Re: [Asterisk-Users] newbie question; can * screen calls?

[Ken Alker]

--On Friday, January 09, 2004 10:11 PM -0600 Alan Andrews 
[EMAIL PROTECTED] wrote:

On Fri, 2004-01-09 at 20:55, Ken Alker wrote:
Does * have the capability to screen calls?  IOW, if someone calls in
from  outside (ie. not a local extension), can * ask the calling party
to state  their name, record it, ring the recipient, play the caller's
name for the  recipient, then give the recipient the choice of answering
or forcing the  call to voice mail?
I thought that's what caller ID was for.
There are many cases where caller ID will not suffice:

1) many people share the same phone number (a family, or roommates)
2) a company where their entire group of phone numbers appears as one 
calling number (thus, you don't know who it is within the company that is 
calling)
3) someone calling from a number that isn't theirs (payphone, friend's 
house, borrowed cell, work cell, etc)
4) CallerID is blocked by caller
5) In my area caller ID is about $7.50/mo./line which makes it priced too 
high to be a justifiable expense for my company.

I find that callerID is only effective in about 25% of the cases (I have it 
at home).  If you don't have callerID, automated call screening is the next 
best thing.  In fact, I have nearly a 100% success rate with it, so it's 
better than callerID, IMHO.  I use this feature on a Nortel mudular ICS. 
They refer to it as screened transfer.  It saves me from having to speak 
with sales people who make it past my employee barricade, or who figure out 
my direct extension.  I'd guess it saves me an average of 20 minutes per 
day; not bad if you add it up.

/**
Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU
Impulse Internet Services   http://www.impulse.net
Santa Barbara,  San Luis Obispo,  Ventura, Los Angeles, Orange
T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo
***/
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Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

[Chris Albertson]

--- Steve Underwood [EMAIL PROTECTED] wrote:
 WipeOut wrote:
 
  Granted five 9's is never easy but in a cluster of 10+ servers the 
  system should survive just about anything short of an act of God..
 
 You do realise that is a real dumb statement, don't you? :-)
 
 A cluster of 10 machines, each on a different site. Guarantees from
 the 
 power company - checked personally to see that aren't cheating - that
 
 you have genuinely independant feeds to these sites. Large UPSs, with
 
 diesel generator backups. Multiple diverse telecoms links between the

If he says cluster he likely means 10 servers in one rack.  But still
you are right.  It is all the other stuff that could break.  You
will need paralleld Ethernet switches (Yes they make these, no, they
are NOT cheap.) you will need some kind of fail over.  The switches
can do that for you. (do a google on level 3 switch)

It's the level three switches that make .9 possible but half or
more of your hardware will be just hot spares so it really will
take a rack full of boxes

Each box should have mirrored drives and dual power supplies and each
AC power cord needs to go to it's own UPS

Has anyone tried to build Asterisk on SPARC/Solaris?  One SPARC
server is almost five nines all by itself as it can do thinks
like boot around failed CPU, RAM or disks.  I've actually
pulled a disk drive out of a running Sun SPARC and applications
continoued to run. 



=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
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[Asterisk-Users] picking a channel bank

[Ken Alker]

I have never had to pick out a channel bank before but I'd like to use one 
with the Digium T-1 card to hook 8 analog CO lines to an * PBX.

Is there a reference somewhere describing and comparing channel banks (old 
and new)?

Can modern channel banks handle translating all the new analog signaling 
features into a T-1 format?  For example, can it interpret the 1200 baud 
FSK caller ID stream that is inserted between the first and second ring and 
translate that into digital caller ID delivery out the RJ-45 port?

How about:

1) caller ID
2) caller ID call waiting
3) distinctive rings
4) call waiting
5) analog 3-way calling (flash hook)
6) analog call transfer (3-way call w/hang up)
7) stutter dial-tone (message waiting)
8) anything else I've missed?
/**
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Impulse Internet Services   http://www.impulse.net
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Re: [Asterisk-Users] file_inlcude .. why not?

[Olle E. Johansson]

Lion Templin wrote:

TeleSIP wrote:

Don't know if this has been addressed, but why isn't there a
file_include style directive for extensions.conf?


there is...search the archives or the wikiits something like #include
filename.conf
Oh yeah, it works, thanks ..

Not entirely obvious, I guess .. I thought it would have taken the form 
of the other directives.  NBD.
Added patch of extensions.conf.sample to bugs.digium.com to cover the
#include statement.
Already covered in the wiki page Asterisk config files

/O

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[Asterisk-Users] Music_on_hold adjust volume

[Ernst Lehmann]

Hi all,

is there a posibility to change the volume of the music-on-hold ??

I tried with the different groups with default, and loud setup, but no
changes.

And the music is a little bit to loud ??

Are there any options, to deal with ?? Or do I have to recode my mp3 in
any way ??

Thanks for any help


---

Bye

Ernst

Email: [EMAIL PROTECTED]



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Re: [Asterisk-Users] ChanIsAvail and SIP

[Olle E. Johansson]

B. J. Bomar wrote:

Hello all.  Has anyone had any success using ChanIsAvail with only SIP 
channels?  Is there another, better way to check if an extension is busy 
without dialing it?
Well, SIP devices live their own life and should really handle this signalling
themselves. That's why ChanIsAvail does not really work with SIP channels,
Asterisk does not control what is happening out there in the wild. The SIP
channel is really a compromise from a business PBX point of view, where you
want to know what is happening out there, which lines are occupied etc etc.
I think that you can use incominglimit and outgoinglimit to limit the number of
calls asterisk place to a SIP device and force busy if there's already a call
going on.
Remember that this limits the number of connections to/from Asterisk, not necessarily
the number of calls on the SIP device.
/O

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Re: [Asterisk-Users] AbsoluteTimeout Users Messages

[WipeOut]

Olle E. Johansson wrote:

Andy Powell wrote:

Nicolas,

I'd appreciate a copy of this if possible... got a url where I can 
grab it?

Thanks

Andy

*** REPLY SEPARATOR  ***

On 09/01/2004 at 10:43 Nicolas Gudino wrote:


Andy Powell wrote:


I'd be nice to be able to play a tone (or message) at 
AbsoluteTimeout -

N
where N is a number os seconds before the cut-off... a bit like pay 
phones
(used?) to do...

