RE: [Asterisk-Users] snom 360 incorrect US indications

[Michael J. Liberatore]

Snom's US tones have always been Terrible..  I have contacted them
several times, they recommeded I "use another countrys tones" u, I
don't think so, I don't think customers will like that too much.  Plus
the call waiting tone is terrible, its loud, cuts out the call, and the
outside party can hear the tone!  It should be very light and
unobtrusive like a normal phones call waiting. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, February 19, 2006 12:35 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] snom 360 incorrect US indications

Anyone noticed the snom 360 indications are incorrect for US zone?

menu->preferences->tone scheme->usa

indications.conf:
[general]
country=us

extensions.conf:
exten => ,1,Answer
exten => ,n,Playtones(dial)
exten => ,n,Wait(30)

exten => ,1,Busy

exten => ,1,Answer
exten => ,n,Playtones(busy)
exten => ,n,Wait(30)


hit speakerphone on the snom 360. listen to the dialtone.
now dial  and compare to asterisk's dialtone.

hit speakerphone on the snom 360. dial .
now compare to the busy signal you get from .

in each case, snom tone is incorrect and asterisk is correct.

-Dan
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Re: [Asterisk-Users] Application Faxing using SIP

[Lee Howard]

J Poz wrote:

Using an analog line is not an option for my service. My application 
runs on a ROOT SERVER of an ASP. So I can do anything I want to the 
server but I can't connect to or get external analog lines. So my 
options are doing faxing via the Internet (VOIP/SIP) or use a faxing 
service. But my experience with faxing services has not been too good 
as I've mentioned.



Traditional faxing (not T.38) pretty much requires a lossless audio 
channel.  Normally the best way to get this is with PSTN channels/lines 
through a Zap device.  That said, VoIP channels can be configured such 
that they are also lossless.  IAXmodem, for example, functions on the 
premise that an IAX2 channel passing over the loopback device will be 
lossless.  I have also seen lossless SIP and IAX channels running over a 
WAN, but they were very specificially configured, and I wouldn't expect 
most connections with traditional VoIP providers to be anything near the 
kind of losslessness that is required for this to work well.


For the most part, I suspect that those VoIP providers that promise fax 
support (over VoIP G.711) are doing so on a type of gamble... that ECM 
support of most fax machines will compensate, that they can control 
enough of the communication to mitigate the problem substantially, and 
that the remaining (say, 10%) error rate will not cause significant 
enough complaints from the users to cause it to be unprofitable.


So, be forewarned that faxing over VoIP channels is usually not going to 
work extremely well for you... not unless you can mitigate the problem 
by creating near-lossless connections between you and the endpoint with 
the PSTN connection.


Lee.
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[Asterisk-Users] snom 360 incorrect US indications

[asterisk]

Anyone noticed the snom 360 indications are incorrect for US zone?

menu->preferences->tone scheme->usa

indications.conf:
[general]
country=us

extensions.conf:
exten => ,1,Answer
exten => ,n,Playtones(dial)
exten => ,n,Wait(30)

exten => ,1,Busy

exten => ,1,Answer
exten => ,n,Playtones(busy)
exten => ,n,Wait(30)


hit speakerphone on the snom 360. listen to the dialtone.
now dial  and compare to asterisk's dialtone.

hit speakerphone on the snom 360. dial .
now compare to the busy signal you get from .

in each case, snom tone is incorrect and asterisk is correct.

-Dan
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[Asterisk-Users] co-location providers in Ottawa, Canada

[Richard OSS]

Anybody know if there are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server.     One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it.     Thanks.     richard___
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RE: [Asterisk-Users] Application Faxing using SIP

[J Poz]

MD,     Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Internet (VOIP/SIP) or use a faxing service. But my experience with faxing services has not been too good as I've mentioned.     Does your company provide an affordable, reliable, and somewhat real-time faxing service? Or can you recommend one? Otherwise, I have to experiment and try to see the results I can get with doing Internet faxing. Remember my experience so far with fax service providers - single faxes take 40+ minutes to eventually be sent (and the delays are within the faxing service and and not the receiving fax line - I've researched this).        Technical Support <[EMAIL PROTECTED]> wrote: 
 J:     We developed the mail2fax application (www.generationd.com) - so we should be able to give some insight.  I think you are confusing the time to "process" the incoming (by email) fax document, and the time to fax the document.  Fax over IP causes an enormous number of retries - thus delays.  I would suggest you do some experimenting with an analog line connected to your asterisk box.     MD  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Saturday, February 18, 2006 4:17 PMTo: Philip Edelbrock; Asterisk Users Mailing List - Non-Commercial DiscussionCc: J PozSubject: Re: [Asterisk-Users] Application Faxing using SIP   Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response a
 s
 possible (assuming the fax machines on the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see that but so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff, etc).     I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny.     And this definit
 ely is
 not anything related to spam fax, etc. - legit business but right now can't fully reveal.     So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration.     Thanks,  JPhilip Edelbrock <[EMAIL PROTECTED]> wrote:  On Feb 18, 2006, at 11:35 AM, J Poz wrote:> I have a specific business problem that I'm hoping someone has > ideas and/or has already worked out a solution.>> My application needs to be able to automatically create and issue > faxes to many different fax machines. The volume is going to be > very high. And it is only about sending faxes and not r
 eceiving
 them.>> My application is hosted by an ASP but the Linux (Fedora 2) server > is mine (dedicated). So the option of having PSTN lines to do faxes > is not an option since I don't own nor can put anything in the data > center. I found a SIP/VOIP provider that says they do faxing (and I > can connect to them using my own device (meaning asterisk or > something else if necessary)). Their requirement for faxing to work > on their end is to make sure i send them via their voip service > using G.711 codec.>> So I've done alot of research on faxing and asterisk and hylafax > but I' m still at a loss. F or starters, what is the architecture > that I need?>> my application --> QUESTION MARK??? > VOIP Provider ---> PSTN > ---> Fax Machine.>> So first question, what should QUESTION MARK be? Is it just > Asterisk or a combination of Asterisk
  and
 something like hylafax > (fax manager). And depending on that answer, what is the > configuration that has to be made on it. Even reference to material > that explains the configuration would be very helpful to me at this > time.>> Thanks in advance for the help,The missing link might be iaxmodem. It has two interfaces: IAX channel for asterisk, and a serial device (in /dev/) which emulates a faxmodem. Then, fax away using hylafax. I have tried faxing over SIP through a provider (broadvoice) to a coworker's fax on the pstn this way, and it worked. I haven't done any testing in volume, though.So you would have something like:Doc -> hylafax -> iaxmodem -> * -> voi

Re: [Asterisk-Users] Asterisk as MGCP User Agent

[David Lublink]

Hey,I  don't know enough about C to be to tackle this kind of a project. I am really busy and learning a new language would take at least a full week of learning, and I don't have that kind of time. In my searches I did find the site 
http://www.vovida.org/ and it seems like the code is free, maybe someone with C experience could make that work with asterisk?DavidOn 2/18/06, 
Stewart Nelson <[EMAIL PROTECTED]> wrote:
Hi David and all,> I have a voip provider that uses mgcp and I would like to connect that> provider to my asterisk.> Anyone succeed in doing this?I have a similar interest, for Free Télécom (France) DSL, which
includes an MGCP based VoIP service.  I have been too lazy to tacklethis myself, but would be willing to contribute some code.Another possibility is a standalone gateway program that acts as SIPserver and MGCP UA.
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RE: [Asterisk-Users] Application Faxing using SIP

[Technical Support]

J:
 
We developed the mail2fax application (www.generationd.com) - so we should be 
able to give some insight.  I think you are confusing the time to "process" 
the incoming (by email) fax document, and the time to fax the document.  
Fax over IP causes an enormous number of retries - thus delays.  I would 
suggest you do some experimenting with an analog line connected to your asterisk 
box.
 
