again, but now no digium card
will
 work in the other slot again. Another 5 weeks or so passes and again one
of
 the Digium quad cards stops working. At this point I swap out the entire
 system(including quad cards) with another system that has been running for
6
 months with no problem and put the malfunctioning system in production
with
 a single quad card(which now has been running fine 4 months later) and
after
 6 weeks it happens to the new system. The whole process repeats itself and
I
 am now on my 3rd set of completely different components serving in this
 role(even with different brands of components) and my first PCI slot just
 failed last week. We need to have the capability to handle 7 T1s on this
 machine and it is not over-heated or overloaded from a system load
 standpoint. We also have $200 550W Enermax power supplies in these servers
 that have never failed us before.
 
 So here's the question, do two Digium TE405P boards draw too much power or
 do something else that would harm a brand new motherboard over time?
 
 Does anyone else out there run two quad board in production? if so what
 hardware do you use?
 
 I'm just looking for some user feedback before I contact Digium hardware
 support on this.
 
 Thanks,
 
 MATT---
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-- 
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http://www.lathama.com
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
If any of the above are not working,
we have bigger problems than my email.
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RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds

[mattf]

I have 2 temperature probes in the server, they record peak temperature and
neither have gotten within 5 degrees of our peak usage Asterisk servers'
average temperature. Also the current machine has all new components and no
dust buildup or fan blockage.

Our server room is monitored by two independant room temperature sensors
that log temperature every 15 minutes and if it gets over 85F the system
will phone 3 of us every 15 minutes until the temperature goes down or the
system is turned off. We have not had any AC problems since we put the new
AC system in 6 months ago.

We went to this length because we have had several of the things you
mentioned happen to us as well, The server room has a dedicated AC unit, you
need a key to change the thermostat temperature, and our machines have very
good air flow front to back with either very few or no significant heat
traps. 

This seems to be a power or motherboard issue that I cannot figure out. Does
anyone have the actual power usage rating of the Digium TE405P card?

Thanks,

MATT---

-Original Message-
From: Race Vanderdecken [mailto:[EMAIL PROTECTED]
Sent: Monday, April 18, 2005 2:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P
cards


Just from long term experience it might be a heat problem.

Check the really basic stuff first.

The air flow might not be adequate for the box. Make sure your ribbon
cables and such are not blocking flow. 

Two cards might draw too much power, causing the power supply to
overheat causing everything to overheat. 

Don't add more fans, put in better/more efficient fans or a better power
supply.

After six months are you getting dust build up on the fans or vents?
More dust traps more heat which cause more power to be needed to run
fans and convert AC/DC which causes more heat, and so on. But you are
reporting a five week breakdown.

Put a recording thermometer in your boxes. It could be the cooling is
not running as expected in the room. Do you own the room?

I once had a room where the janitor would shut the air-conditioning off
at night because he knew nobody was in there. Then he would turn it back
on in the morning before I got there. The machine was dead, but the room
was ice cold. That took three weeks and a lot of IBM repair guys later
to discover. I only found it because I checked the room on a weekend and
it was 90+ in there.

Don't over tax the air-conditioners. I once had a room where the company
insisted on keeping it at 60 because things kept over heating, every
time there was an over heating they pushed the thermostat lower. Turns
out the air-conditioner was turning itself off because it was
overheating from the demand. Then after a few hours off it would
comeback on, cool and overheat itself because it was unable to keep the
room as cold as a meat locker. 

Even better, another time someone brought in a portable cooler to keep a
room/closet with a switch in it like an icebox. They vented the heat
from the portable cooler out of the room into the dropped ceiling via a
20 foot exhaust hose. By stuffing the exhaust hose into the plenum the
hose was accidentally pointed at the thermostat of the HVAC thermostat
for the entire office. So with 90+ degree air pointed at the office HVAC
thermostat the office HVAC thought it was 90 inside and kept the place
so cold we could barely work there during the winter.

Moral of the story, sometimes it ain't anything you are doing.

Race the Tyrant Vanderdecken

-Original Message- 
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, April 18, 2005 10:35 AM
To: 'asterisk-users@lists.digium.com'
Subject: [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards

Hello,

I have spend a long time trying to figure out exactly what is the
problem
with one of my Asterisk servers, it is the only one at any of our
locations
that has two Digium quad T1 cards in it with 7 T1s connected to it. Most
of
the rest of our Asterisk servers run identical hardware except that they
only have a single TE405P board in them. Here's what seems to happen to
this
system starting 6 months ago:

Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI
drives and two TE405P Digium quad T1 boards. Hook up one local and one
long
distance T1, hook up 4 crossover PRIs to other telco equipment, hook up
one
channelbank.

The system will run perfectly for about 5 weeks, then randomly the
channel
bank users will notice a weird audio cracking sound and the system will
crash. Upon investigation the second TE405P card will have it's lights
all
off and on reboot they will not go back on again. After frantically
switching the PCI slot that the lights-out card was in to a free slot
the
card works again and everything is happy again, but now no digium card
will
work in the other slot again. Another 5 weeks or so passes and again one
of
the Digium quad cards stops

RE: [Asterisk-Users] Sangoma A101 + Rhino channelbank

[mattf]

Keep on bugging the Sangoma guys, I know they are working on several RBS T1
issues right now(They called me Friday to go over a few things) They just
need help from users like you and I to find the bugs in their drivers.

Have you tried any other signalling types other than LOOP?

MATT---


-Original Message-
From: Felician CHELU [mailto:[EMAIL PROTECTED]
Sent: Monday, April 11, 2005 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sangoma A101 + Rhino channelbank


Hello,

I have Asterisk 1.0.6 - I  try to setup Sangoma A101 T1 board together with
the Rhino fxs chanelbank.
Things done:
-  T1 cross cable = I have carrier, signalling and framnig leds on
the channelbank green.
- channelbank configuration:
t1 - Proto: LOOP  Frame: esf  Clock: slave   Coding:
b8zs
channels(analog) : Function:A-fxsMode:loop
- zaptel.conf
span=2,1,0,esf,b8zs
fxols=32-55
(i have a span 1 with a digium e1)
- zapata.conf
 signalling=fxo_ls
- wanpipe1.conf

[devices]
wanpipe1 = WAN_AFT, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 10
PCIBUS  = 2
FE_MEDIA= T1
FE_LCODE= B8ZS
FE_FRAME= ESF
FE_LINE = 1
TE_CLOCK= MASTER
ACTIVE_CH   = ALL
TE_HIGHIMPEDANCE= NO
LBO = 0DB
INTERFACE   = V35
CLOCKING= EXTERNAL
BaudRate= 0
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO

[w1g1]
PROTOCOL= HDLC
HDLC_STREAMING  = YES
ACTIVE_CH   = ALL
IDLE_FLAG   = 0x7E
MTU = 1500
MRU = 1500
TDMV_SPAN   = 2
TDMV_ECHO_OFF   = NO
MULTICAST   = NO
TRUE_ENCODING_TYPE  = NO


I already called Sangoma and Rhino support, but after hours of long distance
call conversation the problem is still not solved. Finnaly, a guy from Rhino
told me that their asterisk expert (which was not avaliable) knows about
this problem and that it is that the sangoma driver is not communicating
with asterisk.

The wanrouter starts ok, after ztcfg I see the channels configured.
The problem: i don't have dialtone on phones.

Question: When i enter zttoll, if i go to the sangoma span and I make loop
then it freezes. Is it normal?

If someone has experienced this combination and made it work please give me
a sign.

Thank you.

PS:

Felician

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RE: [Asterisk-Users] Using manager interface to play aanouncments in aMeetMe

[mattf]

I've wondered about this as well. I suggest posting a bug to the bug tracker
and see if you can get a clarification or better yet, get someone to fix
this. It would be nice to override the clearing of the vars for Local
channels.

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Friday, April 08, 2005 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin aMeetMe


A little more googling and wiki browsing shows that the default
behaviour
of the Local channel is to dump the variables.  According to the wiki I
can append /n to the channel identifier to preserve variables, but this
does
not seem to be working.  I'm running 1.0.7.

Can anyone lend me a clue?

Thanks,
Dan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Friday, April 08, 2005 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin aMeetMe

I've run into a snag.  I make extensive use of dynamic conferences
and thought it would be a no-brainer to pass the conference number
with the manager interface.  Looking at the wiki and sample code,
I thought I had it right, but no joy.

PHP manager call-
$res = $as-connect();
if (!$res){ echo 'Error connection to the manager!'; exit();}
$res = $as-send_request('Originate', array('Channel' =
'Local/[EMAIL PROTECTED]', 'Context' = 'mm-announce', 'Exten' = '',
'Priority' = '1', 'Variable' = 'confNo=$confNo'));
$res = $as-disconnect();

extensions.conf-
 
[mm-announce]
exten = 9998,1,Answer
exten = 9998,2,noop,${confNo} ;to test, it was meetme(${confNo})
exten = 9998,3,Hangup

exten = ,1,Answer
exten = ,2,Playback(this-conf-will-end-in-5-minutes)
exten = ,3,Hangup

Console output-
  == Manager 'MeetMe' logged on from 127.0.0.1
-- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new
stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, ) in new stack
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new
stack
-- Executing Answer(Local/[EMAIL PROTECTED],1, ) in new
stack
-- Executing Playback(Local/[EMAIL PROTECTED],1,
this-conf-will-end-in-5-minutes) in new stack
-- Playing 'this-conf-will-end-in-5-minutes' (language 'en')
  == Manager 'MeetMe' logged off from 127.0.0.1

So it appears that my variable ${confNo} is not being set, or at least
honored. 