I have implemented an 'horrible' patch that sort of works. I'm not very
good
at C, and I'm new to asterisk. It makes a tone at 40, 30, 20 and 10 
second
before absolute-timeout. I can provide you with the patch, but its 
really
really ugly, with lots of if/endifs.

Please add the ugly patch to bugs.digium.com - maybe someone else will 
take it
up and clean your code. Not everything on bugs need to be clean at 
start, but
it's good to have it in the repository. If it's that ugly, there's no 
chance
of it getting into CVS until someone cleans it up :-)

Thank you for your contribution.
/O
And make sure to send in a  disclaimer otherwise it will not even be 
looked at.. :)

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Re: [Asterisk-Users] Chagres Technologies, Inc

[Olle E. Johansson]

Mail John Brown at Chagres. [EMAIL PROTECTED]

He usually responds quickly and I get information about where my products are.
Yes, I also have rest orders, but I have acceptable responses on why and when
they are expected to arrive in this snowy winterland...
/O

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Re: [Asterisk-Users] Called Party Identification

[Olle E. Johansson]

Steven Critchfield wrote:

On Fri, 2004-01-09 at 22:40, Brent Franks wrote:

Does * support Called Party Identification?  Say for example, I dial
extension 2000, SIP sends back John Doe from the sip.conf file where
extension 2000 is defined?  Would this violate the SIP RFC?


Maybe you didn't think about the fact that extensions aren't defined in
sip.conf. Also it is possible for many extensions to end up on any
physical phone. So sending essentially caller ID back to the calling
phone doesn't really make sense. 
Agreed, the SIP channel doesn't really now anything about extensions, until
called. But when getting a call, we match with a user/peer and could in
theory send back a name. I don't know if I want this, though, of privacy
reasons. Maybe when I accept a call. And I haven't checked the RFCs on
where this should be placed in the SIP headers. Interesting question.
Propably belongs in the 200 OK message.
Anyone else?

/O

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Re: [Asterisk-Users] Called Party Identification

[Olle E. Johansson]

Brent Franks wrote:

No, but the Caller ID Information for a SIP extension is stored in
sip.conf, so yes, I did think about that.
As far as making sense, many meridian systems do this, and it is quite
helpful.  This could help with the implementation of gastman, and also
end user phones.  On the Cisco's and Polycom's, when you place a call on
hold, rather than seeing an extension, you would see the name and you
could toggle between the calls and see the name, rather than number
(O.K. that part is a convenience thing).  I know on my Meridian system
at work, if you accidentally dial the wrong extension, the name pops up
after it starts ringing, and you know your calling the wrong person.
You can hang up, or tell the person real quick, hey sorry, I meant to
call someone else.
It's one thing when you have an internal PBX, but when you open up for
external SIP calls from the Internet - do you really want them to always
get your full name?
Maybe a filter would be good.
Anyway, could you provide a SIP trace of a call setup with this feature?

/O

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Re: [Asterisk-Users] disclaimed or not - that's the question /* New subject */

[Olle E. Johansson]

WipeOut wrote:



And make sure to send in a  disclaimer otherwise it will not even be 
looked at.. :)

How do we know what is disclaimed or not disclaimed?
/O
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RE: [Asterisk-Users] file_inlcude .. why not?

[Senad Jordanovic]

Lion Templin wrote:
 Don't know if this has been addressed, but why isn't there a
 file_include style directive for extensions.conf?
 
 I find that my extensions.conf grows a lot, and it would be a lot
 nicer to have a tree of files rather than one big file to try and
 navigate. Also, I've got a couple different 'systems' running
 concurrently on one asterisk box (ie, completely different groups of
 people with different I/O lines) and would like to break the config
 into seperate files that could be maintained by seperate people
 without having to expose the entire system to everyone.
 
 Suggestions?
 
 
 Lion Templin

I would agree with this as well. This way it should be much easier to
provide virtual asterisk services!
We all agree that * will be apache of VOIP! :)  Well, apache has virtual
directives and include directives!!!

Ta
SJ

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RE: [Asterisk-Users] USA dial plan

[Senad Jordanovic]

Title: Message



Yes, in most places in the USA local 
calls are totally free, no per mincharge.
This is not true 
in the US for business lines. Residentiallines have a "free" local 
calling area. However, business lines from an incumbent local exchange 
carrier like SBC nearlyalways charge rates for 7-digit local calls, 
usually, but not alwaysbased on mileage zones. Rates vary based on 
the local carrier, time of dayand the distance. Different rate 
schemes apply in different parts of the country. Some use ZoneUsage 
Measured(ZUM) schemes, others use Flat Rate or Measured Rate schemes. 
There are different rate plans for the same carriers for local toll 
callsthat fall outside the local calling area but are within the same 
LATA.

Some states do 
allow 10-digit dialing without a 1+. Washington DC (202) is an example of 
this for making local calls to other adjacent area codes. The entire North 
America Numbering Plan (NANP) is in a constant state of change as new area codes 
are added. There are 4 different dialing plans for each area code that can 
vary with regard to the number of digits required and whether a 1+ is 
required:

  
  - Home NPA Local Calls
  - Foreign NPA Local Calls
  - Home NPA Toll Calls
  - Foreign NPA Toll Calls
  
  If you go to www.nanpa.com and click on the "Dialing Plans" 
  option in the left column, you can get the current ("Standard") and evolving 
  ("Permissive") dialing plan for any area code. For example, 310 is 
  currently setup this way:
  
  
  Dialing 
  PlanStandard 
  Permissive
  - Home NPA Local 
  Calls 
  7D 
   1+10D
  - Foreign NPA Local 
  Calls 
  1+10D 
  NA
  - Home NPA Toll 
  Calls 
  7DNA
  - Foreign NPA Toll 
  Calls 
  1+10D
  
  This says you currently dial local calls within 310 as 7 
  digits, but the plan will change to require 1+10 digits which is currently 
  permitted.
  