MD


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of J 
PozSent: Saturday, February 18, 2006 4:17 PMTo: Philip 
Edelbrock; Asterisk Users Mailing List - Non-Commercial DiscussionCc: 
J PozSubject: Re: [Asterisk-Users] Application Faxing using 
SIP

 
Thanks for the information. I prefer to try to develop/configure something 
myself versus using an external provider. I currently use one (have tried 
another) and there are no guarantees on reliability, timelines, etc. I'm in need 
of as close to real-time response as possible (assuming the fax machines on 
the other end are operational). Something like within 5 minutes 85% of the time. 
My experience with 2 external mailtofax providers so far is terrible. Just 
yesterday, a simple 1 page fax took 42 minutes to finally send it. Other 
scenarios were 20, 26 minutes, etc. The provider tells me that they don't 
gaurantee anything but most of the time sending a single page fax within 5 
minutes 80% of the time. I wish I would see that but so far I'm seeing 
terrible response times from the few I've tried. I also need to know status of 
fax (in queue, failed, etc) in real-time so my application and react 
appropriately (send notification to support staff, etc).
 
I've found one that does have some sort of guarentee by the cost of through 
the roof and would kill my business model/plan (and the gaurantees are wishy 
washy). So I think I need to "control" my own destiny.
 
And this definitely is not anything related to spam fax, etc. - legit 
business but right now can't fully reveal.
 
So I'll have to research abit on IAXModem to use it. But your suggestion is 
a good one. Can you share what Asterisk configuration you use to both receive 
the iaxmodem feed and interface to the VOIP provider for such a 
configuration.
 
Thanks,
JPhilip Edelbrock <[EMAIL PROTECTED]> 
wrote:
On 
  Feb 18, 2006, at 11:35 AM, J Poz wrote:> I have a specific business 
  problem that I'm hoping someone has > ideas and/or has already worked 
  out a solution.>> My application needs to be able to 
  automatically create and issue > faxes to many different fax machines. 
  The volume is going to be > very high. And it is only about sending 
  faxes and not receiving them.>> My application is hosted by an 
  ASP but the Linux (Fedora 2) server > is mine (dedicated). So the 
  option of having PSTN lines to do faxes > is not an option since I 
  don't own nor can put anything in the data > center. I found a SIP/VOIP 
  provider that says they do faxing (and I > can connect to them using my 
  own device (meaning asterisk or > something else if necessary)). Their 
  requirement for faxing to work > on their end is to make sure i send 
  them via their voip service > using G.711 codec.>> So 
  I've done alot of research on faxing and asterisk and hylafax > but I' 
  m still at a loss. F or starters, what is the architecture > that I 
  need?>> my application --> QUESTION MARK??? > VOIP 
  Provider ---> PSTN > ---> Fax Machine.>> So first 
  question, what should QUESTION MARK be? Is it just > Asterisk or a 
  combination of Asterisk and something like hylafax > (fax manager). And 
  depending on that answer, what is the > configuration that has to be 
  made on it. Even reference to material > that explains the 
  configuration would be very helpful to me at this > 
  time.>> Thanks in advance for the help,The missing 
  link might be iaxmodem. It has two interfaces: IAX channel for asterisk, 
  and a serial device (in /dev/) which emulates a faxmodem. Then, fax away 
  using hylafax. I have tried faxing over SIP through a provider 
  (broadvoice) to a coworker's fax on the pstn this way, and it worked. I 
  haven't done any testing in volume, though.So you would have something 
  like:Doc -> hylafax -> iaxmodem -> * -> voip provider 
  -> pstn -> fax machinePhilPS- I suppose if you had 
  multiple SIP accounts with a provider, you could create multiple iaxmodems 
  and do things in parallel (assuming enough bandwidth and cpu).PPS- 
  I hope you're not doing fax-spamming with this set up! ;')


Yahoo! 
Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and 
used cars.
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Re: [Asterisk-Users] Asterisk as MGCP User Agent

[Stewart Nelson]

Hi David and all,

> I have a voip provider that uses mgcp and I would like to connect that
> provider to my asterisk.
> Anyone succeed in doing this?

I have a similar interest, for Free Télécom (France) DSL, which
includes an MGCP based VoIP service.  I have been too lazy to tackle
this myself, but would be willing to contribute some code.

Another possibility is a standalone gateway program that acts as SIP
server and MGCP UA.

--Stewart

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Re: [Asterisk-Users] Re: asterisk t.38 pass

[Stagg Shelton]

I'm very interested in implementing this patch.  Currently I am using 
code that came from ionidea that I managed to get compiled into asterisk 
1.0.9.  I've been digging through all the comments in this bugtracker 
issue for the past week or so, and trying to find the time to give this 
new code a try.  If it works well in my site, I'll put it on a few of 
our customers sites who complain about faxing the most, and let you know 
how it goes.  We are a wireless ISP delivering voice over our wireless 
highspeed data network. I can't imagine a worse condition for faxing 
than long distance point to point wireless.  Voice is great though :)


Stagg Shelton
www.oneringnetworks.com

Adolfo R. Brandes wrote:


turby wrote:


yes, with last patch works well. thanks.



Glad to be of service!

Adolfo

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RE: [Asterisk-Users] Bridged line appearance

[mustardman29]

 
You make some good points Clint,

I honestly don't think that trying to force feed this to the customer as it
is is the way to go.  Key systems have been used for many many years and the
market has decided that they are what people want in the lower end.  I have
sat in small offices and witnessed the elegant simplicity of a key system.
It's all 1 single button press to do ANYTHING.  The button label and the
light beside it tells you everything you need to know.  It works!  No multi
button sequences or *xx key presses to know.  People on this forum might not
have a problem with more complexity in exchange for more flexibility etc.
but I don't think the people on this forum are anything like an average
user.

Perhaps Asterisk will never be appropriate for they low end Key market or
the Key/PBX hybrid market.  I don't know.  There are IP phones around with
plenty of buttons to do the job.  The Aastra9133i has something like 9
programmable buttons in addition to 3 incoming line buttons which is plenty
for most small businesses.  Their latest firmware now fully supports BLF and
apparently SLA.

Did I come across as complaining?  Just trying to make a case for what I see
as a highly desireable feature.  I do take exception to anyone trying paint
a picture of me being an ungrateful open source software user looking for a
free ride.  Digium has made PLENTY of money off of me.  If I could pay
another $300 to get the features I want I would.  It's not all about saving
a few bucks!  If I wanted to do that I would buy Bizfon's.

> -Original Message-
> From: Clint Sharp [mailto:[EMAIL PROTECTED]
> Sent: Saturday, February 18, 2006 5:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bridged line appearance
> 
> I'm having a very hard time justifying trying to sell this to the SOHO 
> market on price or parity with key systems.  I've installed key 
> systems and large scale PBXs, and while working around the SLA problem 
> isn't that hard, the price point for a key system is very hard to 
> compete with.  I've never understood why people would want to use an 
> SLA system, honestly, as it's a really poor model.  I hate sitting in 
> offices with constant paging "Call for blah, line 1".  The PBX model 
> to me is much more preferable, and working around it is simply a 
> training problem.
> 
> The problem with asterisk isn't the lack of SLA, it's the price point.  
> It's going to be very hard for IP phone vendors to compete on price at 
> this point, and so far the quality issues in low-priced hardware to me 
> means I can't really sell this to anyone who's not willing to pay 
> $200-$300 a phone (retail).
> 
> Not that it's impossible, it's a different sales strategy.  
> Perhaps people who are wanting to sell this to the SOHO market should 
> attempt to change the game, as PBX like functionality doesn't exist in 
> the SOHO market because it hasn't been affordable previously.  
> Asterisk systems are pretty cheap in terms of the features they offer, 
> such that the sales pitch really depends on cost for features and 
> maintenance and infrastructure savings rather than overall cost.
> 
> Admittedly though, the voicemail system's navigation issues are a big 
> problem.
> 
> Clint
> 
> 
> On 2/18/06, John Novack <[EMAIL PROTECTED]> wrote:
> 
> 
>   Many very low cost hybrid key/pbx systems for the small business
SOHO
>   market have 12 or more programmable buttons, so regardless of what
is
>   done with Asterisk, until the IP phone manufacturers take off their
>   blinders and manufacture competing equipment, this market will be
out 
> of
>   reach. These same systems now have voice mail systems with 
> capabilities
>   and features that make Comedian Mail the correct name.  
> Asterisk isn't 
>   alone regarding these shortfalls, of course. IP phone system 
> designers
>   have failed to understand the small business market for several 
> years.
>   
>   Defensive responses with lengthy explanations  why it can't be done,

> or
>   why it hasn't been done and will be difficult miss the point. Either
>   Asterisk needs to change to move into this market, or another
product 
> will
>   
>   JMO
>   
>   John Novack
>   
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Re: [Asterisk-Users] TE411P Really Bad Echo

[Stagg Shelton]

This is with a TE411P (Digium Quad Span PRI with Voice Processing
Module). When I say pulled the zaptel trunk source, I mean I issued the
following command which pulled the latest available source code at that
time:

svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel

I followed up to a message previously with the exact trunk version that
I downloaded:  SVN-trunk-r941

Stagg Shelton
www.oneringnetworks.com


Eric Bishop wrote:
Is this with the TE411P? Also what do you mean by "pulled
the zaptel trunk source"?
  