Any thoughts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe

Hello,

Here's jsut a simple manager Action to send, make sure that you have an
extension set up to play the message(exten = 1234,1,Playback(file)) and
that's the extension that will be called from the meetme room. Also,
make
sure that that extension calls in to the meetme room extension with the
'q'
flag so that noone hears the welcome and leaving tone.

exten = 1234,1,Answer
exten = 1234,2,Playback(out_of_time)
exten = 1234,3,Hangup


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: default
Exten: 1234
Priority: 1


where 78600051 is the exten to get to your meetme room.

Let me know if you have any questions,

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe


A sample would be great.  I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.

Dan 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe

just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.

Let me know if you would like an example and I'll whip one up.

We do this kind of thing in astGUIclient to play DTMF tones
automatically in
meetme rooms.

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Using manager interface to play aanouncments
in a MeetMe


I am wrapping up a PHP addon script to my scheduling
framework and have it properly tracking and closing
conferences.

I need to play an announcement into the room that the
conference will end soon.  I haven't found a great way
to do that.  One way that I have thought of, but would
like to avoid is adding a Playback

[Asterisk-Users] My Sangoma Experience - Review

[mattf]

My Sangoma Experience in Asterisk:  2005-04-07

Having pushed my Digium Asterisk systems to their capacity many times and
figuring out the limits of the Digium hardware I decided it was time to test
an Asterisk-compatible Sangoma Quad T1/E1 card(AFT-A104u) to see if they
live up to their hype of being more efficient than the Digium
variety(T405P). I had talked with someone from Sangoma before at Astricon,
but it was rather informal, he didn't have any literature and I was rather
swamped at the time as it was. Then I saw a posting on the asterisk-users
list about the claims that the Sangoma card does echo-cancelation better as
well as using far less interrupts than Digium hardware(a big bottleneck with
busy Digium systems).

I emailed Sangoma(they are located in Canada) for a quote and quickly
received a phone call from them. They were very interested in getting my
feedback on using their quad port T1/E1 card with Asterisk and they quoted
me a discounted price of $1190 US for the card(They said retail was $1700 US
[Digium quad-cards are $1495 retail but you can get them through resellers
for a couple hundred less]). The Sangoma card comes with a 30-day money back
guarantee and a 3 year warranty.

When I received the card I noticed a couple things right away, it was a very
professionally packaged item and it came with 4 T1 cables in the box as well
as documentation and all of the other pretty things you expect in a retail
package. The second thing I noticed is that the card was compatible with a
2U form-factor(That's right, they crammed 4 T1/E1 ports together so it can
fit in a 2U case vertically) This was achieved in-part because the ports are
actually on a fixed daughter card, but it did bring up the thought that they
could actually cram 6 ports on one of these cards :)

Next I started to sort through the documentation and files on their FTP
site. I noticed something I wish Digium cards had: User-upgradable firmware
on the board(I have previously had to return an early version of the T410P
Digium board to get a newer one with newer firmware on it).

Let the installation begin. I started by downloading and installing Asterisk
as usual(zaptel, libpri, asterisk[version 1.0.6]), then I downloaded and
installed Wanpipe release 2.3.2 beta6. I could now see my card and went into
the wancfg utility to configure my card. Here's when it stopped being a
smooth experience. I tried installing it by the asterisk instructions found
on the FTP site(which I found out later were out of date and incorrect) and
eventually it all worked up until the final starting step. The drivers saw
the card, but said nothing was connected to them which I thought was a
strange problem since you don't have to have anything connected to a Digium
card for Asterisk to fully startup. So I emailed tech support and walked
through some reconfiguration steps and then after a few more emails back and
forth it came out that they had a problem with D4/AMI signalling on a RBS
T1(which they say they will have a fix for at some undefined time in the
future). After switching the wanpipe config for the first span to B8ZS/ESF
with a PRI T1 I was able to run ztcfg and asterisk. I placed some test calls
and all went well, at least until I tried hooking up a live RBS(Robbed-bit,
24 full channels not PRI) EM Wink T1. It turns out that the guys at Sangoma
have never had a customer that used EM Wink start and accordingly they have
never tested their cards with it, and of course it didn't work. So another
email and call to Sangoma and they started working on a fix. Two days later
they added a Wink for wink start T1s and sent me a new version of the
software. I loaded it and it worked, but all audio and call detects stopped
working if I tried to use more than 10 of the RBS T1 channels, so back to
Sangoma for another new driver version. After a few days, and a few more
driver versions, they came up with one that seemed to fix all of the
problems I was having before so I did my simple stress test of picking up,
hanging up and redirecting to meetme of about 52 Zap lines and all went
well. Now on to the performance testing.

For a performance test, I swapped out an identically configured machine that
had a Digium T405P with my test machine and put it live in company
inbound/outbound call center during off-hours to test(This server usually
handles over 20,000 calls in/out a day with lots of recording going on
across T1s, SIP phones and some IAX2 trunks). This server has two RBS T1s,
one PRI T1 and one Channel Bank. I placed a test call out of the channel
bank through the PRI and then started automated calls from the two RBS T1s
to go into meetme conferences. The performance test ran great and it did
prove that there is reduced CPU usage on a Sangoma board as compared to a
Digium board. For a running time of about an hour the CPU usage was between
30% and 50% lower with the Sangoma board on the identically configured
machine. This was just doing some 

RE: [Asterisk-Users] open source Asterisk Application of the year ?

[mattf]

As an Asterisk-related Open-Source project developer I would very much like
this idea :)

We could have a competition that ends yearly during Astricon at which point
the application is chosen.

But who would judge which is best?

I'm pretty sure that the front runners if this was done this year would
either be the [EMAIL PROTECTED], the AMP package or the Flash-Operator-Panel.

Anybody want to organize this? I nominate Olle.

MATT---

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 10:10 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] open source Asterisk Application of the year?


Hello,

I was just wondering if there were a prize like the open source application
of
the year relative to Asterisk?

All these developer doing good job and all free need some present sometime
that
we can all donate.

Anything like that exists?



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RE: [Asterisk-Users] My Sangoma Experience - Review

[mattf]

Several of these RBS T1s have been here for many years and before we moved
to Asterisk a few pieces of phone hardware we used were not PRI-compatible.
There is also the fact that we still use Channel banks which are also RBS.
We have started a long process of switching to PRIs as our RBS T1 contracts
expire, but that is going to take another 2 years. Pricing was not really an
issue.

MATT---


-Original Message-
From: Tom [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] My Sangoma Experience - Review


Thanks for the informative review Matt.  Please tell why you are using RBS 
T1 trunks instead of PRIs.  Is it the cost or availability issue from the 
ILEC/CLEC or is there some other advantage.  PRIs and RBS T1s are about the 
same price in my part of the world.

Tom

At 09:20 AM 4/7/2005, you wrote:
My Sangoma Experience in Asterisk:  2005-04-07

Having pushed my Digium Asterisk systems to their capacity many times and
figuring out the limits of the Digium hardware I decided it was time to
test
an Asterisk-compatible Sangoma Quad T1/E1 card(AFT-A104u) to see if they
live up to their hype of being more efficient than the Digium
variety(T405P). I had talked with someone from Sangoma before at Astricon,
but it was rather informal, he didn't have any literature and I was rather
swamped at the time as it was. Then I saw a posting on the asterisk-users
list about the claims that the Sangoma card does echo-cancelation better as
well as using far less interrupts than Digium hardware(a big bottleneck
with
busy Digium systems).

I emailed Sangoma(they are located in Canada) for a quote and quickly
received a phone call from them. They were very interested in getting my
feedback on using their quad port T1/E1 card with Asterisk and they quoted
me a discounted price of $1190 US for the card(They said retail was $1700
US
[Digium quad-cards are $1495 retail but you can get them through resellers
for a couple hundred less]). The Sangoma card comes with a 30-day money
back
guarantee and a 3 year warranty.

When I received the card I noticed a couple things right away, it was a
very
professionally packaged item and it came with 4 T1 cables in the box as
well
as documentation and all of the other pretty things you expect in a retail
package. The second thing I noticed is that the card was compatible with a
2U form-factor(That's right, they crammed 4 T1/E1 ports together so it can
fit in a 2U case vertically) This was achieved in-part because the ports
are
actually on a fixed daughter card, but it did bring up the thought that
they
could actually cram 6 ports on one of these cards :)

Next I started to sort through the documentation and files on their FTP
site. I noticed something I wish Digium cards had: User-upgradable firmware
on the board(I have previously had to return an early version of the T410P
Digium board to get a newer one with newer firmware on it).

Let the installation begin. I started by downloading and installing
Asterisk
as usual(zaptel, libpri, asterisk[version 1.0.6]), then I downloaded and
installed Wanpipe release 2.3.2 beta6. I could now see my card and went
into
the wancfg utility to configure my card. Here's when it stopped being a
smooth experience. I tried installing it by the asterisk instructions found
on the FTP site(which I found out later were out of date and incorrect) and
eventually it all worked up until the final starting step. The drivers saw
the card, but said nothing was connected to them which I thought was a
strange problem since you don't have to have anything connected to a Digium
card for Asterisk to fully startup. So I emailed tech support and walked
through some reconfiguration steps and then after a few more emails back
and
forth it came out that they had a problem with D4/AMI signalling on a RBS
T1(which they say they will have a fix for at some undefined time in the
future). After switching the wanpipe config for the first span to B8ZS/ESF
with a PRI T1 I was able to run ztcfg and asterisk. I placed some test
calls
and all went well, at least until I tried hooking up a live RBS(Robbed-bit,
24 full channels not PRI) EM Wink T1. It turns out that the guys at
Sangoma
have never had a customer that used EM Wink start and accordingly they
have
never tested their cards with it, and of course it didn't work. So another
email and call to Sangoma and they started working on a fix. Two days later
they added a Wink for wink start T1s and sent me a new version of the
software. I loaded it and it worked, but all audio and call detects stopped
working if I tried to use more than 10 of the RBS T1 channels, so back to
Sangoma for another new driver version. After a few days, and a few more
driver versions, they came up with one that seemed to fix all of the
problems I was having before so I did my simple stress test of picking up,
hanging up and redirecting 

RE: [Asterisk-Users] My Sangoma Experience - Review

[mattf]

Hello,

This would be software since I still don't see the Digium echo-cancellers
anywhere for sale and don't know how to get one. If Digium wants to send me
one I would gladly test it.