  Hope this helps 
  David Schlossman
  [EMAIL PROTECTED]
  
  
  hmmm... Damn Outlook.. It wont do the quoata again.. Sorry about 
  this.!!!
  
  Anyway,
  
  Dabid, do you know whenalllocal calls will have to be 
  dialed with "1" appended?
  
  Ta
  SJ
  
  

Re: [Asterisk-Users] disclaimed or not - that's the question /* New subject */

[WipeOut]

Olle E. Johansson wrote:

WipeOut wrote:



And make sure to send in a  disclaimer otherwise it will not even be 
looked at.. :)

How do we know what is disclaimed or not disclaimed?
/O
Digium have all the Disclaimers and will not develop or include any code 
into the CVS without one.. Thats all I was saying.. :)

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[Asterisk-Users] Call transfer message

[Senad Jordanovic]

Hi all,

A feature I think should be included in 1.0 version is playing 
a message to calling and called party while the call is being
transferred.

Something like this:

Calling party (whose call is being transferred)
Please wait, your call is being transferred

Called party (who is transferring call)
You have successfully transferred last call
OR (in case of failure)
You have not successfully transferred current call. Please try again.

Is this feature present?

Ta
SJ

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Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

[Steve Underwood]

Hi,

I don't want to drag this into a long thread, but note the original says 
the system should survive just about anything short of an act of God, 
and suddenly you are talking about a reliable server and a few switches. 
These are quite different things. I have yet to see a 5 x 9's server 
room. Fire, mechanical damage and other factors will normally keep the 
location itself well below 5 x 9's. Think system instead of server 
equipment, and the picture looks very different. Even for a single PC 
type server, downtime due to telecoms lines, power problems, fire, 
flood, typhoon damage, theft and a mass of other stuff mught well exceed 
the server unavailablility itself. I've seen many servers not fail in 5 
years. I have yet to see the best location go that long without causing 
at least one substantial period of downtime. 5 x 9's allows about 6 
minutes downtime a year. That means 100% of all failures must have 
automated failover, as manuals repair could never be achieved so fast. 
Physical diversity if essential for that.

Regards,
Steve
Chris Albertson wrote:

--- Steve Underwood [EMAIL PROTECTED] wrote:
 

WipeOut wrote:

   

Granted five 9's is never easy but in a cluster of 10+ servers the 
system should survive just about anything short of an act of God..
 

You do realise that is a real dumb statement, don't you? :-)

A cluster of 10 machines, each on a different site. Guarantees from
the 
power company - checked personally to see that aren't cheating - that

you have genuinely independant feeds to these sites. Large UPSs, with

diesel generator backups. Multiple diverse telecoms links between the
   

If he says cluster he likely means 10 servers in one rack.  But still
you are right.  It is all the other stuff that could break.  You
will need paralleld Ethernet switches (Yes they make these, no, they
are NOT cheap.) you will need some kind of fail over.  The switches
can do that for you. (do a google on level 3 switch)
It's the level three switches that make .9 possible but half or
more of your hardware will be just hot spares so it really will
take a rack full of boxes
Each box should have mirrored drives and dual power supplies and each
AC power cord needs to go to it's own UPS
Has anyone tried to build Asterisk on SPARC/Solaris?  One SPARC
server is almost five nines all by itself as it can do thinks
like boot around failed CPU, RAM or disks.  I've actually
pulled a disk drive out of a running Sun SPARC and applications
continoued to run. 
 



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Re: [Asterisk-Users] Forums Need Help

[Olle E. Johansson]

Steve Totaro wrote:
check it out at www.asteriskhelpdesk.com/forums 
http://www.asteriskhelpdesk.com/forums
I hope that you are aware there already are one or several forums,
mostly ignored by the community.
See http://asterisk.xvoip.com/
Xvoip also tried setting up a business list for Asterisk.
If you want to split up, make sure you limit the topic and attract the audience.

Good luck and thank you for assisting with the community!

/Olle

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Re: [Asterisk-Users] (no subject)

[Jeremy McNamara]

T. Chan wrote:

I recently came across DynEXTENdb, a way to be able to include 
thousands of Extensions (routes). In my application which is VOIP, we 
need to include more than 50,000 area codes due to the USA LATA 
routing, and there is simply no way to do that with extensions.conf. 
The way DynEXTENdb is designed seems to be a possible way to make it 
happen, I wonder if any of you Asterisk colleagues have had a chance 
to try it out and how reliable it is. Thanks for your information.

It is the absolute wrong way to deal with dynamic extensions.   Also, 
why in hell do you need so many extensions?  Apparently you are not 
tackling the problem appropriately.

Jeremy McNamara

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Re: [Asterisk-Users] USA dial plan

[Eric Wieling]

In some places, yes, but not all places.  In Louisiana, for example 
business can get unlimited local calling (and most do).  When I lived in 
Calif unlimited local calling was not available to businesses.

Scott Stingel wrote:

Just a little clarification on USA local calling:

Local calls are generally free for residential customers, unless they are on
a increasingly rare measured local service.  However, business customers
almost always pay for local calls on a measured basis.
Regards
Scott
Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England

Email:  scott at evtmedia.com  
URL:www.evtmedia.com  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Friday, January 09, 2004 11:53 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] USA dial plan
Generally speaking, Yes. The usual dial plan in the USA is as follows:

NXX- (Free Local Call to number in same Area Code)
NXX-NXX- (Free Local Call to number in different Area Code)
1-NXX- (Toll Call to number in same Area Code)
1-NXX-NXX- (Toll Call to number in different Area Code)
1-800-NXX- (Toll Free Call)
1-855-NXX- (Toll Free Call)
1-866-NXX- (Toll Free Call)
1-877-NXX- (Toll Free Call)
1-888-NXX- (Toll Free Call)
Yes, in most places in the USA local calls are totally free, no per min
charge.
Some parts of the USA have Local Toll Calls, that is calls that are
dialed as NXX- that are not free, but have a very small per min
cost.  Los Angels is one of these places I think.
On Fri, 2004-01-09 at 12:50, Senad Jordanovic wrote:

Hi,

Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with 1 in order 
To successfully make a call to other USA destinations?