  On 2/17/06, Stagg Shelton <
[EMAIL PROTECTED]> wrote:
  This
is my last update to my issue.  Finally my echo problem is

resolved.  On Monday morning 2/13/06 I pulled the the zaptel trunk
source.  That night after my customers core business hours we built the
new zaptel drivers, rebuilt libpri, asterisk, asterisk-addons.  My echo
disappeared almost entirely  we made a few tweaks with the tx and rx
gain settings.  My echo problem disappeared completely with the
additional tweaks to txgain.  Occasionally at the very beginning of a
local call echo will exist for a second or so, but then it goes away.


In two operating days there has only been one notice from a user about
experiencing an echo.  All the users were informed that they should
notify us of any echo experiences.

Here are my final configurations

zaptel trunk pulled 2/13/06 approx 10:00am est.
Asterisk 1.2.4
LibPri 1.2.2
Asterisk-Addons 1.2.1
Asterisk-Sounds 1.2.1


/etc/zaptel.conf
=
span=1,1,0,esf,b8zs
bchan=1-23

dchan=24
#bchan=25-47
#dchan=48
#bchan=49-71
#dchan=72
#bchan=73-95
#dchan=96
fxoks=97-100
loadzone = us
defaultzone=us

/etc/asterisk/zapata.conf

context=from-pstn

switchtype=national
pridialplan=national
signalling=pri_cpe
resetinterval=never
faxdetect=incoming
usecallerid=yes
echocancel=yes
echotraining=800
rxgain=4.5
txgain=-13.5
group=0
channel=>1-23



Thank You for all of your pointers and support in this issue.

Stagg Shelton
www.oneringnetworks.com

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Re: [Asterisk-Users] Asterisk as MGCP User Agent

[David Lublink]

thanks,I had seen that thread and I was hoping something concrete had come up since then...oh well,DavidOn 2/18/06, Leo Ann Boon
 <[EMAIL PROTECTED]> wrote:David Lublink wrote:
> Hey,>> I have a voip provider that uses mgcp and I would like to connect that> provider to my asterisk.>> Anyone succeed in doing this?Asterisk can only function as an MGCP call agent not gateway. There was
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[Asterisk-Users] COMMPARTNERS Resellers

[Lists]

Is anyone on the list a Commpartner reseller?

 

 






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[Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning

[Adam Robins]

 
After many days of playing with the new jitterbuffer and trunking options for 
IAX2, I have finally received almost acceptable quality.  I am receiving 5-8 
complaints a day of calls "breaking up" from both the customer and agent sides. 
 What I have discovered is that in most of these cases, the new jitterbuffer 
performed a resync during the call.  Currently, I have the resyncthreshold, and 
all other jb parameters at their default levels  The traffic is running over a 
fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC).  
Idle ping times run about 85ms.
 
Below are the resync messages for this past Friday.  Knowing that I have a slow 
connection, should I set the resync at a much higher level?  I appreciate any 
assistance you may provide.
 
Thanks,
Adam
 
Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -34, 
this delay 1651, threshold 1488, new offset -1651
Feb 17 09:07:42 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, 
this delay -1684, threshold 1000, new offset 33
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, 
this delay 1835, threshold 1126, new offset -1835
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 32, 
this delay 1673, threshold 1062, new offset -1673
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -1663, threshold 1300, new offset -172
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -1635, threshold 1300, new offset -38
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -22, 
this delay 2335, threshold 1054, new offset -2373
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 11, 
this delay 2363, threshold 1082, new offset -2535
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -71, 
this delay 2249, threshold 1054, new offset -2249
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -180, 
this delay -2359, threshold 1360, new offset -14
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -2354, threshold 1300, new offset -181
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, 
this delay -2297, threshold 1240, new offset 48
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 109, 
this delay 1556, threshold 1136, new offset -1556
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -30, 
this delay -1439, threshold 1000, new offset -117
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, 
this delay 1608, threshold 1048, new offset -1725
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -29, 
this delay -1616, threshold 1058, new offset -109
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 21, 
this delay 1751, threshold 1620, new offset -1751
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, 
this delay 1724, threshold 1686, new offset -1724
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, 
this delay -1716, threshold 1000, new offset -8
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -119, 
this delay -1757, threshold 1000, new offset 6
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 75, 
this delay 1421, threshold 1326, new offset -1421
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 274, 
this delay 1595, threshold 1282, new offset -1595
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1311, 
this delay 820, threshold 1824, new offset -2415
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, 
this delay 761, threshold 1752, new offset -2182
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299, 
this delay -2127, threshold 1598, new offset -288
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -270, 
this delay -2106, threshold 1540, new offset -76
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 98, 
this delay 1878, threshold 1206, new offset -1878
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 44, 
this delay 1799, threshold 1150, new offset -1799
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 28, 
this delay 1781, threshold 1146, new offset -1781
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -1753, threshold 1000, new offset -46
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, 
this delay -1765, threshold 1000, new offset -16
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -149, 
this delay -1747, threshold 1298, new offset -131
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -44, 
this delay 1136, threshold 1064, new o

Re: [Asterisk-Users] Bridged line appearance

[Clint Sharp]

I'm having a very hard time justifying trying to sell this to the SOHO
market on price or parity with key systems.  I've installed key
systems and large scale PBXs, and while working around the SLA problem
isn't that hard, the price point for a key system is very hard to
compete with.  I've never understood why people would want to use
an SLA system, honestly, as it's a really poor model.  I hate
sitting in offices with constant paging "Call for blah, line 1". 
The PBX model to me is much more preferable, and working around it is
simply a training problem.

The problem with asterisk isn't the lack of SLA, it's the price
point.  It's going to be very hard for IP phone vendors to compete
on price at this point, and so far the quality issues in low-priced
hardware to me means I can't really sell this to anyone who's not
willing to pay $200-$300 a phone (retail).

Not that it's impossible, it's a different sales strategy. 
Perhaps people who are wanting to sell this to the SOHO market should
attempt to change the game, as PBX like functionality doesn't exist in
the SOHO market because it hasn't been affordable previously. 
Asterisk systems are pretty cheap in terms of the features they offer,
such that the sales pitch really depends on cost for features and
maintenance and infrastructure savings rather than overall cost.

Admittedly though, the voicemail system's navigation issues are a big problem.

ClintOn 2/18/06, John Novack <[EMAIL PROTECTED]> wrote:
Many very low cost hybrid key/pbx systems for the small business SOHOmarket have 12 or more programmable buttons, so regardless of what isdone with Asterisk, until the IP phone manufacturers take off their
blinders and manufacture competing equipment, this market will be out ofreach. These same systems now have voice mail systems with capabilitiesand features that make Comedian Mail the correct name.  Asterisk isn't
alone regarding these shortfalls, of course. IP phone system designershave failed to understand the small business market for several years.Defensive responses with lengthy explanations  why it can't be done, or
why it hasn't been done and will be difficult miss the point. EitherAsterisk needs to change to move into this market, or another product willJMOJohn Novack___
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[Asterisk-Users] PSTN to SIP MoH always choppy, SIP to SIP good

[Zach A]

Hi all,

I've tried everything over past month but no success. When somebody
calls in from a PSTN line or cell phone to my asterisk box, which is a
connected to a SIP provider, MoH doesn't work good, it is very choppy.
I've tried native format, silence suppression, ulaw and gsm music files,
changing RTP ports but nothing helped. It plays ok only for incoming SIP
calls from the same provider. Any guess why it doesn't work for other
incoming calls and is always choppy?