The overall machine load is also affected by the way interrupts are used and
the fact that Sangoma uses significantly less than a Digium card uses.

MATT---



Hi,


Could you please include if you used the software zaptel echo canceller
or the daughterboards for the te4xxp ?
As that would explain the difference in cpu usage.

I have no daughterboards for the te410p cards yet, nor do i own any
sangoma things, so no testing here.

Joachim.

Joachim.

Tom wrote:

 Thanks for the informative review Matt.  Please tell why you are using
 RBS T1 trunks instead of PRIs.  Is it the cost or availability issue
 from the ILEC/CLEC or is there some other advantage.  PRIs and RBS T1s
 are about the same price in my part of the world.

 Tom

-Original Message-
From: Zoa [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] My Sangoma Experience - Review


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RE: [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe

[mattf]

just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.

Let me know if you would like an example and I'll whip one up.

We do this kind of thing in astGUIclient to play DTMF tones automatically in
meetme rooms.

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Using manager interface to play aanouncments
in a MeetMe


I am wrapping up a PHP addon script to my scheduling
framework and have it properly tracking and closing
conferences.

I need to play an announcement into the room that the
conference will end soon.  I haven't found a great way
to do that.  One way that I have thought of, but would
like to avoid is adding a Playback command to the 
MeetMeAdmin commands.

If anyone knowns of another way, I would be delighted.

Dan
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RE: [Asterisk-Users] Using manager interface to play aanouncments in a MeetMe

[mattf]

Hello,

Here's jsut a simple manager Action to send, make sure that you have an
extension set up to play the message(exten = 1234,1,Playback(file)) and
that's the extension that will be called from the meetme room. Also, make
sure that that extension calls in to the meetme room extension with the 'q'
flag so that noone hears the welcome and leaving tone.

exten = 1234,1,Answer
exten = 1234,2,Playback(out_of_time)
exten = 1234,3,Hangup


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: default
Exten: 1234
Priority: 1


where 78600051 is the exten to get to your meetme room.

Let me know if you have any questions,

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe


A sample would be great.  I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.

Dan 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe

just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.

Let me know if you would like an example and I'll whip one up.

We do this kind of thing in astGUIclient to play DTMF tones
automatically in
meetme rooms.

MATT---


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Using manager interface to play aanouncments
in a MeetMe


I am wrapping up a PHP addon script to my scheduling
framework and have it properly tracking and closing
conferences.

I need to play an announcement into the room that the
conference will end soon.  I haven't found a great way
to do that.  One way that I have thought of, but would
like to avoid is adding a Playback command to the 
MeetMeAdmin commands.

If anyone knowns of another way, I would be delighted.

Dan
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RE: [Asterisk-Users] Petition for IAX firmware

[mattf]

We target Sipura because they are relatively a small company, the core
developers at Sipura used to work for Cisco and worked on their ATA product
before they started their own company. A small company is much more likely
to try something new with little lead-time. Also access to decision-makers
is much easier at a company like Sipura, I talked to one of the developers
about a year and a half ago when they were going to market with the SPA-2000
and at that time they seemed willing to try lots of new things. I even
mentioned Asterisk and IAX to them but they had never heard of it.

I'm sure if they get a list of hundreds of people who would buy an IAX
adapter that they would at least do a test release of IAX2 firmware.

MATT---


-Original Message-
From: Adam Goryachev [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 06, 2005 5:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Petition for IAX firmware


   Hi all,
  
   I've put together a quick petition, in hopes that we can possibly 
   persuade Sipura (or any other large-scale IP handset manufacturer) to 
   include firmware support for IAX. The IAXy has proven that an IAX 
   product is in demand, and very useful, and I think we'd all like to 
   see a handset manufacturer follow Digium's lead. I'm not particularly 
   endorsing Sipura, however I do know that they have seriously 
   considered support for IAX, and have decided to hold off until the 
   demand is there. I'm hoping that with some numbers, we can prove to 
   them that the demand is already here, and that IAX is already a viable

   technology.
  
   I'd like to encourage everyone to show your support -- hopefully 
   Sipura, and/or other manufacturers will see these hard names and 
   numbers, and realize it's time to move something into production.
  
   Petition:
   http://www.petitiononline.com/IAXPhone
  

My 2c is to re-title the petition to include IAX2 support in all
ATA's/phones. Why target sipura? What happened to snom who at one stage
had a IAX (version 1) firmware? Why not target polycom/cisco, or any of
the other 100 manufacturers/resellers/re-badging companies??

Regards,
Adam

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RE: [Asterisk-Users] Sangoma VS. Digium

[mattf]

Hello,

I need to correct myself on one of the points I made in my reply last night.
As a very polite developer from Sangoma stated to me(with evidence I might
add)they have in the past and continue to today contribute code to GPL
Asterisk. It doesn't say so on their website but their developers have been
bug-checking, patching and contributing new code to Asterisk for some time
now. They just started directly giving credit from Sangoma for some of these
contributions in the bugtracker starting this week. While it is true that
they probably don't have as many full-time dedicated Asterisk developers as
Digium does, a portion of a Sangoma AFT card purchase will go towards
further development of Asterisk. So you can feel a little less-bad about
buying those Sangoma cards now.

MATT---

-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 31, 2005 2:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sangoma VS. Digium


Isamar Maia wrote:
 Technically speaking not. But Sangoma's support seems to be pretty much
 better.
 

My understanding is that to an extent when we buy Sangoma we're putting 
the dagger to Digium.  They're glad to use Asterisk as a selling point 
for their hardware, but unwilling to donate anything back to the 
Asterisk community.

I'll be glad to stand corrected, but if that assertion is in fact true, 
we should be careful to do things that actually damage Digium's ability 
to leverage their development of Asterisk with their hardware sales.

FWIW.

b.
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RE: [Asterisk-Users] Sangoma VS. Digium

[mattf]

Here's an idea, Digium buys Sangoma with the massive amounts of cash they
are getting from venture capitalists and just integrate Sangoma designs into
their boards. Not sure how Sangoma would feel about this idea though.

MATT---


-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 31, 2005 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sangoma VS. Digium


Brian Capouch wrote:

 I'll be glad to stand corrected, but if that assertion is in fact
 true, we should be careful to do things that actually damage Digium's
 ability to leverage their development of Asterisk with their hardware
 sales.

It sucks that its such a fine line. On the one had, it is good to have
competition. Keeps prices in check, and gets new features out faster.

But on the other hand, yes, buying from someone else may say to Digium
well, I guess we can stop now that they are buying someone elses cards.

-Matthew

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RE: [Asterisk-Users] Sangoma VS. Digium

[mattf]

Sangoma is a publicly traded company and has a company value of $8 million
canadian. I think that would be a small price to pay to absorb a growing
competitor, enter a new market(data gateways) and gain many product
enhancements.

Digium could easily get funding to do this with their current financers.
Although I realy don't know what complications would come up with a US-based
private company buying a canadian public-traded company.

MATT---


-Original Message-
From: Dana Olson [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 31, 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sangoma VS. Digium


Do you think Digium has enough money for that? I don't know how large
Sangoma is, but they've been around almost as long as I have...

I think it'd be a great idea though, if they have the cash for it..  I
bet it'd pay off too.



On Thu, 31 Mar 2005 11:00:48 -0500, mattf [EMAIL PROTECTED] wrote:
 Here's an idea, Digium buys Sangoma with the massive amounts of cash they
 are getting from venture capitalists and just integrate Sangoma designs
into
 their boards. Not sure how Sangoma would feel about this idea though.
 
 MATT---
 
 
 -Original Message-
 From: Matthew Boehm [mailto:[EMAIL PROTECTED]
 Sent: Thursday, March 31, 2005 10:30 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Sangoma VS. Digium
 
 Brian Capouch wrote:
 
  I'll be glad to stand corrected, but if that assertion is in fact
  true, we should be careful to do things that actually damage Digium's
  ability to leverage their development of Asterisk with their hardware
  sales.
 
 It sucks that its such a fine line. On the one had, it is good to have
 competition. Keeps prices in check, and gets new features out faster.
 
 But on the other hand, yes, buying from someone else may say to Digium
 well, I guess we can stop now that they are buying someone elses cards.
 
 -Matthew
 
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RE: [Asterisk-Users] Asterisk Hardware Requirements for a 50-100 Seat Call Center

[mattf]

You're going to have to go a little more in depth into what you are doing in
this call center.

- Are you going to be doing inbound or outbound? (if so how much of each)
- What kind of phones are you planning on using?
- What is the maximum number of concurrent conversations you plan on having?
- What kind of T1s will you have connected to the server?(PRI,RBS)
- Are you planning on doing a lot of recording?

MATT---

-Original Message-
From: Matt Roth [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 24, 2005 9:09 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Hardware Requirements for a 50-100
Seat Call Center


I am looking for estimated hardware requirements for running a 50 to 100 
seat call center off of a single Asterisk server.

The Asterisk server will have one quad T1 card installed (probably a 
Digium TE410P) with two T1s connected.  The OS is Debian GNU/Linux 
(woody) with a custom 2.4.xx kernel installed.  It is preferable for the 
server to have a single CPU and no shared IRQs.