I have not been to USA (yet) :)
Ta
SJ
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Re: [Asterisk-Users] AbsoluteTimeout Users Messages

[Nicolas Gudino]

  Andy Powell wrote:
 
  Nicolas,
 
  I'd appreciate a copy of this if possible... got a url where I can 
  grab it?
 
  Thanks

You can grab a copy from the bugtracker:

http://bugs.digium.com/bug_view_page.php?bug_id=773

I've already sent the disclaimer to Digium..

Best regards,

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RE: [Asterisk-Users] Mailing list growth

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Philipp von Klitzing
 Sent: Saturday, January 10, 2004 10:35 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Mailing list growth
 
 
 - asterisk-users: VoIP and Asterisk in general (including newbies)
 - asterisk-tdm: Use if part of your problem/question involves T1/TDM
 - asterisk-biz: new topics, not yet really covered on -users
 
 Effects:
 - newbies only need to subscribe and read a lower volume -users
 - all readers have the same amount of traffic, but get some nice 
 filtering help at least

Reasonable, but may need some serious topic policing at first (requiring
multiple list admins per list), again due to the fact that people often
will not know where their problem lies.

Also, just as an example.the VoIP list would have discussions on it
like the recent calling card appwell, that doesn't sounds newbieish
at all.

Has anyone actually taken the time to do a message/category
classification and breakdown to see if the proposed split even makes
sense?  Would we end up with 10 messages a day in -biz, 25 or so in -tdm
and 100 in -users?

 As Robert pointed out LISTSERV has some nice topic features 
 that could 
 help, however the license ist costly (we have two LISTSERVs 
 running). Let 
 me add, though, that besides topic management LISTSERV can 
 also provide 
 super lists that are great to fight cross-postings - super 
 lists group 
 one or more normal lists or super lists. My guess is that 
 there are other 
 MLMs out there that have similar features.

LISTSERV is evil, and yes, there are (listserv is evil mostly because of
the abhorrent cost of something that is available via open source/free
alternatives and a couple of perl/awk/sed scripts).

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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Re: [Asterisk-Users] Mailing list growth

[admin]

everything is free or the cost of shipping if you think...

dont worry, newbs will land at my forums but i still wanna know if i can cut
and paste FAQs and the like.  I plan on it so sue me, rofl.


- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 10:55 AM
Subject: RE: [Asterisk-Users] Mailing list growth


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Philipp von Klitzing
 Sent: Saturday, January 10, 2004 10:35 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Mailing list growth


 - asterisk-users: VoIP and Asterisk in general (including newbies)
 - asterisk-tdm: Use if part of your problem/question involves T1/TDM
 - asterisk-biz: new topics, not yet really covered on -users

 Effects:
 - newbies only need to subscribe and read a lower volume -users
 - all readers have the same amount of traffic, but get some nice
 filtering help at least

Reasonable, but may need some serious topic policing at first (requiring
multiple list admins per list), again due to the fact that people often
will not know where their problem lies.

Also, just as an example.the VoIP list would have discussions on it
like the recent calling card appwell, that doesn't sounds newbieish
at all.

Has anyone actually taken the time to do a message/category
classification and breakdown to see if the proposed split even makes
sense?  Would we end up with 10 messages a day in -biz, 25 or so in -tdm
and 100 in -users?

 As Robert pointed out LISTSERV has some nice topic features
 that could
 help, however the license ist costly (we have two LISTSERVs
 running). Let
 me add, though, that besides topic management LISTSERV can
 also provide
 super lists that are great to fight cross-postings - super
 lists group
 one or more normal lists or super lists. My guess is that
 there are other
 MLMs out there that have similar features.

LISTSERV is evil, and yes, there are (listserv is evil mostly because of
the abhorrent cost of something that is available via open source/free
alternatives and a couple of perl/awk/sed scripts).

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp
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RE: [Asterisk-Users] file_inlcude .. why not?

[Jared Smith]

 I would agree with this as well. This way it should be much easier to
 provide virtual asterisk services!
 We all agree that * will be apache of VOIP! :)  Well, apache has virtual
 directives and include directives!!!
 
 Ta
 SJ

If you understand contexts and how to use them correctly, you don't need
virtual directives.  You can *easily* achieve the same effect with
contexts.

Jared

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RE: [Asterisk-Users] E100P - Error 500

[Scott Stingel]

I get lots of these in a very busy system, along with PRI frame
errors/retransmissions.  It is my understanding that this is due to an
inadequate buffering mechanism in asterisk.  Mark Spencer is aware of the
problem, and has said he'll work on it soon.

In small numbers, these can be safely ignored.

Regards

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]   
URL:www.evtmedia.com http://www.evtmedia.com   



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bichara
Sent: Saturday, January 10, 2004 3:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] E100P - Error 500


Hi,

I am running * with E100P board. At least every our I got an Error 500 
message and ISDN-PRI restarts:

Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 13:33:48 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500


Any clue?

Daniel

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Re: [Asterisk-Users] crontab

[Philipp von Klitzing]

oHi!

 Ladies and Gentlemen, can anyone please help and let me know what is
 the way to start Asterisk automatically using a cronjob, thanks 

http://www.voip-info.org/wiki-Asterisk+administration

Philipp




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Re: [Asterisk-Users] Fwd: new cvs build failure

[Martin]

On Saturday 10 January 2004 12:36 am, Martin wrote:


 I did a previous  find / -name install and it couldn't find it, but I just 
 couldn't believe it.
 The only thing I did recently was a kernel upgrade from 2.4.21-0.25mdk to 
 2.4.21-0.27mdk but via rpmdrake. Did mandrake really remove it ???
 
 Martin


Hello.  Update and end of thread.

I found a newer coreutils-5.0-6.1.92mdk

Now its working again.

Go figure.

Thanks Paul.

Regards...Martin
-- 
A penny saved is a penny taxed.

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Re: [Asterisk-Users] E100P - Error 500

[Daniel Bichara]

Scott Stingel wrote:

I get lots of these in a very busy system, along with PRI frame
errors/retransmissions.  It is my understanding that this is due to an
inadequate buffering mechanism in asterisk.  Mark Spencer is aware of the
problem, and has said he'll work on it soon.
In small numbers, these can be safely ignored.
 