Thanks,

Zach A.

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Re: [Asterisk-Users] Asterisk as MGCP User Agent

[Leo Ann Boon]

David Lublink wrote:


Hey,

I have a voip provider that uses mgcp and I would like to connect that 
provider to my asterisk.


Anyone succeed in doing this?


Asterisk can only function as an MGCP call agent not gateway. There was 
a thread about this, a couple of months back. I don't remember anything 
concrete coming out of that thread.


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Re: [Asterisk-Users] sipdiscount

[Roshan Sembacuttiaratchy]

I use the following in my Asterisk 1.2.4:

[out-sipdiscount]
type=peer
secret=mypass
username=myuser
fromuser=myuser
host=sip1.sipdiscount.com
call-limit=1
disallow=all
allow=ulaw
allow=g726
allow=g729
allow=g723.1

Using this configuration, I am able to place calls using a Dial
statement such as:

Dial(SIP/[EMAIL PROTECTED])

HTH,

Roshan
http://roshan.info

On Fri, Feb 17, 2006 at 12:31:17PM +0100, Alejandro Vargas scribbled:
> 2006/2/17, Peter Bowyer <[EMAIL PROTECTED]>:
> > A pretty simple setup works for me:
> 
> The problem may be the username/password. But the page says this:
> 
> SIP Discount offers the possibility to test our service right away,
> for free! No need to sign up: just enter the account details below in
> your favorite softphone or ATA and start calling! You can call all
> destinations marked with * in our rate list . (Trial calls are limited
> to a maximum duration of 1 minute). To enjoy unlimited calls, simply
> sign up for SIP Discount.
> 
> User Name: test
> Password: test
> Domain/Realm: sipdiscount.com
> SIP Proxy/registrar:  sip1.sipdiscount.com
> SIP Outbound Proxy (optional):sip1.sipdiscount.com
> STUN server (optional):   stun.sipdiscount.com
> 
> The only problem I say is asterisk is not sending stun.sipdiscount.com
> or sipdiscount.com as domain. It is sending sip1.sipdiscount.com.
> 
> --
> Alejandro Vargas
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Re: [Asterisk-Users] Bridged line appearance

[John Novack]

Michael J. Liberatore wrote:


Man, I am all for shared line appearances.  I have asterisk systems in several 
small businesses and they all cry for it.  But there are ways around it as 
well, after a week all the bussinesses have gotten used to asterisk w/o bla.  
Plus, past 4 lines, its hard to implement cause lots of phones only have 4 
lines.  Trust me though arguing on this list wont get you the feature quicker, 
I have read tons of e-mails on here and have seen a pattern :)

Many very low cost hybrid key/pbx systems for the small business SOHO 
market have 12 or more programmable buttons, so regardless of what is 
done with Asterisk, until the IP phone manufacturers take off their 
blinders and manufacture competing equipment, this market will be out of 
reach. These same systems now have voice mail systems with capabilities 
and features that make Comedian Mail the correct name.  Asterisk isn't 
alone regarding these shortfalls, of course. IP phone system designers 
have failed to understand the small business market for several years. 

Defensive responses with lengthy explanations  why it can't be done, or 
why it hasn't been done and will be difficult miss the point. Either 
Asterisk needs to change to move into this market, or another product will


JMO

John Novack

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RE: [Asterisk-Users] sipura 3000 and other probs

[Jean-François Rousseau]

Hi, I had some problems similar to you, but they disapeared when I installed
the 2.x firmware instead of the 3.x

You might want to try.

Good luck 


___
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www.sys-tech.net
[EMAIL PROTECTED]
Tél. 24h (418) 520-0739Télec. (418) 520-4554
1-877-969-tech
Ouverture Technologique

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : 9 février 2006 11:38
À : asterisk-users@lists.digium.com
Objet : RE: [Asterisk-Users] sipura 3000 and other probs

With pen in hand, Technical Support succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said ...
> There's a vox forum that focuses on Sipuras - post your query there 
> for good tech help.  We've deployed a number of Sipura's and haven't 
> experienced that problem (yet).  Have you started with the basics:
> firmware version, analog cabling, etc.
>
> MD
>

Oh yeah... ex Satcom/ISDN tech here and so I checked all the basics, swapped
caples, firmware is the latest, etc. I'll check on the voxilla forum, that
completely slipped my mind.

I was hoping that maybe someone on the list had seen something like this,
particularly given the error from the log I described below:

chan_sip.c: That's odd...  Got a response on a call we dont know about.
Cseq 102 Cmd SIP/2.0

Does anyone have the time to let me know exactly what this error points to?
When I say exactly, of course I don't mean exactly in my configuration which
none of you have seen, I mean, what is the software seeing that it isn't
prepared to handle. As I also mentioned, I looked at the code, but I'm an
amateur programmer at best, with little experience, not to mention I'm still
learning about the SIP protocol, so I'm just not sure what this is telling
me.

I strongly suspect that this is probably tied in with my situation, but
according to all the docs/forums/setups I've researched, my setup looks OK.

Regards,

John C.

> -Original Message-
> From: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] sipura 3000 and other probs
>
>...
>...
> For example, I'll be talking to an incoming caller and an echo starts  
>quietly in the background. Within a minute or so, I'll lose the  
>connection,  with my voice 100% reflecting back at me, and the caller 
>says the same  occurs at their end, his/her voice 100% reflecting back 
>to him/her.
>
> I usually have to reset the device completely at that point. Other 
> times, I'll get a call and talk for 1/2 hour with no problems whatsoever.
>
> I have it set up with [EMAIL PROTECTED] V 2.2 according to the setup at
>
>   http://mundy.org/blog/index.php?p=65
>
> Also I get the following strange error:
>
> chan_sip.c: That's odd...  Got a response on a call we dont know about.
> Cseq 102 Cmd SIP/2.0
>
> I can't seem to pin it down. I've checked the source (chan_sip.c) and  
>because I'm not well aquainted with the protocol, I'm about as clueless  
>as could be as to what its telling me.
>...


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RE: [Asterisk-Users] Bridged line appearance

[Michael J. Liberatore]

Man, I am all for shared line appearances.  I have asterisk systems in
several small businesses and they all cry for it.  But there are ways
around it as well, after a week all the bussinesses have gotten used to
asterisk w/o bla.  Plus, past 4 lines, its hard to implement cause lots
of phones only have 4 lines.  Trust me though arguing on this list wont
get you the feature quicker, I have read tons of e-mails on here and
have seen a pattern :)

Now, I don't code C, but would like the feature for some customers.  If
you would be interested in forming a bounty with me, I would be possibly
willing to donate some money to the bounty with you. But if you just
want to complain then good luck getting this implemented quicker.

Mike


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
mustardman29
Sent: Saturday, February 18, 2006 12:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridged line appearance

 
>  1) Yes. There are "plans for it".
GREAT!  What is the current status and expected timeline?
> 
>  2) No. It won't be easy as Asterisk is a multi-protocol PBX and 
> usually when we consider introducing a feature like this, its intent 
> is for it to function across all of the protocols that Asterisk 
> supports, VoIP or not. Everyone else you've mentioned needs only worry