I would really appreciate any comments regarding the feasibility of this 
scenario, as well as any suggestions about hardware requirements.

Thanks,

Matthew Roth

http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debia
n
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RE: [Asterisk-Users] Asterisk/Zaptel on Mac G5 or Xserve

[mattf]

I've been trying to get a test G5 in our office from Terrasoft for the last
few months. They are very interested and we have offered to give them a
deposit for the machine while we test it for a week, but they don't seem to
have a machine that they want to send us. Anyone else know of another
Asterisk-eager vendor that might lend us a machine for a week for us to test
so we can post our results?

an Xserve wouldn't be the best option because it can only hold one TE405P
card, we would prefer to test on a Dual G5 tower. Then again beggers can't
be choosers and we'd just like any high end G5 mac made in the last 6 months
to test on.

MATT---


-Original Message-
From: Geoff Nordli [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 22, 2005 12:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk/Zaptel on Mac G5 or Xserve




 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C. Tomlinson
 Sent: Tuesday, March 22, 2005 12:11 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Asterisk/Zaptel on Mac G5 or Xserve
 
 Out of interest why use a G5 over an x86 PC? Do you feel the performance
 will be better, or do you just prefer Mac's?
 
 Thanks
 
 C

There has been lots of discussion on the list about PC based hardware
problems.  It seems that the Mac is better designed to handle the IO
requirements demanded of an audio centric application like Asterisk.

I don't know the answer.  I haven't seriously touched a Mac in 20 years.  I
will probably be taking the plunge though.

Geoff  


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RE: [Asterisk-Users] OT: Mexico area codes

[mattf]

Hello,

We created an areacode, country code, GMT offset, country code file for the
astGUIclient project last year. I believe it has all Mexican area codes in
it. If you find any errors we've love to hear about it.

http://astguiclient.sourceforge.net/phone_codes_GMT.txt

Hope this helps,

MATT---

P.S. Yes, I also think they could've done this much better, but what do you
expect from bureaucrats.


-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: Friday, March 18, 2005 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] OT: Mexico area codes


After I have finished to key in the area codes for Mexico I would like 
to propose:

The guy how created the numbers should be stoned to death with the dice 
he created the numbers !!!

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RE: [Asterisk-Users] Manager API - Redirect command

[mattf]

You 
should be able to get the full channel values by doing a "Action: 
Command Command: Show Channels" and 
picking your SIP extension out of the list it gives you of active channels. Then 
you can take that and the channel that you are currently connected to, also 
taken from the "Show channels" output(I am assuming that you want to take both 
parties and dump them into the meetme room) and use those two channel values in 
the Redirect command.

If you 
do it right, neither you nor the person you were talking to would notice that 
you just moved into the meetme room. We use this method with the astGUIclient 
client application to transfer an existing conversation into a meetme room and 
it works great.

Hope 
this helps,

MATT---

  -Original Message-From: Vyom A 
  [mailto:[EMAIL PROTECTED]Sent: Friday, March 18, 2005 8:21 
  AMTo: Asterisk_users_mailing_listSubject: 
  [Asterisk-Users] Manager API - Redirect command
  I read the Wiki pages about the Redirect command, but,if I want to do 
  a redirect into a MeetMe room, from a *remote* machine, how do I *query* 
  Asterisk and get the Channel details?i.e the values for the Channel 
  and ExtraChannel.I am using *SIP only*.Also, when redirected, one end 
  Hangs up. Is this the intendedbehavior? 
  
  
  Do you Yahoo!?Yahoo! Small Business - Try 
  our new resources site! 
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RE: [Asterisk-Users] Caller ID on EM Wink

[mattf]

With a RBS(Robbed-bit) T1(in other words, not a PRI) the CallerID(Called
ANI) is sent in the digits and come across in Asterisk as part of the
extension. It is not standard, you do need to ask for it to be enabled and
you usually have to specify how you want it.

A standard way of receiving ANI on a RBS T1 is this: *NXXNXX*DNIS where
NXXNXX is the callerID and the DNIS is the last 4 digits of the number
the caller dialed.

There is no option of receiving callerIDname with RBS T1s but you do get
that 24th channel to use for voice that you don't get with a PRI.

Hope this helps,

MATT---

-Original Message-
From: Michael Welter [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 17, 2005 4:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Caller ID on EM Wink


Scott Nelson wrote:
 I am an Asterisk newby, and I cannot seem to get Caller ID information 
 from our T1 line.   When calls appear at the phones, they say the call 
 came from asterisk and unknown number.
 
 I know how Caller ID information is passed on an analog phone line 
 (between the rings) but with a T1 line, I don't know technically how it 
 is done.
 
With an EM T-1, I think CallerID number comes across as DTMF digits. 
That's the difference with a PRI circuit--you get the caller name as 
well as caller number.


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RE: [Asterisk-Users] Call Center software opensource or commercia l

[mattf]

Hello,

We tried a Dual Processor AMD system last year and were greatly
dissapointed. A single P4 system was much cheaper and actually outperformed
the Dual AMD. 

Is anyone actually running an octal AMD system out there? 

In our experience having more processors doesn't really matter on the x86
platform because of the limitations of the motherboard bus.

I would love to see real benchmarks for PowerPC/Apple hardware as well as
other Asterisk-capable hardware platforms. And seeing the cost of those
systems would be very helpful as well(to get a price per line ratio on each
system).

MATT---

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 16, 2005 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center software opensource or
commercial


Thanks Kevin for this info,

If we want a box that can perform 60 calls. What would be apoproximate
budget
for that using AMD x86-64 ?

µSelon Kevin P. Fleming [EMAIL PROTECTED]:

 Erick Perez wrote:
  And what people are using to deploy super servers with astersik?
  Itanium with linux? clusters of itanium with linux? or some RISC
  processor with some *nix? cause it seems asterisk is only 100%
  supported on Linux/Intel
  or am i totally wrong?

 The highest-performing standard hardware to run Asterisk on today
 would be quad/octal Opteron (AMD X86-64) boxes.

 In fact, hardware like that will very likely outperform the Altix system
 that Signate did their benchmarking on, for quite a lot less money.
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RE: [Asterisk-Users] Call Center software opensource or commercia l

[mattf]

Hello,

We use and develop the astGUIclient suite. It is Open-source(as in GPL) and
offers Inbound and Outbound call center functions with reports, ACD,
monitoring, recording and very basic IVR scripts. Complex IVR functions need
to be custom programmed within Asterisk but that is not really that hard. It
works across multiple Asterisk servers and we are using it currently at 5
locations including our main office which has over 100 agent seats.

http://astguiclient.sf.net/

There is also Aheeva- http://www.aheeva.com/ for a commercial Asterisk call
center solution that offers a ton of functionality for a price.

That's about all I know of for full Asterisk call center suites.

Let me know if you have any questions about astGUIclient.

MATT---



-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 15, 2005 10:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Call Center software opensource or commercial


Hi there, we are looking for an opensource or commercial * based Call
Center.
Full ACD, call monitoring, multiple queue, IVR, voicemail, management,
reporting, CDR, etc is needed. over 100 seat can be the initial target
and will grow in a very short time.

SIP phones will be used and multiple E1 lines incoming, so to provide
full failover a cluster of * machines or some other form of redundancy
must be used.

I'm sure custom programming will be requiered so offerings are
accepted but all work will be done remotely since we are in Central
America (unless you happen to live in our country of course...)

Any real experiences with * on this?

please for commercial offer reply off-list to [EMAIL PROTECTED] since
I think the rules of this forum prohibits commercial offerings.

So far i have found
http://www.aspect.com/
http://www.ebiitech.com/

Thanks in advance,


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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[Asterisk-Users] New astGUIclient version released 1.1.0

[mattf]

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.0

http://astguiclient.sf.net/

Screen shots: http://astguiclient.sourceforge.net/screenshots.html

The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or Zap
phones and Zaptel or IAX trunks.

In addition to many bug fixes and some major changes on the back end, we've
added more documentation to the web admin pages(demo here:
http://astguiclient.sourceforge.net/admin_demo.html) and the ability for
remote users of VICIDIAL to have calls sent to them wherever they have a
phone.

To upgrade from 1.0.6 you need to update all of the server apps and web
pages as well as use the new client apps(older client apps will not run on
this version), and run the update sql script.

Let me know what you think.

Thanks,

MATT---
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RE: [Asterisk-Users] Asterisk Management API

[mattf]

The best way to figure out the manager protocols is through looking at the
manager.c source code and trial and error.

Some things just don't behave the way you think they should, some things are
not fully documented and some actions do not work in certain cercumstances
while others will.

And when you figure something new out, please put it in the Wiki, I've added
a lot to the Wiki in the manager API section and it took me quite a while to
figure out some of it.

Good luck,

MATT---


-Original Message-
From: Umar Sear [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 08, 2005 11:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk Management API


Hi all, 

I am trying to write an application to monitor queues using the
Asterisk Management API.

So far I have had some level of sucess, basically reverse engineering
the protocol and the event messages using ethereal etc.

I know there are a couple of pages on the Wiki that attempt (no
dis-respect to who ever did it as it has been a great help)  to
document the API and was wondering if there is more information
available.

Any pointers will be greatly appreciated. I hope to document my
findings on the Wiki once I have definative information.

Thanks

Umar
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RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls

[mattf]

Hello,
You can do either, you can send multiple Originate actions in a long line
without waiting for a response back(although the responses do usually come
back very fast) or you can open multiple connections using each one to
Originate a new call. We use the multiple connection method in the
astGUIclient suite because if you get a pause or lag in Manager output on a
single connection(which does happen) it will hold up all of the Actions you
are trying to send after it.