Hi Scott,

But sometimes it closes or destroys all open Zap channels (put call 
onhook).

Daniel

Regards

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]   
URL:www.evtmedia.com http://www.evtmedia.com   



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bichara
Sent: Saturday, January 10, 2004 3:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] E100P - Error 500
Hi,

I am running * with E100P board. At least every our I got an Error 500 
message and ISDN-PRI restarts:

Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 13:33:48 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500

Any clue?

Daniel

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[Asterisk-Users] Oops!

[Terence Parker]

Didn't realise that replies are still tagged to specific threads in the 
mail headers. Oops!

A few of my postings so far have been replies (to save me retyping the 
list address) - but aren't really replies (they are completely off 
topic).

Hope this doesn't cause too many problems in the archives!

But... at least now I know!

Terence

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Re: [Asterisk-Users] picking a channel bank

[TC]

 Is there a reference somewhere describing and comparing channel banks (old
 and new)?
not comprehensive  but a start http://voip-info.org/wiki-Asterisk+Hardware
 Can modern channel banks handle translating all the new analog signaling
 features into a T-1 format?  For example, can it interpret the 1200 baud
 FSK caller ID stream that is inserted between the first and second ring
and
 translate that into digital caller ID delivery out the RJ-45 port?
yes that an issue you need to watch out for

 How about:

 1) caller ID
 2) caller ID call waiting
yea this you need to verify
 3) distinctive rings
 4) call waiting
 5) analog 3-way calling (flash hook)
 6) analog call transfer (3-way call w/hang up)
 7) stutter dial-tone (message waiting)
4-7 are mostly handled
 8) anything else I've missed?
1) far end disconnect supervision on fxo cards
2) echo echo echo echo echo
some like the ADIT 600 have a feature called dynamic impedance matching
which realy seems to fix this cause echo at the souce
3) RF interferance
I have had my local AM radio station play on my FXO cards
4) Flexibility
how do you expand in groups of 2,4,8,12 or are they fixed at 24 ports
(fxs or fxo)
5) Support
Some vendors if buying in the after market will plain not support used
gear
some will only provide support usually = in value to what you pay for
the channel bank
even if its only a manual you want
6) Price :)
If buying new this gear is way over priced, eg a new ADIT 600 with 24fxo
2fxs i got quoted
$4500us



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Re: [Asterisk-Users] ChanIsAvail and SIP

[Philipp von Klitzing]

Hi!

  Hello all.  Has anyone had any success using ChanIsAvail with only SIP 
  channels?  Is there another, better way to check if an extension is busy 
  without dialing it?
 
 Well, SIP devices live their own life and should really handle this signalling
 themselves. That's why ChanIsAvail does not really work with SIP channels,
 Asterisk does not control what is happening out there in the wild.

With the Manager API you have lots of options - probably ExtensionState 
could be one way for you to get closer to a solution.

An easier solution might be to employ AGI and use CHANNEL STATUS 
[channelname], provided this works with SIP and not only Zap (I just 
don't know). 

But first you'd need to find out about the channel name though since SIP 
channels have this random numbering: A show channels or sip show 
inuse at the CLI can provide that, and you can issue those commands also 
remotely from any script using asterisk -rx command and parse the 
result.

You could also use database show SIP/Registry on the CLI to see who is 
registered and who is not before attempting to place a call.

I probably missed a million other ways (that could include your own 
little SIP protocol query sent to the desired destination, for example). 
Just keep in mind that the SIP client can be busy even though for 
Asterisk it is not.

Cheers, Philipp


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Re: [Asterisk-Users] At last!!! :)

[CW_ASN]

Jess:

Try with:

Dial(SIP/[EMAIL PROTECTED],20,t)

Remove 'r' option from your dial command, maybe 'show application Dial' from
CLI could help you more.

Regards,

Gus


- Original Message -
From: Jess Magnaye
To: [EMAIL PROTECTED]
Sent: Friday, January 09, 2004 7:55 PM
Subject: [Asterisk-Users] At last!!! :)


I can smile now.  I made my * work with my Cisco. Finally.  First problem
was Ethernet on my Linux.  After installing * on a different machine, I got
rid of that icmp udp unreachable error.  My next problem was the call
stays on on Cisco gateway, but the ATA drops it.  I figured out it was my
mistake on dialplan in extensions.conf --- (it took me a day to notice it..
damn!).  my config was: exten=_.,1,Dial(SIP/[EMAIL PROTECTED],tr).  The
reason why my ATA is getting fast busy (or dropping the call immediately)
while Cisco gateway (myprovider) is trying to connect my call, was that I am
missing the seconds parameter.  When I changed this to
Dial(SIP/[EMAIL PROTECTED],20,tr), I was able to connect.

There is one little problem left though.  How come after I diale the number
from ATA, I am getting false ringback.  I meant, local ringback from ATA,
instead of the ringback coming from my Cisco (myprovider).

I appreciate any bright ideas and advise from anybody.

Thank you and have a happy weekend!



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[Asterisk-Users] R2 Digital - Brazil

[Daniel Bichara]

Hi all,

I will start testing libr2 for brazilian R2. Any clue?

Daniel

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Re: [Asterisk-Users] crontab

[info-lists]

Philipp von Klitzing said:
 oHi!

 Ladies and Gentlemen, can anyone please help and let me know what is
 the way to start Asterisk automatically using a cronjob, thanks

 http://www.voip-info.org/wiki-Asterisk+administration

 Philipp



Guess maybe I don't leave my system running long enough for it to crash
but seems to me that if the Asterisk process is crashing that we should
fix the reason it stops and not just keep on restarting it. On the WiKI
there are some writeups of fairly large installations.   Are they also
crashing?