> about their own device supporting a standard or their own system only 
> supporting devices that they manufacture to support the feature. That 
> makes things somewhat easier for implementation and Asterisk has no 
> such luxury given it's completely open nature which most of us see as 
> an advantage.
Thanks for explaining the details of why it will be difficult
> 
>  3) The other solutions you've mentioned above all have 
> salaried engineering staffs whose job it is to implement 
> features as decided by product management folks also employed 
> by that company who are driven by the comments and feedback 
> of users such as yourself who fork over large sums of money 
> compared to what you pay for your Asterisk to have such 
> solutions. Had you sent such an email to one of these 
> companies at the time you did on a Friday night in the 
> states, my bet is on the fact that it wouldn't have even 
> solicited an initial response from a product management 
> resource until Monday morning.
Ummm.ok.  Asterisk=open source community.  That just goes without
saying.  Other than that I don't know what your point is.  So there are
no
salaried software engineers at Digium working on Asterisk?
> 
>  4) The SPA-9000 is devoid of features like, Voicemail, which 
> Asterisk already has. If a system without BLA is a 
> "non-starter" for you and these small business you have 
> cited, why not consider a combined solution where Asterisk 
> provides features (call queues/ACD, voicemail,
> etc) that the SPA-9000 does not have and then you use the 
> SPA-9000 for what it is good for (an IP key system - which is 
> not what Asterisk is)? Asterisk can be whatever and play 
> whatever part you want it to play in your solution. It 
> doesn't have to be the entire solution.
> Because of its open nature, it usually integrates and 
> interoperates well with many existing products/solutions. The 
> SPA-9000 is no exception.
Thanks for pointing out the differences.  Yes, I have thought about
creating
a Frankenstein system which takes advantage of the strengths of both the
SPA-9000 and Asterisk.  Perhaps using Asterisk as a POT's gateway and
voicemail server.  The cost starts to creep up though.  This is a
concept I
have been mulling over for awhile now.  It remains to be seen what the
best
direction is.  When in doubt the best strategy is KISS.  The simplest,
cheapest, and presumably most robust solution is to have everything in
one
box.
> 
>  5) There are thousands of small businesses already, my own 
> being one of them, that would disagree that Asterisk is a 
> "non starter" for them. Asterisk is what you make of it, and 
> for us, it's a criticial communications tool for our business.
At the end of the day it is what the user thinks, not the Linux people.
For
you, me and most others on this board I think we can all agree that
Asterisk
works just fine for us.  For some companies used to PBX like
functionality
it will probably work just fine as well which I have already pointed
out.
For many many other companies used to key system like functionality it
is a
non-starter mostly because of the lack of BLA IMHO. If you don't believe
me
that it is a VERY important feature then ask yourself why a LOT of IP
phones
and VoIP systems support it or are starting to support it.  If Asterisk
wants to be a main stream phone system then I feel it should support it.
Has nothing to do with open source vs proprietary.  Just giving my
opinion
based on user feedback. 
> 
>  These things being said, what was your original intent for 
> writing such an email? Is there something you'd like to

Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

[Tzafrir Cohen]

On Fri, Feb 17, 2006 at 09:16:41AM +0100, Giorgio Incantalupo wrote:
> Hi Chuck,
> my solution may be considered a bit strange but I chose it after trying 
> asterisk code without success, trying to use Tzafrir patch but I had to 
> change asterisk user umask too
> The right solution could be something like a "voicemail_dir_permissions" 
> parameter in voicemail.conf so anyone could change permissions without 
> modifying asterisk code. The externnotify parameter solution I used was 
> the faster and less "invasive". If you want to make a script to install 
> asterisk, it is better to copy voicemail.conf and a script file than 
> patching.

What should that parameter do? Why not just fix the problem (as in our
packages, and as I tried to push for in the above bug report).

You seem to want an explicit chmod on file creation (to override the
umask). That will still create some races where voicemail files are not
yet readable.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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RE: [Asterisk-Users] Grandstream GXP-2000

[Michael J. Liberatore]

Well the gxp-2000 has BLF, the polycom 501 does not correct?  I had an
astra 480i and it was prety bad, but I was going to test the 9133i for
an inexpensive phone to compete with the gxp2000.  The gxp2000 is not
bad though, the new firmware helps a lot, but once they work out the
echo bugs fully and the various minor stuff it will be a good sub $100
phone.  I am yet to find a phone under $300 that's perfect... The snom
360 is nice, but I have lots of problems with those too.  I havent tried
any polycom's though and starting to think they might be some of th
ebest... 

Mike
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, February 18, 2006 7:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream GXP-2000

On Fri, 17 Feb 2006, mustardman29 wrote:
> The GXP2000 firmware is not bad for features and ease of use but still

> buggy.  The hardware is junk to be quite honest and I don't think 
> firmware will ever fix that.  The Aastra 9133i hardware is 10x better.

> The 9133i firmware is still a work in progress though but they are 
> coming out with new firmware every few months and each iteration 
> improves the operation.  Long term I believe any of the Aastra phones
are a MUCH better.

why bother with an aastra 9133i when you can have a polycom 501. better
phone, same price.

-Dan
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RE: [Asterisk-Users] An array of extensions in my lab

[Rob Thomas]

We buy them? Is this a trick question?

--Rob


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ed Greenberg
> Sent: Sunday, 19 February 2006 7:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] An array of extensions in my lab
> 
> When working on prototypeing asterisk installations, I sometimes need
an
> array of extensions in my lab.
> 
> How do others handle this?
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[Asterisk-Users] Asterisk as MGCP User Agent

[David Lublink]

Hey,I have a voip provider that uses mgcp and I would like to connect that provider to my asterisk.Anyone succeed in doing this?Thanks,David
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[Asterisk-Users] An array of extensions in my lab

[Ed Greenberg]

When working on prototypeing asterisk installations, I sometimes need an 
array of extensions in my lab.


How do others handle this?
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Re: [Asterisk-Users] Grandstream GXP-2000

[Clint Sharp]

If you can live without a speakerphone, the Polycom 301 is an excellent
phone and is only $30US more expensive than the GXP-2000s.  I tend
to trust the Polycom brand, and they haven't really steered me wrong
yet in the IP phone hardware.  I'm interested though in any
reports of success with the Aastra phones, as I'm certainly looking for
something with better quality than Grandstream with lower-end
pricing.  Anybody have any reports on using the Uniden phones?


ClintOn 2/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

why bother with an aastra 9133i when you can have a polycom 501. betterphone, same price.
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Re: [Asterisk-Users] Fwd: Which ATA device do you recommend?

[Ed Greenberg]

For single and two-port applications, I've had very good luck with Sipura 
2000s. Now available as Linksys PAP2-NA.




--On Wednesday, February 15, 2006 3:08 PM + Marco Mouta 
<[EMAIL PROTECTED]> wrote:



-- Forwarded message --
From: Marco Mouta <[EMAIL PROTECTED]>
Date: Feb 15, 2006 1:58 PM
Subject: Which ATA device do you recommend?
To: [EMAIL PROTECTED]


Hello,

I'm developing a Voip Solution for a client, which ATA SIP do you
recommend? there are some ATA devices fully tested with Asterisk?

I hope that Asterisk experient users could give me their advice based
on their experiencies.

Thanks to all,
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Re: [Asterisk-Users] Aasterisk large-scale deployment w/analog phones

[Ed Greenberg]

Consider also a set of AudioCodes MP124s. These are 24 port ATAs. Work well 
for us.


Four of them would give 96 ports.

No zap hardware needed to connect the channel banks.



--On Wednesday, February 15, 2006 2:14 PM +0200 maka <[EMAIL PROTECTED]> 
wrote:



hello,

I am planning a fairly large hotel VoIP system, using analog phones. It
will consist of about 100 analog phones, that must have access to a VoIP
server. I am considering an option to use a couple of asterisk boxes,
bundled with a total of four TDM2460E cards, and one TDM2451E card.

Has anyone on this list done something similar? It would be great to hear
some comments regarding a smilar setuyp/planning - Do you think is it
better to distribute resources among multiple (more than two),
lower-port-density  asterisk servers? Or is it better to use a
channelbank for that purpose?