Take a look at the ACQS(Asterisk Central Queue System) part of the
astGUIclient suite. It allows you to queue up Actions in a database and the
server will send the actions to the asterisk server almost immediately.
We've been using this for quite a while now and it is very reliable.

MATT---


-Original Message-
From: Stephen Owen hosted [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 02, 2005 7:28 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls


Been researching connecting over TCP\IP to Asterisk Manager API to initiate
several concurrent calls to dial out. Prefer not to generate ASCII .call
files.
 
Question : I read in places that you use originate command and wait for an
event back, does that mean you cannot place another originate until the
event comes back ?
 
Is it true that multiple API connections to Asterisk Manager API will crash
it (thinking of alternative way to crack the nut)
 
All help would be welcome - thanks
 
Stephen Owen
 
sip:[EMAIL PROTECTED]
IM:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls

[mattf]

ActionID does not return in all events related to an 
Action sent, sometimes it will just send you a success message and nothing more. 
Just try Originating a call from a meetme room over an outside line. You will 
get about 150 lines of output and only one message will have the ActionID in it, 
the success message. On the other hand the callerID is placed on many more of 
the events in the output. It is still the case that if you do complex Manager 
Actions, the ONLY solution for tracking a call is to use a custom 
CallerID.

Action: OriginateExten: 8600080Channel: 
local/[EMAIL PROTECTED]Context: defaultPriority: 1Callerid: 
DF345678901234567890Actionid: AID45678901234567890

MATT---


  -Original Message-From: Bill Seddon 
  [mailto:[EMAIL PROTECTED]Sent: Wednesday, March 02, 2005 
  8:06 AMTo: Stephen Owen hosted; Asterisk Users Mailing List - 
  Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 
  Manager API - multi "Originate" calls
  
   read 
  inplaces that you use "originate" command and wait for an event back, 
  does that mean you cannot place another "originate" until the event comes back 
  ?
  
  Not in my experience. 
  Originate will not send an event to the caller until either the intended 
  caller (that is the extension used in Originate) has picked up their phone or 
  a timeout occurs because the intended caller does not pick up their 
  phone. You can send as many originate requests as you like but they will 
  fail if more than one uses the same extension at the same 
  time.
  
  The issue you will face is 
  determining which event generated by Asterisk belongs to which origination 
  request. For this reason, the Manager API allows you to specify an 
  ActionID on any command. An ActionID is any string of characters that you 
  use to uniquely identify each command use issue. Asterisk will include 
  the ActionID with each related event so you know which events to respond to 
  and which to ignore. You will see many events generated by Asterisk only 
  some of which will relate to your command. The others will be events 
  that Asterisk raises (for example when a phone registers) or events in 
  response to commands issues by other Manager API users and at the command 
  line.
  
  Take a look at Nicolas Gudinos 
  Flash Operator Panel (www.asternic.org) 
  as it used the manager API extensively (albeit through a proxy) and will 
  typically make many requests via the Manger API. 
  
  Is it true that multiple 
  API connections to Asterisk Manager API will crash it (thinking of alternative 
  way to crack the nut)
  
  Again, not in my experience. 
  
  
  
  Lyquidity Solutions 
  Limited 
  +44 (0) 208 241 
  0500 
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Owen 
  hostedSent: March 02, 2005 
  12:28 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk 
  Manager API - multi "Originate" calls
  
  
  Been researching connecting over 
  TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial 
  out. Prefer not to generate ASCII .call files.
  
  
  
  Question : I read inplaces 
  that you use "originate" command and wait for an event back, does that mean 
  you cannot place another "originate" until the event comes back 
  ?
  
  
  
  Is it true that multiple API 
  connections to Asterisk Manager API will crash it (thinking of alternative way 
  to crack the nut)
  
  
  
  All help would be welcome - 
  thanks
  
  
  
  Stephen 
  Owen
  
  
  
  sip:[EMAIL PROTECTED]IM:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls

[mattf]

Well, I'm not sure about the current release as I have not tested this, but
on older releases for RBS T1s you would get a manager event showing a RING
state. As for PRI, SIP and IAX2 I'm not sure, this is an inconsistent
feature that differes depending on what kind of trunk you are using and what
network the person you are calling is on. The only sure thing you can tell
is a call pickup in all cases, ringing is much harder to detect.

MATT---


-Original Message-
From: Thomas Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 02, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate
cal ls


Hi Matt, in your experience is there a 100% reliable
way to know that the callee phone is ringing? In my
situation I don't need to know if they pick up or not,
I need to know (as reliably as possible) if the calee
phone number is ringing.

Thanks, Tom


--- mattf [EMAIL PROTECTED] wrote:

 ActionID does not return in all events related to an
 Action sent, sometimes
 it will just send you a success message and nothing
 more. Just try
 Originating a call from a meetme room over an
 outside line. You will get
 about 150 lines of output and only one message will
 have the ActionID in it,
 the success message. On the other hand the callerID
 is placed on many more
 of the events in the output. It is still the case
 that if you do complex
 Manager Actions, the ONLY solution for tracking a
 call is to use a custom
 CallerID.
  
 Action: Originate
 Exten: 8600080
 Channel: local/[EMAIL PROTECTED]
 Context: default
 Priority: 1
 Callerid: DF345678901234567890
 Actionid: AID45678901234567890
 
  
 MATT---
  
 
 -Original Message-
 From: Bill Seddon [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, March 02, 2005 8:06 AM
 To: Stephen Owen hosted; Asterisk Users Mailing List
 - Non-Commercial
 Discussion
 Subject: RE: [Asterisk-Users] Asterisk Manager API -
 multi Originate calls
 
 
 
  read in places that you use originate command
 and wait for an event
 back, does that mean you cannot place another
 originate until the event
 comes back ?
 
  
 
 Not in my experience.  Originate will not send an
 event to the caller until
 either the intended caller (that is the extension
 used in Originate) has
 picked up their phone or a timeout occurs because
 the intended caller does
 not pick up their phone.  You can send as many
 originate requests as you
 like but they will fail if more than one uses the
 same extension at the same
 time.
 
  
 
 The issue you will face is determining which event
 generated by Asterisk
 belongs to which origination request.  For this
 reason, the Manager API
 allows you to specify an ActionID on any command.
 An ActionID is any
 string of characters that you use to uniquely
 identify each command use
 issue.  Asterisk will include the ActionID with each
 related event so you
 know which events to respond to and which to ignore.
  You will see many
 events generated by Asterisk only some of which will
 relate to your command.
 The others will be events that Asterisk raises (for
 example when a phone
 registers) or events in response to commands issues
 by other Manager API
 users and at the command line.
 
  
 
 Take a look at Nicolas Gudino's Flash Operator Panel
 ( www.asternic.org
 http://www.asternic.org/ ) as it used the manager
 API extensively (albeit
 through a proxy) and will typically make many
 requests via the Manger API.  
 
  
 
 Is it true that multiple API connections to
 Asterisk Manager API will
 crash it (thinking of alternative way to crack the
 nut)
 
  
 
 Again, not in my experience.  
 
  
 
 Lyquidity Solutions Limited 
 +44 (0) 208 241 0500 
 
 
   _  
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Stephen Owen
 hosted
 Sent: March 02, 2005 12:28 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Asterisk Manager API -
 multi Originate calls
 
  
 
 Been researching connecting over TCP\IP to Asterisk
 Manager API to initiate
 several concurrent calls to dial out. Prefer not to
 generate ASCII .call
 files.
 
  
 
 Question : I read in places that you use originate
 command and wait for an
 event back, does that mean you cannot place another
 originate until the
 event comes back ?
 
  
 
 Is it true that multiple API connections to Asterisk
 Manager API will crash
 it (thinking of alternative way to crack the nut)
 
  
 
 All help would be welcome - thanks
 
  
 
 Stephen Owen
 
  
 
 sip:[EMAIL PROTECTED]
 IM:[EMAIL PROTECTED]
 
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RE: [Asterisk-Users] Asterisk URL and Callcenter Apps

[mattf]

We use astGUIclient suite, it has this functionality. Hard or soft phones
SIP, IAX or Zap
http://astguiclient.sf.net

MATT---


-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 02, 2005 4:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk URL and Callcenter Apps


Guys.

How do those callcenter apps work with Asterisk where a call comes in and *
send a URL and some screen popup up based on callerid or something or
username or id and shows all the customers info?

Anybody done that? What do you need to do that? 
If you are using ATAs or IP Phones, how do those integrate with the PC so
the screen would popup?


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RE: [Asterisk-Users] astguiclient gives me Object not found

[mattf]

Hello,

The astGUiclient suite has it's own mailing list for questions like this:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users

The easy fix is for you to set PHP globals to on and see if it works like
that first, also you could try making that directory writable.

MATT---


-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: Sunday, February 27, 2005 3:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] astguiclient gives me Object not found


I tried to install astguiclient and it gives me for each follow page:

Object not found!