Robert
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[Asterisk-Users] Record all phone calls

[Jimmy Riley]

I want to record all phone calls made inbound and outbound.
I'm new so having a hard time getting this started. Here is what I have
so far but isn't working. Can someone help me out? Thanks,



[macro-record-on]
exten = s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1})
exten = s,2,Monitor(wav,${CALLFILENAME})




[sip]
include = macro-record-on
include = iaxtel 
exten = _,1,macro(record-on,${EXTEN},${CALLERIDNUM})
exten = 1001,1,Dial(SIP/one|20|tr)
exten = 1001,2,VoiceMail,u1001
exten = 1001,102,VocieMail,b1001
exten = 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001
exten = 1002,1,Dial(SIP/two|20|mtr)
exten = 1002,2,VoiceMail,u1002
exten = 1002,102,VoiceMail,b1002
exten = 6001,1,Ringing
exten = 6001,2,Wait(2)
exten = 6001,3,VoicemailMain

RE: [Asterisk-Users] Chagres Technologies, Inc

[mikeu]

My experience has been one of unresponsiveness to my e-mails.  I have
ordered and received devices from other providers in the time I have been
waiting for Chagres.  As of now, based on my experiences and those of others
that I have heard from I would highly recommend avoiding Chagres and Mr.
Brown.  All I want now is a refund.

Mike


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Saturday, January 10, 2004 3:22 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc

Mail John Brown at Chagres. [EMAIL PROTECTED]

He usually responds quickly and I get information about where my products
are.
Yes, I also have rest orders, but I have acceptable responses on why and
when
they are expected to arrive in this snowy winterland...

/O

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Re: [Asterisk-Users] E1 - E100P connected to Cisco - problem - HELP

[Daniel Bichara]

Stephen J. Wilcox wrote:

  You are trying to have both ends act as users, cisco can support emulating a 
network interface (isdn protocol-emulate in serial interface config) but in my 
experience i could get the circuit up but it would bounce and i couldnt get 
signalling to work.. to be fair my IOS is quite old and wouldnt support the 
switch types that I think I needed
  


Hi,

I really need help with this:


My circuit is up but I have signalling problem. Port becomes up and
down, up and down, I changed something at Cisco conf, please give
me a clue. (I have some "-- T200 counter expired, What to do..." and
other T203 messages, could it be timing problem?). Also, I am sending
my PRI INTENSE DEBUG output.

My New Cisco Conf:

!
Cisco AS5300 - ios c5300-is-mz.122-5.bin 

isdn switch-type primary-ni 
isdn voice-call-failure 0



controller E1 3

framing
NO-CRC4 
pri-group timeslots 1-31



interface Serial3:15

no
ip address 
isdn
switch-type primary-ni

isdn
protocol-emulate network 
isdn
guard-timer 2000 
isdn
T203 1 
isdn
T306 3000 
isdn
T310 6 
isdn
bchan-number-order ascending

no cdp enable



My INTENSE debug output:

 [
 [00
 [00 01
 [00 01 01
 [00 01 01 01
 [00 01 01 01 ]
 [00 01 01 01 ]
 Supervisory frame:
 SAPI: 00 C/R: 0 EA: 0
 TEI: 000 EA: 1
 Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
 N(R): 000 P/F: 1
 0 bytes of data
-- Restarting T203 counter

 [
 [00
 [00 01
 [00 01 00
 [00 01 00 00
 [00 01 00 00 08
 [00 01 00 00 08 02
 [00 01 00 00 08 02 00
 [00 01 00 00 08 02 00 00
 [00 01 00 00 08 02 00 00 46
 [00 01 00 00 08 02 00 00 46 18
 [00 01 00 00 08 02 00 00 46 18 03
 [00 01 00 00 08 02 00 00 46 18 03 a9
 [00 01 00 00 08 02 00 00 46 18 03 a9 83
 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c
 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79
 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01
 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 80
 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ]
 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ]
 Informational frame:
 SAPI: 00 C/R: 0 EA: 0
 TEI: 000 EA: 1
 N(S): 000 0: 0
 N(R): 000 P: 0
 13 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
 Protocol Discriminator: Q.931 (8) len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
 ChanSel: Reserved
 Ext: 1 Coding: 0 Number Specified
Channel Type: 3
 Ext: 1 Channel: 28 ]
 Restart Indentifier: [ Ext: 1 Spare: 0 Resetting Indicated
Channel (0) ]
-- T200 counter expired, What to do...
-- Retransmitting 17 bytes

 [
 [00
 [00 01
 [00 01 00
 [00 01 00 01
 [00 01 00 01 08
 [00 01 00 01 08 02
 [00 01 00 01 08 02 00
 [00 01 00 01 08 02 00 00
 [00 01 00 01 08 02 00 00 46
 [00 01 00 01 08 02 00 00 46 18
 [00 01 00 01 08 02 00 00 46 18 03
 [00 01 00 01 08 02 00 00 46 18 03 a9
 [00 01 00 01 08 02 00 00 46 18 03 a9 83
 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c
 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79
 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01
 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 80
 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ]
 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ]
 Informational frame:
 SAPI: 00 C/R: 0 EA: 0
 TEI: 000 EA: 1
 N(S): 000 0: 0
 N(R): 000 P: 1
 13 bytes of data
-- Rescheduling retransmission (1)

 [
 [00
 [00 01
 [00 01 01
 [00 01 01 03
 [00 01 01 03 ]
 [00 01 01 03 ]
 Supervisory frame:
 SAPI: 00 C/R: 0 EA: 0
 TEI: 000 EA: 1
 Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
 N(R): 001 P/F: 1
 0 bytes of data
-- ACKing all packets from 0 to (but not including) 1
-- ACKing packet 0, new txqueue is -1 (-1 means empty)
-- Since there was nothing left, stopping T200 counter
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter

 [
 [02
 [02 01
 [02 01 00
 [02 01 00 02
 [02 01 00 02 08
 [02 01 00 02 08 02
 [02 01 00 02 08 02 80
 [02 01 00 02 08 02 80 00
 [02 01 00 02 08 02 80 00 4e
 [02 01 00 02 08 02 80 00 4e 18
 [02 01 00 02 08 02 80 00 4e 18 03
 [02 01 00 02 08 02 80 00 4e 18 03 a9
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 80
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 80 ]
 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 80 ]
 Informational frame:
 SAPI: 00 C/R: 1 EA: 0
 TEI: 000 EA: 1
 N(S): 000 0: 0
 N(R): 001 P: 0
 13 bytes of data
-- ACKing all packets from 0 to (but not including) 1
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
 Protocol Discriminator: Q.931 (8) len=13
 Call Ref: len= 2 (reference 32768/0x8000) (Terminator)
 Message type: RESTART ACKNOWLEDGE (78)
 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
 ChanSel: Reserved
 Ext: 1 

[Asterisk-Users] Bridging ethernet over hdlc

[Christian Hoffmeyer]

I'm attempting to bridge ethernet over hdlc between two * boxes.  If anyone
has any information they can offer concerning this, it would be greatly
appreciated.