Cheers





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Re: [Asterisk-Users] Application Faxing using SIP

[J Poz]

   Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response as possible (assuming the fax machines on the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see that but so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff,
 etc).     I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny.     And this definitely is not anything related to spam fax, etc. - legit business but right now can't fully reveal.     So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration.     Thanks,  JPhilip Edelbrock <[EMAIL PROTECTED]> wrote:  On Feb 18, 2006, at 11:35 AM, J Poz wrote:> I have a specific business problem that 
 I'm
 hoping someone has > ideas and/or has already worked out a solution.>> My application needs to be able to automatically create and issue > faxes to many different fax machines. The volume is going to be > very high. And it is only about sending faxes and not receiving them.>> My application is hosted by an ASP but the Linux (Fedora 2) server > is mine (dedicated). So the option of having PSTN lines to do faxes > is not an option since I don't own nor can put anything in the data > center. I found a SIP/VOIP provider that says they do faxing (and I > can connect to them using my own device (meaning asterisk or > something else if necessary)). Their requirement for faxing to work > on their end is to make sure i send them via their voip service > using G.711 codec.>> So I've done alot of research on faxing and asterisk and hylafax > but I' m still at a loss. F
 or
 starters, what is the architecture > that I need?>> my application --> QUESTION MARK??? > VOIP Provider ---> PSTN > ---> Fax Machine.>> So first question, what should QUESTION MARK be? Is it just > Asterisk or a combination of Asterisk and something like hylafax > (fax manager). And depending on that answer, what is the > configuration that has to be made on it. Even reference to material > that explains the configuration would be very helpful to me at this > time.>> Thanks in advance for the help,The missing link might be iaxmodem. It has two interfaces: IAX channel for asterisk, and a serial device (in /dev/) which emulates a faxmodem. Then, fax away using hylafax. I have tried faxing over SIP through a provider (broadvoice) to a coworker's fax on the pstn this way, and it worked. I haven't done any testing in volume, though.So you 
 would
 have something like:Doc -> hylafax -> iaxmodem -> * -> voip provider -> pstn -> fax machinePhilPS- I suppose if you had multiple SIP accounts with a provider, you could create multiple iaxmodems and do things in parallel (assuming enough bandwidth and cpu).PPS- I hope you're not doing fax-spamming with this set up! ;')
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RE: [Asterisk-Users] Bridged line appearance

[David Ankers]

Simply amazing.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mustardman29
Sent: Sunday, 19 February 2006 4:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridged line appearance

 
>  1) Yes. There are "plans for it".
GREAT!  What is the current status and expected timeline?
> 
>  2) No. It won't be easy as Asterisk is a multi-protocol PBX 
> and usually when we consider introducing a feature like this, 
> its intent is for it to function across all of the protocols 
> that Asterisk supports, VoIP or not. Everyone else you've 
> mentioned needs only worry about their own device supporting 
> a standard or their own system only supporting devices that 
> they manufacture to support the feature. That makes things 
> somewhat easier for implementation and Asterisk has no such 
> luxury given it's completely open nature which most of us see 
> as an advantage.
Thanks for explaining the details of why it will be difficult
> 
>  3) The other solutions you've mentioned above all have 
> salaried engineering staffs whose job it is to implement 
> features as decided by product management folks also employed 
> by that company who are driven by the comments and feedback 
> of users such as yourself who fork over large sums of money 
> compared to what you pay for your Asterisk to have such 
> solutions. Had you sent such an email to one of these 
> companies at the time you did on a Friday night in the 
> states, my bet is on the fact that it wouldn't have even 
> solicited an initial response from a product management 
> resource until Monday morning.
Ummm.ok.  Asterisk=open source community.  That just goes without
saying.  Other than that I don't know what your point is.  So there are no
salaried software engineers at Digium working on Asterisk?
> 
>  4) The SPA-9000 is devoid of features like, Voicemail, which 
> Asterisk already has. If a system without BLA is a 
> "non-starter" for you and these small business you have 
> cited, why not consider a combined solution where Asterisk 
> provides features (call queues/ACD, voicemail,
> etc) that the SPA-9000 does not have and then you use the 
> SPA-9000 for what it is good for (an IP key system - which is 
> not what Asterisk is)? Asterisk can be whatever and play 
> whatever part you want it to play in your solution. It 
> doesn't have to be the entire solution.
> Because of its open nature, it usually integrates and 
> interoperates well with many existing products/solutions. The 
> SPA-9000 is no exception.
Thanks for pointing out the differences.  Yes, I have thought about creating
a Frankenstein system which takes advantage of the strengths of both the
SPA-9000 and Asterisk.  Perhaps using Asterisk as a POT's gateway and
voicemail server.  The cost starts to creep up though.  This is a concept I
have been mulling over for awhile now.  It remains to be seen what the best
direction is.  When in doubt the best strategy is KISS.  The simplest,
cheapest, and presumably most robust solution is to have everything in one
box.
> 
>  5) There are thousands of small businesses already, my own 
> being one of them, that would disagree that Asterisk is a 
> "non starter" for them. Asterisk is what you make of it, and 
> for us, it's a criticial communications tool for our business.
At the end of the day it is what the user thinks, not the Linux people.  For
you, me and most others on this board I think we can all agree that Asterisk
works just fine for us.  For some companies used to PBX like functionality
it will probably work just fine as well which I have already pointed out.
For many many other companies used to key system like functionality it is a
non-starter mostly because of the lack of BLA IMHO. If you don't believe me
that it is a VERY important feature then ask yourself why a LOT of IP phones
and VoIP systems support it or are starting to support it.  If Asterisk
wants to be a main stream phone system then I feel it should support it.
Has nothing to do with open source vs proprietary.  Just giving my opinion
based on user feedback. 
> 
>  These things being said, what was your original intent for 
> writing such an email? Is there something you'd like to 
> contribute to help get this feature implemented? You don't 
> need to be a developer to contribute. There's testing, 
> documentation, bounties to be set for features one "must 
> have", and all sorts of other areas that could use the 
> assistance of folks like yourself that aren't software developers.
Sure, what is the development schedule?  I get your point.  No need to beat
me over the head with it.  I read these sorts of comments about how it's
"your fault for not being a software coder" and "if you don't like it too
bad, it's your fault for not getting more involved" and frankly I am sick of
it.  We all know this is open source, we mostly all know the advantages and
disadvantages of it and we would not

Re: [Asterisk-Users] Calling number rewriting

[Jean-Christophe Heger]

You may change the CallerID with SetCallerID function, and the
presentation with CallingPres, before dialing.

In theory, you may place the CallerID you want, but your phone operator
could refuse it if it doesn't belong to you (will show a default
number). You also might have to use CallingPres to tune the presentation
flags. For Swisscom with a ZapHFC BRI card, working values are 0: show,
and 32: hide.

Jean-Christophe

ADEGOKE ARUNA a écrit :
> Hi all,
>
> I will be glad if I can get response to my need.
>
> What I am trying to do is;
>
> I have a set up where my asterisk box is directly connected to a digitalk IN
> platform.
>
> However, between the asterisk and digitalk is a104d sangoma card having e1
> with pri. The link between digitalk and my big alcatel switch is e1 with ss7
> signalling and this finally lead to my mobile operator
>
> What I am planning to do is to rewrite the caller id so that the calling
> number presented to the mobile operator is going to be the number set on the
> pri channels
>
> I will be glad, if anyone can just lead me on this
>
> goksie
>
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Re: [Asterisk-Users] Application Faxing using SIP

[Peer Oliver Schmidt]

J Poz schrieb:
I have a specific business problem that I'm hoping someone has ideas 
and/or has already worked out a solution.
 
My application needs to be able to automatically create and issue faxes 
to many different fax machines. The volume is going to be very high. And 
it is only about sending faxes and not receiving them.

[..]

my application --> QUESTION MARK???  > VOIP Provider ---> PSTN ---> 
Fax Machine.


don't use VoIP, but an external mail2fax provider. I am sure everythng 
else is asking for trouble. I am also positiv, a couple of businesses on 
this list would love to help you out with this.


rgds
posde
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Re: [Asterisk-Users] Application Faxing using SIP

[Philip Edelbrock]

On Feb 18, 2006, at 11:35 AM, J Poz wrote:

I have a specific business problem that I'm hoping someone has  
ideas and/or has already worked out a solution.


My application needs to be able to automatically create and issue  
faxes to many different fax machines. The volume is going to be  
very high. And it is only about sending faxes and not receiving them.


My application is hosted by an ASP but the Linux (Fedora 2) server  
is mine (dedicated). So the option of having PSTN lines to do faxes  
is not an option since I don't own nor can put anything in the data  
center. I found a SIP/VOIP provider that says they do faxing (and I  
can connect to them using my own device (meaning asterisk or  
something else if necessary)). Their requirement for faxing to work  
on their end is to make sure i send them via their voip service  
using G.711 codec.


So I've done alot of research on faxing and asterisk and  hylafax  
but I' m still at a loss. For starters, what is the architecture  
that I need?


my application --> QUESTION MARK???  > VOIP Provider ---> PSTN  
---> Fax Machine.


So first question, what should QUESTION MARK be? Is it just  
Asterisk or a combination of Asterisk and something like hylafax  
(fax manager). And depending on that answer, what is the  
configuration that has to be made on it. Even reference to material  
that explains the configuration would be very helpful to me at this  
time.