Looking into the apache log file I find:
[Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does 
not exist: /srv/www/htdocs/astguiclient/method=POST, referer: 
http://vpbx.elmit.com/astguiclient/phone_stats.php?extension=gs102server_ip
=192.168.250.20

I have uncommented the
### If you have globals turned off uncomment these lines
$PHP_AUTH_USER=$_SERVER['PHP_AUTH_USER'];
$PHP_AUTH_PW=$_SERVER['PHP_AUTH_PW'];
$ADD=$_GET[ADD];
### AST GUI database administration
### admin.php

How can I fix it?


bye

Ronald

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RE: [Asterisk-Users] listening to gsm files

[mattf]

The free utility WavePad for Win32 will play and edit GSM files as well:
http://www.nch.com.au/wavepad/

To convert to/from GSM on Win32 you can use DBpowerAMP:
http://www.dbpoweramp.com/dmc.htm

And for Linux or Win32 you could use Sox of course:
http://sox.sourceforge.net/

MATT---



-Original Message-
From: Time Bandit [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 26, 2005 8:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] listening to gsm files


 Anybody knows what one should do to listen to GSM files?
I know QuickTime can play gsm files. Maybe your users that succeeded
had it installed.

hth
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RE: [Asterisk-Users] Wierd asterisk-perl compilation problem

[mattf]

Hello,

A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at
least not use rpms or the preinstalled perl on the OS. RedHat has done a lot
to screw up how perl works in the last several versions and there are a lot
of angry perl developers that have just given up on the distro altogether.

I suggest using another ditro(I know that's a little drastic, but you'll be
better off in the long run) or at least install ActivePerl from ActiveState
or download perl source and compile it on your system and use that.

I use Slackware now and have no problems with perl or asterisk-perl on stock
installs.


MATT---


-Original Message-
From: David Carroll [mailto:[EMAIL PROTECTED]
Sent: Sunday, February 27, 2005 12:25 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Wierd asterisk-perl compilation problem


On Sat, 2005-02-26 at 23:19 -0600, David Carroll wrote:
 I am running a fully updated Fedora Core 3 server, and installed a
 pretty thin system, and have just installed packages as needed.
 
 My problem is that I am trying to get asterisk-perl installed, but it
 keeps segmentation faulting on me.  I know a little python but perl
 baffles me.
 
 # perl Makefile.PL
 Segmentation fault
 ==Strace

Oops, accidentally hit send instead of attach :).

I'm attaching the strace and the env outputs to see if that helps
someone figure out what I have going on wrong.  Asterisk is up and going
great, I just can't seem to figure out what package I'm missing, or what
is broken by the output.

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RE: [Asterisk-Users] Asterisk-HEAD more stable than Asterisk-1.0. 5

[mattf]

We are running HEAD from last night and 1.0.5 and 1.0.3 and 1.0.2 and they
all are running just fine in production environments each handling thousands
of calls a day.

I suppose reliability depends upon what you are using, but for our purposes
they all are very stable. I could do without the memory leaks though.

MATT---


-Original Message-
From: Florian Lefeuvre [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk-HEAD more stable than Asterisk-1.0.5


hi everyone,

just a poll toknow if someone out there is using intensively 
asterisk-HEAD version (mean the very last version
of asterisk).
I currently used asterisk.1.0.5 and sometimes I need to kill the process 
because it's freezing (deadlock maybe, or something else...).
is this kind of problem occurs to someone else?
is this problem disapears with the HEAD version?

Thanks

Florian



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[Asterisk-Users] Zap call bridge drops randomly

[mattf]

Hello,

We have a call redirection system setup inhouse to send calls from an
incoming line on a T1 to an external dialed out number:
  Zap(call comes in) - Asterisk - Zap(call dials out)

The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
We are using Asterisk release 1.0.5

Randomly the calls will drop less than a minute into the call. The Debug
messages at the end of the call always say something like this: (incoming
call on 58, outgoing on 73) 

Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58
Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0)
Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58
Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 

The strange thing is that the person that called in did not hang up, in fact
they are usually talking when this call goes dead. Only about 10% of the
calls that redirect have this happen to them, and it seems to be random as
to which ones drop. Calls that dial out on either T1 and calls that come in
and are not redirected never seem to have these problems.

I have callprogress=no and busydetect=no but that doesn't seem to help.

Anyone have an idea on this?

Is there any way to make Asterisk less sensitive to hangups if that's even
the cause?

Just looking for some feedback before I post on the bugtracker.

Thanks,

MATT---
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RE: [Asterisk-Users] Zap call bridge drops randomly

[mattf]

Would enabling Busydetect really help if Asterisk thinks it detects an
On-Hook?

MATT---

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap call bridge drops randomly


 We have a call redirection system setup inhouse to send calls from an
 incoming line on a T1 to an external dialed out number:
   Zap(call comes in) - Asterisk - Zap(call dials out)
 
 The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a
PRI.
 We are using Asterisk release 1.0.5
 
 Randomly the calls will drop less than a minute into the call. The Debug
 messages at the end of the call always say something like this: (incoming
 call on 58, outgoing on 73) 
 
 Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58
 Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0)
 Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58
 Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 
 
 The strange thing is that the person that called in did not hang up, in
fact
 they are usually talking when this call goes dead. Only about 10% of the
 calls that redirect have this happen to them, and it seems to be random as
 to which ones drop. Calls that dial out on either T1 and calls that come
in
 and are not redirected never seem to have these problems.
 
 I have callprogress=no and busydetect=no but that doesn't seem to help.
 
 Anyone have an idea on this?
 
 Is there any way to make Asterisk less sensitive to hangups if that's even
 the cause?
 
 Just looking for some feedback before I post on the bugtracker.

You might try:
busydetect=yes
busycount=6

in the top section of zapata.conf and see if that helps. I've not
tried this on a T1, but it certainly clears up the same issue on
the TDM pstn lnterfaces.


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[Asterisk-Users] PRI and echocancel

[mattf]

Hello,

I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1
line and currently have echocancel=yes and echocancelwhenbridged=yes on
those spans in zapata.conf. I was discussing CPU load with another Asterisk
user and he mentioned that PRIs don't need echo cancelation and that turning
it off will reduced CPU load on the server. I checked many sample configs
and the archives and noticed that half of the people have echocancel on for
PRIs and half do not. I checked the Digium site and indeed in the FAQ they
say: There should also be no echo on PRI connections. 

Does it really matter that much in terms of CPU usage and will it hurt at
all if I turn it off for the crossover PBX connection or the telco PRI that
I have? 

Thanks,

MATT---
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RE: [Asterisk-Users] Monitoring Conferences

[mattf]

Use the manager API to send a call from the meetme room to an extension that
does Monitor for a specified period of time. That is how we do it in the
astGUIclient suite and it works great.

; extensions.conf entry:
; this is used for recording conference calls, the client app sends the
filename
;value as a callerID, recordings go to /var/spool/asterisk/monitor
exten = 8309,1,Answer
exten = 8309,2,Monitor(wav,${CALLERIDNUM})
exten = 8309,3,Wait,3600
exten = 8309,4,Hangup


Manager Command:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: default
Extension: 8309
Priority: 1
CallerID: FilenameGoesHere1234


MATT---

-Original Message-
From: Michael Blood [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 16, 2005 8:25 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitoring Conferences


I have benn having trouble with the Monitor Command.
 
Basically any time that I send a call into a MeetMe room I am only able to
monitor half of the conversation.
File-in is recorded with the incoming voice but file-out does NOT record
anything.
I have tried this with both the b and m option as well as without any
options to the MeetMe command.
Also the Monitor correctly records both sides of the channel when I do not
send the channel to a MeetMe room (send them to another extension).
 
The only thing I can think may be wrong is that for some reason the call is
never considered bridged when I send the call directly to a MeetMe room.
 
Any Ideas?  Are there any ways to record a MeetMe room when there is more
than one channel connected?
 
Thanks
 
Michael
 
 
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RE: [Asterisk-Users] Monitoring Conferences

[mattf]

In our client apps, we track the call placed and Hangup that call when
conference is over. All you need to do is either have a function that hangs
up those recording channels if they are the only one in the conference(perl
script running periodically parsing Show Channels) Or you could link a
button on an app or web page to a function that would hangup the channels
that are connected to a specific meetme room.

It's not the easiest thing to program, but it always works and it is the
only reliable way that Asterisk lets you record conferences. Our company has
done hundreds of thousands of recordings this way over the last 2 years.

MATT---


-Original Message-
From: dean collins [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 16, 2005 10:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Monitoring Conferences


Matt,
How do you stop the recording if it is set for a period of time? Eg if
set the period as 30 minutes and the call finishes early will it cease
recording or hold up the line for 30 mins



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Wednesday, February 16, 2005 10:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Monitoring Conferences

Use the manager API to send a call from the meetme room to an extension
that
does Monitor for a specified period of time. That is how we do it in the
astGUIclient suite and it works great.

; extensions.conf entry:
; this is used for recording conference calls, the client app sends the
filename
;value as a callerID, recordings go to /var/spool/asterisk/monitor
exten = 8309,1,Answer
exten = 8309,2,Monitor(wav,${CALLERIDNUM})
exten = 8309,3,Wait,3600
exten = 8309,4,Hangup


Manager Command:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: default
Extension: 8309
Priority: 1
CallerID: FilenameGoesHere1234


MATT---

-Original Message-
From: Michael Blood [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 16, 2005 8:25 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitoring Conferences


I have benn having trouble with the Monitor Command.
 
Basically any time that I send a call into a MeetMe room I am only able
to
monitor half of the conversation.
File-in is recorded with the incoming voice but file-out does NOT record
anything.
I have tried this with both the b and m option as well as without any
options to the MeetMe command.
Also the Monitor correctly records both sides of the channel when I do
not
send the channel to a MeetMe room (send them to another extension).
 
The only thing I can think may be wrong is that for some reason the call
is
never considered bridged when I send the call directly to a MeetMe
room.
 
Any Ideas?  Are there any ways to record a MeetMe room when there is
more
than one channel connected?
 
Thanks
 
Michael
 
 
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RE: [Asterisk-Users] asterisk GUI's that supports zap fxs extensi ons

[mattf]

by GUI do you mean a configuration utility or a User Interface?

MATT---


-Original Message-
From: Jon Gabrielson [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 10, 2005 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] asterisk GUI's that supports zap fxs
extensions


Are there any gui's that support zap fxs extensions?
AMP seems to be one of the more popular gui's but
it doesn't support zap fxs devices.