Here's the configuration the companies IT guy wants to bridge.  I have it
working already without a bridge, but he wants the head box's ethernet to be
on the same subnet as the ethernet across the span.

Pardon my horrible representation:

main network --eth([head box] .10.0)--- hdlc([span between
buildings])T1hdlc([span between buildings])--eth([head box]
.10.0)subnet

Google searches aren't turning up much for me.

Here are two resources Mark found, but I can't see the application and one
of them looks like it is not free.

http://sweb.cz/Frantisek.Rysanek/sync/dscc4+HDLC-Mini-HOWTO.html

http://www.etinc.com/index.php?page=etutils.htm  = not free

Thanks,
Christian

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Re: [Asterisk-Users] Record all phone calls

[Robert Mann]

See Below

- Original Message - 
From: Jimmy 
Riley 
To: '[EMAIL PROTECTED]' 

Sent: Saturday, January 10, 2004 10:01 AM
Subject: [Asterisk-Users] Record all phone calls


I want to record all phone calls 
made inbound and outbound. I'm new so having a hard time getting this started. 
Here is what I have so far but isn't working. Can someone help me out? 
Thanks,

[macro-record-on]exten = 
s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1})exten = 
s,2,Monitor(wav,${CALLFILENAME})

[sip]include = macro-record-oninclude = 
iaxtel exten = _,1,macro(record-on,${EXTEN},${CALLERIDNUM})exten 
= 1001,1,Dial(SIP/one|20|tr)exten = 1001,2,VoiceMail,u1001exten 
= 1001,102,VocieMail,b1001exten = 
2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001exten = 
1002,1,Dial(SIP/two|20|mtr)exten = 1002,2,VoiceMail,u1002exten = 
1002,102,VoiceMail,b1002exten = 6001,1,Ringingexten = 
6001,2,Wait(2)exten = 6001,3,VoicemailMain

There are afew issues I can see with this but your two big 
problems are as follows.

You never want to 
include a macro.
include = macro-record-on
So remove that line altogether.

You show exten = 
_,1,macro(record-on,${EXTEN},${CALLERIDNUM})the _ tells asterisk that you 
are going to want to match characters but then you dont tell it what you want to 
match.
so exten = _.,1etc... See the . after _ 
this tells * to match the rest of the characters (digits)

Those are your two big issues with not getting the 
recording to start.

Re: [Asterisk-Users] E100P - Error 500

[Bob Knight]

Daniel Bichara wrote:

Hi,

I am running * with E100P board. At least every our I got an Error 500 
message and ISDN-PRI restarts:

Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500
Jan 10 13:33:48 WARNING[98311]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI:
Read on 27 failed: Unknown error 500

Any clue? 
Unknown error 500 is an ELAST return code from zaptel driver.
It is telling libpri that there is an event in the queue.
If the read/write routines see that there is an event in the queue,
it just returns ELAST.  Libpri needs to do an ZT_GETEVENT to clear the event
and should do some error handling if needed.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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[Asterisk-Users] Mediatrix 1204

[Gonzalo Gasca Meza]

Someone have the MIB for MEdiatrix 1204 version 2.4.10.68?
thanks
--
Almada Tres SA de CV
Mitel Networks 
Eng. Gonzalo Gasca Meza
Service Engineer
52+(55)53730570


Mexico City, Mexico 


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[Asterisk-Users] Free Software or not -- that's the question /* New subject */

[asterisk]

(removed In-Reply-To header)

On Sat, Jan 10, 2004 at 10:01:12AM +, WipeOut wrote:
 
 And make sure to send in a  disclaimer otherwise it will not even be 
 looked at.. :)
 
 How do we know what is disclaimed or not disclaimed?
 /O
 
 Digium have all the Disclaimers and will not develop or include any code 
 into the CVS without one.. Thats all I was saying.. :)

And the disclaimers waive all of your rights to the code,
allow Digium to include it in their proprietary product,
and then they may or may not release it in the Asterisk
public CVS under the GPL.

Consider:

 A: Software licensed under the GPL is Free Software
 B: One of the freedoms relevant to Free Software is the
ability to make use of other Free Software in such
a way as to reduce duplication of effort.
 C: Digium will not include Free Software in the Asterisk
CVS.

So Digium releases Free Software while maintaining
strong centralized control of the project, to the point
of making dubious design decisions.

First of all, I applaud the recent decision to start 
making more formal releases of the software. This is a
big step forward.

Now, a case in point to illustrate C. Asterisk includes
a Berkeley DB implementation in its source tree. It lives
in the db1-ast subdirectory. Now every modern UNIX has a
Berkeley DB implementation included. These days it is 
usually DB3 from Sleepycat. Not the Sleepycat license under
which DB3 is released is basically the standard BSD license
with a bit of GPLish language added in. 

Though Digium supports Free Software to the point of releasing
code under the GPL, they are afraid enough of the idea
of Free Software, that they included an ancient (obsolete,
deprecated) implementation of a standard part of most operating
systems, in order to avoid GPL-like terms.

And why is this unnecessary cruft included in the source 
tree? So that Digium can leverage the Free Software
community into developing proprietary software for 
them.

Am I way off the mark?

-w
-- 
/~\  The ASCII Ribbon Campaign
\ /No HTML/RTF in email
 X No Word docs in email
/ \  Respect for open standards
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Re: [Asterisk-Users] Chagres Technologies, Inc

[John Brown (CV)]

For the list,

Mike received a partial order shipped 15-Dec, SN ending 4CD8.

Mike received email replies on 3-Dec  and 17-Dec advising him
on his order.