Thanks in advance for the help,



The missing link might be iaxmodem.  It has two interfaces: IAX  
channel for asterisk, and a serial device (in /dev/) which emulates a  
faxmodem.  Then, fax away using hylafax.  I have tried faxing over  
SIP through a provider (broadvoice) to a coworker's fax on the pstn  
this way, and it worked.  I haven't done any testing in volume, though.


So you would have something like:

Doc -> hylafax -> iaxmodem -> * -> voip provider -> pstn -> fax machine


Phil

PS- I suppose if you had multiple SIP accounts with a provider, you  
could create multiple iaxmodems and do things in parallel (assuming  
enough bandwidth and cpu).


PPS- I hope you're not doing fax-spamming with this set up! ;')
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[Asterisk-Users] Calling number rewriting

[ADEGOKE ARUNA]

Hi all,

I will be glad if I can get response to my need.

What I am trying to do is;

I have a set up where my asterisk box is directly connected to a digitalk IN
platform.

However, between the asterisk and digitalk is a104d sangoma card having e1
with pri. The link between digitalk and my big alcatel switch is e1 with ss7
signalling and this finally lead to my mobile operator

What I am planning to do is to rewrite the caller id so that the calling
number presented to the mobile operator is going to be the number set on the
pri channels

I will be glad, if anyone can just lead me on this

goksie

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[Asterisk-Users] Application Faxing using SIP

[J Poz]

I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution.     My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them.     My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec.     So I've done alot of research on faxing and asterisk and  hylafax but I'
 m still
 at a loss. For starters, what is the architecture that I need?     my application --> QUESTION MARK???  > VOIP Provider ---> PSTN ---> Fax Machine.     So first question, what should QUESTION MARK be? Is it just Asterisk or a combination of Asterisk and something like hylafax (fax manager). And depending on that answer, what is the configuration that has to be made on it. Even reference to material that explains the configuration would be very helpful to me at this time.  Thanks in advance for the help,     J...   
		  
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RE: [Asterisk-Users] Bridged line appearance

[mustardman29]

 
>  1) Yes. There are "plans for it".
GREAT!  What is the current status and expected timeline?
> 
>  2) No. It won't be easy as Asterisk is a multi-protocol PBX 
> and usually when we consider introducing a feature like this, 
> its intent is for it to function across all of the protocols 
> that Asterisk supports, VoIP or not. Everyone else you've 
> mentioned needs only worry about their own device supporting 
> a standard or their own system only supporting devices that 
> they manufacture to support the feature. That makes things 
> somewhat easier for implementation and Asterisk has no such 
> luxury given it's completely open nature which most of us see 
> as an advantage.
Thanks for explaining the details of why it will be difficult
> 
>  3) The other solutions you've mentioned above all have 
> salaried engineering staffs whose job it is to implement 
> features as decided by product management folks also employed 
> by that company who are driven by the comments and feedback 
> of users such as yourself who fork over large sums of money 
> compared to what you pay for your Asterisk to have such 
> solutions. Had you sent such an email to one of these 
> companies at the time you did on a Friday night in the 
> states, my bet is on the fact that it wouldn't have even 
> solicited an initial response from a product management 
> resource until Monday morning.
Ummm.ok.  Asterisk=open source community.  That just goes without
saying.  Other than that I don't know what your point is.  So there are no
salaried software engineers at Digium working on Asterisk?
> 
>  4) The SPA-9000 is devoid of features like, Voicemail, which 
> Asterisk already has. If a system without BLA is a 
> "non-starter" for you and these small business you have 
> cited, why not consider a combined solution where Asterisk 
> provides features (call queues/ACD, voicemail,
> etc) that the SPA-9000 does not have and then you use the 
> SPA-9000 for what it is good for (an IP key system - which is 
> not what Asterisk is)? Asterisk can be whatever and play 
> whatever part you want it to play in your solution. It 
> doesn't have to be the entire solution.
> Because of its open nature, it usually integrates and 
> interoperates well with many existing products/solutions. The 
> SPA-9000 is no exception.
Thanks for pointing out the differences.  Yes, I have thought about creating
a Frankenstein system which takes advantage of the strengths of both the
SPA-9000 and Asterisk.  Perhaps using Asterisk as a POT's gateway and
voicemail server.  The cost starts to creep up though.  This is a concept I
have been mulling over for awhile now.  It remains to be seen what the best
direction is.  When in doubt the best strategy is KISS.  The simplest,
cheapest, and presumably most robust solution is to have everything in one
box.
> 
>  5) There are thousands of small businesses already, my own 
> being one of them, that would disagree that Asterisk is a 
> "non starter" for them. Asterisk is what you make of it, and 
> for us, it's a criticial communications tool for our business.
At the end of the day it is what the user thinks, not the Linux people.  For
you, me and most others on this board I think we can all agree that Asterisk
works just fine for us.  For some companies used to PBX like functionality
it will probably work just fine as well which I have already pointed out.
For many many other companies used to key system like functionality it is a
non-starter mostly because of the lack of BLA IMHO. If you don't believe me
that it is a VERY important feature then ask yourself why a LOT of IP phones
and VoIP systems support it or are starting to support it.  If Asterisk
wants to be a main stream phone system then I feel it should support it.
Has nothing to do with open source vs proprietary.  Just giving my opinion
based on user feedback. 
> 
>  These things being said, what was your original intent for 
> writing such an email? Is there something you'd like to 
> contribute to help get this feature implemented? You don't 
> need to be a developer to contribute. There's testing, 
> documentation, bounties to be set for features one "must 
> have", and all sorts of other areas that could use the 
> assistance of folks like yourself that aren't software developers.
Sure, what is the development schedule?  I get your point.  No need to beat
me over the head with it.  I read these sorts of comments about how it's
"your fault for not being a software coder" and "if you don't like it too
bad, it's your fault for not getting more involved" and frankly I am sick of
it.  We all know this is open source, we mostly all know the advantages and
disadvantages of it and we would not be here if we didn't want it to work.
Let's just move on.  I am sorry for not being able to code.  I am sorry I am
not contributing as much as I should.  It's my fault this feature is not
getting off the ground.  There are you happy?  Can we move on now?

> 
>  Thanks for yo

RE: [Asterisk-Users] Grandstream GXP-2000

[asterisk]

On Fri, 17 Feb 2006, mustardman29 wrote:

The GXP2000 firmware is not bad for features and ease of use but still
buggy.  The hardware is junk to be quite honest and I don't think firmware
will ever fix that.  The Aastra 9133i hardware is 10x better.  The 9133i
firmware is still a work in progress though but they are coming out with new
firmware every few months and each iteration improves the operation.  Long
term I believe any of the Aastra phones are a MUCH better.


why bother with an aastra 9133i when you can have a polycom 501. better 
phone, same price.


-Dan
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RE: [Asterisk-Users] how to add stun functionality in asterisk

[Koopmann, Jan-Peter]

On Friday, February 17, 2006 7:34 PM Matt wrote:

> Yes Sir!   This is what I use:
> http://www.vovida.org/applications/downloads/stun/
> 
> Works like a charm!  Been running it in production for about a year.

Good hint. Can you possibly provide a bit more insight on this? Are you running 
STUN so that your phones behind NAT can easily connect to your server or the 
other way around? I would really like to see the relevant parts of your setup.