Thanks,


Jon.
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RE: [Asterisk-Users] Callerid problems with 1.0.5

[mattf]

http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003490

apply the patch:  app_dial_CID_nodelete.patch  
and the deleting of the original callerid will stop in v1.0.5.

Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.


MATT---


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 11:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Callerid problems with 1.0.5


I just upgraded my two asterisk boxes to 1.0.5 stable and I've noticed
that callerid is not functioning properly. My setup looks like this:

SIP Phone -- SER -- Asterisk -- Asterisk --- PSTN

No iax is being used at this time.

The problem can be best described by the following scenarios:

1.) SIP to PSTN call: When a SIP phone calls a PSTN bound number, the
callerid displayed on the PSTN phone is the number of the PSTN phone
instead of the SIP phone's number.

2.) PSTN to SIP call: When a PSTN phone calls a SIP Phone number, the
callerid displayed on the SIP phone is the number of the SIP phone instead
of the PSTN phone's number.

For both scenarios - ${CALLERID}, ${EXTEN}, and ${CALLERIDNUM} all have the
number of the called phone for ZAP to SIP, SIP to ZAP, and SIP to SIP. I
have noticed that explicitly declaring SetCallerID(${CALLERID}) before my
dial seems to fixe this issue for only the ZAP to SIP piece. In the next
Asterisk where a SIP to SIP relay is occurring ${CALLERID} ends up matchign
${EXTEN} again.
This is causing some havoc with users calling cell phone from SIP phones.
Some users are being dumped into certain company's cell phone voicemail
because the callerid is keyed to the called phone's number.

Has anyone else experienced this problem with 1.0.5 stable? I checked the
bugs.digum.com page and found a similar bug with regard to the call being
delivered to the manager API. Also, I searched the configs and I did not
see any new settings related to callerid. If this is a simple
configuration change introduced into version 1.0.5, any info would be
greatly appreciated.
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RE: [Asterisk-Users] Callerid problems with 1.0.5

[mattf]

Supposedly when a call is parked and/or transferred they wanted the callerid
to reflect the person who is on that phone call. That's the only reason I
saw mentioned for the change. As for why it was made default I have no idea,
but now in CVS_HEAD at least you can turn that feature off. And in v1.0.5
you can patch your system to remove that feature.

MATT---


-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5


mattf wrote:

 Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
 the dial string.

I have to wonder why the default behavior was changed to this 
non-standard usage though; in what situations do we want the CLID/CNAM 
of the _recipient_ to be passed to them?
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RE: [Asterisk-Users] vicidial and mysql ........help

[mattf]

Hello,

First, there is a mailing list for the astGUIclient suite:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users

As for your problem, If you have everything set up correctly you should just
be able to run the AST_VDhopper.pl script from your Asterisk server to fill
your lead hopper. Make sure you have the dialing restrictions in the
campaign screen set to 24 hours for testing and the dial level set to 1 or
higher and the hopper level to 1 or greater.

If you are still having problems, send the output of the AST_VDhopper script
to the astguiclient-users list and you'll get some help.

MATT---

-Original Message-
From: Hussain Umair [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 4:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] vicidial and mysql help


hi all.well ive installed each and everything according to the scratch 
installation but the problem is when i try to login a user through 
vicidialgui application it gives an error that there are 0 leads in the 
hopper to dialwell im pasting the result of a few queries .plz if 
any one can do help me out 
this is what the error looks like...

SELECT count(*) FROM vicidial_hopper where campaign_id = '13' and 
status='READY';
0 - leads left to call in hopper
so the login fails..


but these queries at the mysql cmd line tell me i got 7 leads in the 
database

select * from vicidial_list;
7 rows in set (0.00 sec)


but when i use count it gives this
mysqlselect count(1) from vicidial_hopper;
+--+
| count(1) |
+--+
|0 |
+--+
1 row in set (0.00 sec)

mysql select status,count(1) from vicidial_hopper group by status;
Empty set (0.00 sec)

mysql select campaign_id from vicidial_hopper group by campaign_id;
Empty set (0.00 sec)


please any ideas or help woud be greatly appreciatedim kinda lost in 
this mysql lala land cant figure out whats going on.help me out here 
guys thankss

kurt...
Network Engineer plus Asterisk Newbie u can say

_
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RE: [Asterisk-Users] Callerid problems with 1.0.5

[mattf]

Hello,

patching v1.0.5 on my system removed the problem for me. But yes it seems
strange that this feature was inserted into a final release with very little
documentation of the wide implications that are caused by the change.

This was corrected in CVS with the addition of a diabling flag for the dial
command, but maybe this is a message that we should start an official beta
release period before a release so that people can test pre-releases even
for just a week to report problems before it is unleashed upon the world as
an official release

MATT---


-Original Message-
From: Mark Eissler [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 9:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5


Yikes!

On Feb 4, 2005, at 1:26 PM, Jay Milk wrote:

 Can someone clarify what's going on here?

 I'm running 1.0.5, and I see caller-id come through just fine from one
 extension to the other, as well as for incoming and outgoing calls
 (iax2).  What are you folks seeing there?


The behavior that was reported by Kevin is/was exactly the same 
behavior that I was experiencing with 1.0.5 and reported in another 
thread. I switched back to 1.0.2 to resolve that problem and another I 
was experiencing (SIP calls ringing forever instead of disconnecting 
even when voicemail had already picked up).

Reading through the bug tracker on this one I must say I'm a bit 
confused. I understand the concept of showing useful/relevant callerid 
when a call is transferred (from park or some other extension) but I 
don't understand why a call should ever show the recipient extension's 
callerid. My understanding is that this is the default behavior when no 
other callerid is present and for some reason inbound callerid is 
getting wiped out because it's not correct.

That some people are experiencing problems with this while others are 
not leads me to believe that it might be a problem that is exacerbated 
depending upon the dialplan setup. I'm just thinking this at the top of 
my head now, haven't looked back at my dialplan yet.

What's annoying, either way, is that when this change was made the 
behavior of existing, functioning setups broke. I don't recall seeing 
any documentation for 1.0.5 that noted this might be the case and if 
the documentation is lacking...well, that's a problem.

-mark

--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com

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RE: [Asterisk-Users] Re: astGUIclient users should not upgrade to Asterisk 1.0.5

[mattf]

I have added a simple patch to the bugnote for this issue:
http://bugs.digium.com/bug_view_page.php?bug_id=0003490

All it really does is delete the code in app_dial.c that wipes out the
callerID. But astGUIclient now runs properly on Asterisk 1.0.5 with this
patch applied. I will also post the patch on the astGUIclient web site.

Still I do believe that this feature is not a bad one, just very poorly
implemented. It really should be an OPTIONAL dial flag not a manditory
hard-coded feature.

MATT---


-Original Message-
From: Nicolás Gudiño [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5


Hello,

 I'm not a astGUIclient user, but I'm puzzled by the following statement:
 
 mattf [EMAIL PROTECTED] wrote:
 
  In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
  once the call picks up, Asterisk will change the callerid to the number
that
  you just dialed, no matter if you set a custom callerID for that call.
 
 What you've said there suggests that the CallerID is being set to the
 DESTINATION number, which sounds to me not what CallerID should be at all.
 CallerID normally indicates the source of a call.

Just wanted to say that Flash Operator Panel users will have the same
problem. I'm puzzled too. IMHO there's something missing or wrong in
the new callerid handling.

If you trace the manager events and try to match the callerid via
Uniqueid, you will notice that the only way to have a match is *after*
the call is bridged. That means that you cannot  find the callerid of
a call before you pick up the phone. At least thats what I'm seing on
Asterisk 1.0.5. (did not try with HEAD) So, the callerid is plain
useless (Users expect to see the callerid before picking it up, dont't
they?)

It would be nice to have the callerid available on the manager when a
phone is RINGING and before picking it up.

I did not look at the Local channels, and it seems that it makes
things harder.. but I still think that we do not have to code
workarounds on manager based applications. We need to have an event in
the manager informing the callerid of the caller in the RINGING event
or associated directly with the Uniqueid of the callee.

Personally I had to downgrade app_dial.c to a previous releaes to get
the callerid as before.

Just my 2 cents...


-- 
Nicolás Gudiño
Buenos Aires - Argentina
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RE: [Asterisk-Users] Re: astGUIclient users should not upgrade to Asterisk 1.0.5

[mattf]

Hello,

Mark has changed app_dial.c in CVS-HEAD to allow for a 'p' dial flag that
will allow for the original callerid to not be altered. If you are using
CVS-HEAD, simply put the p flag at the end of your Dial string in your
extensions.conf file for 91NXXNXX Dial TRUNK step(or wherever you dial
out). Or if you are using Asterisk 1.0.5 simply use the patch mentioned
before to eliminate callerid altering completely.

Thanks Mark!

MATT---


-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 12:32 PM
To: 'Nicolás Gudiño'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5


I have added a simple patch to the bugnote for this issue:
http://bugs.digium.com/bug_view_page.php?bug_id=0003490

All it really does is delete the code in app_dial.c that wipes out the
callerID. But astGUIclient now runs properly on Asterisk 1.0.5 with this
patch applied. I will also post the patch on the astGUIclient web site.

Still I do believe that this feature is not a bad one, just very poorly
implemented. It really should be an OPTIONAL dial flag not a manditory
hard-coded feature.

MATT---


-Original Message-
From: Nicolás Gudiño [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5


Hello,

 I'm not a astGUIclient user, but I'm puzzled by the following statement:
 
 mattf [EMAIL PROTECTED] wrote:
 
  In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
  once the call picks up, Asterisk will change the callerid to the number
that
  you just dialed, no matter if you set a custom callerID for that call.
 
 What you've said there suggests that the CallerID is being set to the
 DESTINATION number, which sounds to me not what CallerID should be at all.
 CallerID normally indicates the source of a call.

Just wanted to say that Flash Operator Panel users will have the same
problem. I'm puzzled too. IMHO there's something missing or wrong in
the new callerid handling.

If you trace the manager events and try to match the callerid via
Uniqueid, you will notice that the only way to have a match is *after*
the call is bridged. That means that you cannot  find the callerid of
a call before you pick up the phone. At least thats what I'm seing on
Asterisk 1.0.5. (did not try with HEAD) So, the callerid is plain
useless (Users expect to see the callerid before picking it up, dont't
they?)

It would be nice to have the callerid available on the manager when a
phone is RINGING and before picking it up.

I did not look at the Local channels, and it seems that it makes
things harder.. but I still think that we do not have to code
workarounds on manager based applications. We need to have an event in
the manager informing the callerid of the caller in the RINGING event
or associated directly with the Uniqueid of the callee.

Personally I had to downgrade app_dial.c to a previous releaes to get
the callerid as before.

Just my 2 cents...


-- 
Nicolás Gudiño
Buenos Aires - Argentina
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[Asterisk-Users] astGUIclient users should not upgrade to Asterisk 1.0.5

[mattf]

Hello,

Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find ways
around the new callerID rules that the asterisk developers have put in place
and hope to have the next release version of the suite be compatible with
asterisk v1.0.5

Here's a more in-depth explanation of the problem:

In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
once the call picks up, Asterisk will change the callerid to the number that
you just dialed, no matter if you set a custom callerID for that call. When
the callerID is changed it is noted in the manager output only by a uniqueid
of the call. Sadly this does not help very much when you place calls to the
Local/ trunk, which can often result in 3 separate uniqueids for a single
call. That used to only be fully traceable with the custom callerID that you
used to be able to define when the call was started. Now we will need to
figure out some way of tracing all uniqueIDs for every call and try to
determine which call instance a specific call event is referring to even
though it may now have the same callerID. I am also hoping that we may be
able to get asterisk to keep a custom callerid value in the calleridname
field. If all else fails, we may have to resort to including an asterisk
source patch that would reinstate custom callerid values throughout a call's
life, in effect disabling some of these new callerid changes.

Thanks,

MATT---
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RE: [Asterisk-Users] Single or Dual Processor? High volume MeetM e

[mattf]

I'm trying to get a souped-up test machine(G5 Xserve) from Terrasoft to do
some testing in a few weeks. If/when I actually get it I'll certainly post
the results here.

In theory the G5 should mop the floor with the Intel for high-volume
Asterisk Zaptel usage, and I have heard from several Mac-heads that they
have run three quad T1 digium cards on the Mac platform with no problems.
That's why I'd like to test this myself and see if it's worth the extra $.

MATT---


-Original Message-
From: Geoff Nordli [mailto:[EMAIL PROTECTED]
Sent: Monday, January 31, 2005 5:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Single or Dual Processor? High volume
MeetMe




 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of William Boehlke
 Sent: Monday, January 31, 2005 2:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Single or Dual Processor? High volume MeetMe
 
 On Intel it is our experience that the constraint is the PC bus.
 Throughput
 tops out at somewhere between 50 and 100 calls depending on disk speed,
 without ever using a meaningful part of one processor.
 
 William Boehlke
 Signate
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Spencer
 Nassar
 Sent: Sunday, January 30, 2005 11:21 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Single or Dual Processor? High volume MeetMe
 
 
 Has anyone benchmarked Asterisk on a dedicated single versus dual
 processor machine?  Or could any Asterisk developers comment on whether
 it is architected in such a way that threads could run on multiple CPUs
 (especially MeetMe2)?
 
 At a higher level, can I host more simultaneous lines and/or
 conferences for MeetMe if I use a dual processor machine versus single?
   Also, any info on memory use with high numbers of conference users
 (100, 1000)?
 
 Thanks!
 

Has anyone done any testing with the Apple X Serve G5
(http://www.apple.com/ca/xserve/) or Sun
(http://www.sun.com/servers/entry/v20z/index.jsp) Sun Fire machines?

I would think that something like the Apple could handle the IO a lot
better.

Geoff

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RE: [Asterisk-Users] Re: Polycom phones

[mattf]

Hello,

When I talked with the VP of VOIP phone sales at Polycom about a year ago,
he was offering a dedicated engineer for the Asterisk community that would
work through issues like people have here. BUT they would ONLY do this if a
reseller came forward and committed to be the Polycom authorized reseller to
Asterisk users and the dedicated engineer at Polycom would only talk to
people at that reseller company. After wasting 3 months unsuccessfully
trying to convince a half dozen resellers to be that company I gave up. That
reseller would have to raise the prices to be able to support the phones,
and as long as you can buy the Polycom phones at a lower cost from over a
dozen companies that company would be losing a lot of business.

I didn't fault anyone at the reseller companies I talked to, they would be
taking a risk by dedicating resources to supporting Asterisk with no
guarantee of exclusivity from Polycom. The problem is with Polycom, they
make about as much by Cisco selling a phone(from licensing of Polycom
technology) as they do by selling their own phones(with many other costs and
liabilities associated with it). They have no real incentive to go all out
and compete with their biggest customers(Cisco, Avaya, etc...) So they don't
try to make their phones a mass-market item. It's a shame, because I really
like the Polycom phones and have several in our office. If someone else
wants to take up the fight, contact me off list and I'll send you my
contacts.

MATT---


-Original Message-
From: Walt Reed [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom phones


On Thu, Jan 27, 2005 at 11:30:26AM -0500, Kanuri, Seshu (Company IT) said:
 My opinion (guess) on Polycom's Asterisk policy is - It is not that
 Polycom does not want their phones to be used with Asterisk. At the
 price these phones are sold, they will not be able provide support for
 all the features (AKA bugs or quirks) of Asterisk and make them
 transparent to Asterisk SIP stack and more notably - be user friendly
 for the Asterisk newbie user community. :)

That does not excuse them from not making the firmware or ducumentation
available. There is no reason for them to not allow downloads or provide
documentation - even requiring registration before download would be
OK.

Furthermore, one of the current issue people have (not being able to
disable call-waiting) is going to be a problem for ANY sip PBX software,
not just asterisk.

If they had ONE internal advocate that monitored this list for 2 hours a
day and provided feedback to internal engineering / product management,
and *occasionally* provided information to the list on major issues
people have, they could sell a LOT more of these phones and we would not
be having this discussion.

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RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

[mattf]

The 1.4.1 firmware and the 2.6.1 bootrom are also now on
http://www.freedomphones.net/polycom/files/

MATT---

-Original Message-
From: mattf 
Sent: Wednesday, January 26, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1


the 1.3.4 firmware is available on
http://www.freedomphones.net/polycom/files/


MATT---


-Original Message-
From: Chad Scott [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1


Noah, I could really use 1.3.4... however, a better question might be 
how are *you* getting 1.3.4?  I can't seem to get this from anyone, 
including my reseller.

On Jan 26, 2005, at 8:42 AM, Noah Miller wrote:

 Hi Chris -

 I am getting to my wits end with these phones (and so is my boss).  I 
 am
 getting an random echo on these phones and I have an issue opened with
 Polycom and its been in their research and development department for
 almost a month with no results.
 I'm amazed it got that far with Polycom.  My experience is that they 
 do not support these phones at all.

 I have noticed that I get a message RFC3389 support incomplete.  Turn
 off on client if possible in asterisk. I have researched this and 
 made
 the change in ipmid.cfg (see below), but I am still getting this RFC
 error.

 --- ipmid.cfg 
 RTP qos.ethernet.rtp.user_priority=5/
RTP qos.ip.rtp.min_delay=0
 qos.ip.rtp.max_throughput=0 qos.ip.rtp.max_reliability=0
 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=0/
RTP tcpIpApp.port.rtp.filterByIp=1
 tcpIpApp.port.rtp.filterByPort=0 tcpIpApp.port.rtp.forceSend=
 tcpIpApp.port.rtp.mediaPortRangeStart=/
 - end 

 I am just wondering if anyone can help me troubleshoot the echo and 
 RFC
 error so I don't have to pull the entire phone system out and purchase
 an entire new system.
 What version of * are you using?  I'm using 1.0.2 with Polycom 1.3.4 
 firmware on IP600's and I haven't seen any of these problems.  If 
 you'd like the 1.3.4 version of the firmware, just let me know off 
 list.

 Stupid Question:  Is the echo on all calls or just on calls to/from 
 the PSTN?  If just PSTN calls, do you have any echo cancellation 
 enabled?

 Thanks,
 Noah

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RE: [Asterisk-Users] Recording a meetme conference

[mattf]

We record meetme rooms by sending a manager Action to place a call from the
meetme room to an extension that is defined to start recording for a
predetermined amount of time, to end that recording we just send an Action
to Hangup that channel. Been working great for over a year now with over
180,000 recordings so far. 

Word of warning on other methods, recording one of the participant Zap or
SIP channels has given us mixed results in the past like one-side audio,
distortion or no audio at all in the recording, that's why we use the method
above.

This is also how the apps in the astGUIclient suite record in meetme rooms. 
http://astguiclient.sf.net/

MATT---

-Original Message-
From: Ben Merrills [mailto:[EMAIL PROTECTED]
Sent: Friday, January 21, 2005 5:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Recording a meetme conference


Is it possible to record a meetme conference? What channel would you
monitor, is there a main channel that all audio goes too?
 
If so, is it possible to use the ast_monitor (iirc) to record that channel?
 
Cheers,
 
Ben
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