Kind regards,
  JP
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Re: [Asterisk-Users] problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)

[nik600]

On 2/17/06, Michael Collins <[EMAIL PROTECTED]> wrote:
> Nik,
>
> This definitely helps!  Please check your dial command. You've got
> "Dial(Zap/0/mynumber)" and I think you might possibly want it to be
> something like this:
> Dial(Zap/1/mynumber)   or
> Dial(Zap/g0/mynumber)
>
> I don't recall there being a zap channel zero, but it is common to have
> a group zero.  I would recommend trying Zap channel 1 -
> Dial(Zap/1/mynumber) - before trying the group.  Again, please get the
> debug info.  The "CHANUNAVAIL" message made it easier to diagnose this
> issue.
>
> Don't give up!  The education you are getting will help you in the long
> run and in a few months you'll be able to help a * newbie with the same
> issues!
>
> -MC
>

ok, thanks for your help, please, be patient because now i've got many
logs to post ... :-)

so, i've made this new entry in extension.conf:

exten => 444,1,Dial(Zap/0/0465670127)
exten => 445,1,Dial(Zap/g0/0465670127)
exten => 446,1,Dial(Zap/1/0465670127)
exten => 447,1,Dial(Zap/g1/0465670127)


and i've reloaded asterisk with:

asterisk -r
>reload
>quit

and then:

 tail -f /var/log/asterisk/full


I CALL 444 or 447:
Feb 18 04:52:54 VERBOSE[5574] logger.c: -- Executing
Dial("SIP/103-c762", "Zap/0/0465670127") in new stack
Feb 18 04:52:54 NOTICE[5574] app_dial.c: Unable to create channel of
type 'Zap' (cause 0 - Unknown)
Feb 18 04:52:54 VERBOSE[5574] logger.c:   == Everyone is
busy/congested at this time (1:0/0/1)
Feb 18 04:52:54 DEBUG[5574] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

I CALL 445 or 446:
Feb 18 04:53:20 VERBOSE[5577] logger.c: -- Executing
Dial("SIP/103-b2ea", "Zap/g0/0465670127") in new stack
Feb 18 04:53:20 VERBOSE[5577] logger.c: -- Making new call for cr 32773
Feb 18 04:53:20 VERBOSE[5577] logger.c: -- Requested transfer
capability: 0x00 - SPEECH
Feb 18 04:53:20 VERBOSE[5577] logger.c: > Protocol Discriminator:
Q.931 (8)  len=43
Feb 18 04:53:20 VERBOSE[5577] logger.c: > Call Ref: len= 2 (reference
5/0x5) (Originator)
Feb 18 04:53:20 VERBOSE[5577] logger.c: > Message type: SETUP (5)
Feb 18 04:53:20 VERBOSE[5577] logger.c: > [Feb 18 04:53:20
VERBOSE[5577] logger.c: > [04Feb 18 04:53:20 VERBOSE[5577] logger.c: >
[04
 03Feb 18 04:53:20 VERBOSE[5577] logger.c: > [04 03 80Feb 18 04:53:20
VERBOSE[5577] logger.c: > [04 03 80 90Feb 18 04:53:20 VERBOSE[5
577] logger.c: > [04 03 80 90 a3Feb 18 04:53:20 DEBUG[3590] channel.c:
Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 DEBUG[3590] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 18 04:53:20 WARNING[3590] channel.c: Avoided initial deadlock for
'0x8c34fb0', 10 retries!
Feb 18 04:53:20 VERBOSE[5577] logger.c: > [04 03 80 90 a3]
Feb 18 04:53:20 VERBOSE[5577] logger.c: > Bearer Capability (len= 5) [
Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
Feb 18 04:53:20 VERBOSE[5577] logger.c: > 
Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
Feb 18 04:53:20 VERBOSE[5577] logger.c: > 
Ext: 1  User information layer 1: A-Law (35)
Feb 18 04:53:20 VERBOSE[5577] logger.c: > [Feb 18 04:53:20
VERBOSE[5577] logger.c: > [18Feb 18 04:53:20 VERBOSE[5577] logger.c: >
[18
 03Feb 18 04:53:20 VERBOSE[5577] logger.c: > [18 03 a9Feb 18 04:53:20
VERBOSE[5577] logger.c: > [18 03 a9 83Feb 18 04:53:20 VERBOSE[5
577] logger.c: > [18 03 a9 83 81Feb 18 04:53:20 VERBOSE[5577]
logger.c: > [18 03 a9 83 81]
Feb 18 04:53:20 VERBOSE[5577] logger.c: > Channel ID (len= 5) [ Ext: 1
 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
Feb 18 04:53:20 VERBOSE[5577] logger.c: >   
ChanSel: Reserved
Feb 18 04:53:20 VERBOSE[5577] logger.c: >   Ext: 1
 Coding: 0   Number Specified   Channel Type: 3
Feb 18 04:53:20 VERBOSE[5577] logger.c: >   Ext: 1
 Channel: 1 ]
Feb 18 04:53:20 VERBOSE[5577] logger.c: > [Feb 18 04:53:20
VERBOSE[5577] logger.c: > [28Feb 18 04:53:20 VERBOSE[5577] logger.c: >
[28
 06Feb 18 04:53:20 VERBOSE[5577] logger.c: > [28 06 64Feb 18 04:53:20
VERBOSE[5577] logger.c: > [28 06 64 65Feb 18 04:53:20 VERBOSE[5
577] logger.c: > [28 06 64 65 76Feb 18 04:53:20 VERBOSE[5577]
logger.c: > [28 06 64 65 76 69Feb 18 04:53:20 VERBOSE[5577] logger.c:
>
 [28 06 64 65 76 69 63Feb 18 04:53:20 VERBOSE[5577] logger.c: > [28 06
64 65 76 69 63 65Feb 18 04:53:20 VERBOSE[5577] logger.c: > [28
 06

Re: [Asterisk-Users] Bridged line appearance

[BJ Weschke]

On 2/18/06, mustardman29 <[EMAIL PROTECTED]> wrote:
> So are there any plans for bridged line appearance support in Asterisk?  The
> new Linksys SPA9000 supports it.  A lot of other VoIP systems from Nortel,
> Sylantro etc. supposedly support it.   Seems to me that Asterisk needs to
> get on the bandwagon or be relegated to call centers, specialized voicemail
> applications, and phone chat businesses.  It's not needed for companies used
> to PBX's but something like 75-95% of all companies are small businesses
> using key systems with BLA type behaviour not PBX behaviour.
>
> Like it or not, the mass market uses and will continue to use BLA or
> whatever they call it in the non VoIP world.  I know that without it,
> Asterisk is a non-starter for most small businesses looking to replace their
> key systems.
>
> I am not a software developer but I remember reading a post by an asterisk
> developer stating that implementing it in Asterisk would be difficult but
> without it I think mass market appeal of Asterisk will be quite limited
> IMHO.  There is a famous quote that states, "nothing worth doing is ever
> easy".
>
> Aastra just released their v1.3.1 firmware which supposedly supports the
> internet draft spec of BLA.  The Polycom phones also supposedly support this
> spec so the ability IS there on the phone side.

 1) Yes. There are "plans for it".

 2) No. It won't be easy as Asterisk is a multi-protocol PBX and
usually when we consider introducing a feature like this, its intent
is for it to function across all of the protocols that Asterisk
supports, VoIP or not. Everyone else you've mentioned needs only worry
about their own device supporting a standard or their own system only
supporting devices that they manufacture to support the feature. That
makes things somewhat easier for implementation and Asterisk has no
such luxury given it's completely open nature which most of us see as
an advantage.

 3) The other solutions you've mentioned above all have salaried
engineering staffs whose job it is to implement features as decided by
product management folks also employed by that company who are driven
by the comments and feedback of users such as yourself who fork over
large sums of money compared to what you pay for your Asterisk to have
such solutions. Had you sent such an email to one of these companies
at the time you did on a Friday night in the states, my bet is on the
fact that it wouldn't have even solicited an initial response from a
product management resource until Monday morning.

 4) The SPA-9000 is devoid of features like, Voicemail, which Asterisk
already has. If a system without BLA is a "non-starter" for you and
these small business you have cited, why not consider a combined
solution where Asterisk provides features (call queues/ACD, voicemail,
etc) that the SPA-9000 does not have and then you use the SPA-9000 for
what it is good for (an IP key system - which is not what Asterisk
is)? Asterisk can be whatever and play whatever part you want it to
play in your solution. It doesn't have to be the entire solution.
Because of its open nature, it usually integrates and interoperates
well with many existing products/solutions. The SPA-9000 is no
exception.

 5) There are thousands of small businesses already, my own being one
of them, that would disagree that Asterisk is a "non starter" for
them. Asterisk is what you make of it, and for us, it's a criticial
communications tool for our business.

 These things being said, what was your original intent for writing
such an email? Is there something you'd like to contribute to help get
this feature implemented? You don't need to be a developer to
contribute. There's testing, documentation, bounties to be set for
features one "must have", and all sorts of other areas that could use
the assistance of folks like yourself that aren't software developers.

 Thanks for your initial feedback and we look forward to your
continued contributions to the Asterisk community.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
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