Mike ack'd those emails. 

This is the first time we have heard anything (phone calls or email)
from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
or calls.

Mike has been sent a private email and has been advised that
we will be issuing him a refund on product not received.  

I can only say that there is a human that answers the phones
at Chagres M-F 9-5 MDT (GMT-7).  

I think I'll change the Auto-Attendent so that it says 
For a Human press 0, instead of To reach an operator 
press 0.  Most people don't seem to press 0

for order status:  orders AT chagres dot net, 

or call  +1 505 830 1200 and please do leave good
information (name, phone number, what you ordered)
we don't always receive enough info to respond back
(missing phone numbers or complete names are common)

If you have any issue you can call my direct number at
+1 505 998 0567.  Thats my desk, ring it.

cheers,

john 

On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
 My experience has been one of unresponsiveness to my e-mails.  I have
 ordered and received devices from other providers in the time I have been
 waiting for Chagres.  As of now, based on my experiences and those of others
 that I have heard from I would highly recommend avoiding Chagres and Mr.
 Brown.  All I want now is a refund.
 
 Mike
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
 Johansson
 Sent: Saturday, January 10, 2004 3:22 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
 
 Mail John Brown at Chagres. [EMAIL PROTECTED]
 
 He usually responds quickly and I get information about where my products
 are.
 Yes, I also have rest orders, but I have acceptable responses on why and
 when
 they are expected to arrive in this snowy winterland...
 
 /O
 
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Re: [Asterisk-Users] Chagres Technologies, Inc

[John Brown (CV)]

Hi List,

Matt hasn't contacted us directly about this.  I've
responded to his previous statement that he hasn't 
recevied the last 20 units, and never heard back from
him.

Matt, again, if this is an issue please do contact us.
Our CDR and SMTP logs show no such attempt.

Our inventory records show 100 Grandstream Serial Numbers
have been shipped to you, along with tracking numbers.

+1 505 830 1200   Office Number, Auto Attendent answers 
  Pressing 0 takes you to a operator
  M-F 9-5 MDT (GMT-7)

orders at chagres dot net  gets email into the order admin
   which replies within 1 biz day
   and you should get a auto reply.

our email system now auto replys to help verify that your
email did reach us.  If you don't get an auto reply to 
the sales or order  role accounts then our SMTP box didn't
get your email.



On Sat, Jan 10, 2004 at 12:57:43AM -0500, mattf wrote:

 As I've said several times on this list[insert usual apology here], I still
 haven't received the last 20  of 100 phones I ordered over 2 months ago. If
 you get a hold of them please let me know
 
 MATT---
 
 
 -Original Message-
 From: mikeu [mailto:[EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 12:41 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Chagres Technologies, Inc
 
 
 
 Anyone else having problems getting product from Chagres?  They took my
 payment almost two months ago and I still have not seen hardware.  They have
 been horribly unresponsive to my e-mails.
 
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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[Brian Capouch]

[EMAIL PROTECTED] wrote:
And why is this unnecessary cruft included in the source 
tree? So that Digium can leverage the Free Software
community into developing proprietary software for 
them.

Am I way off the mark?

I think you're unfairly impugning Digium's motives.  And I also think 
you're--again--salting your post with enough innuendo that a reasonable 
person might suspect you of flame-baiting.

I suscribe to the mailing lists of several OS VoIP solutions, as I'm 
sure do many others on this list.  There is nothing out there like 
asterisk, in terms of it functionality, or the body of minds that have 
collected to work on it.  I have recently found myself embarking on a 
mini-career doing fundamental-level VoIP training to network operators, 
technology freaks, and even some small-telco tech people. I take along a 
laptop with asterisk on it and do a little song-and-dance that shows off 
some of its gee-whiz features.

It is not much of an exaggeration to say that almost always people's 
mouths drop open in amazement at what all that asterisk can do.  It's 
comical sometimes how affected people are.

So I have all this functionality, and I have all the source code to it, 
and I can legally keep it forever at this (mostly happy) level of 
functionality, and if Digium drops off the face of the earth, I can 
start with what's there (we can start with what's there; I know I 
won't be alone) and keep going should that happen.

So I can look at the same set of facts that you do, but in my mind 
Digium is not the nefarious would-be crook that you imply in your 
postings, but rather a brilliant and disruptive force upon the telco 
world.  And they are a *business,* and as many of the people reading 
this sentence are bound to know, one trick of the Open Source world is 
to figure out how to keep things open and free and at the same time how 
to keep bread on the table and enough cashflow to keep up with the 
technology (VoIP in this case) Joneses.

I cannot guess your motives, but I'm pretty sure that I *do* know what 
Digium's motives are, and they are innocuous and altruistic instead of 
the way you portray them.

Where are you trying to take this?

B.
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[Asterisk-Users] IAX v1 Changes

[Mark Spencer]

I plan on removing chan_iax from the normal build process, and making
chan_iax2 register itself as both IAX and IAX2.  IAX1 if built will
register itself as IAX1.  CVS asterisk has already been updated such
that IAX1 can be used to identify an IAX channel.

The removal of chan_iax from the normal build process (it will still be
possible to build by editing the Makefile) will be completed before 0.9.0.

Mark



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Re: [Asterisk-Users] Chagres Technologies, Inc

[admin]

Just refund the guy his money...
- Original Message - 
From:  John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:46 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


 For the list,

 Mike received a partial order shipped 15-Dec, SN ending 4CD8.

 Mike received email replies on 3-Dec  and 17-Dec advising him
 on his order.

 Mike ack'd those emails.

 This is the first time we have heard anything (phone calls or email)
 from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
 or calls.

 Mike has been sent a private email and has been advised that
 we will be issuing him a refund on product not received.

 I can only say that there is a human that answers the phones
 at Chagres M-F 9-5 MDT (GMT-7).

 I think I'll change the Auto-Attendent so that it says
 For a Human press 0, instead of To reach an operator
 press 0.  Most people don't seem to press 0

 for order status:  orders AT chagres dot net,

 or call  +1 505 830 1200 and please do leave good
 information (name, phone number, what you ordered)
 we don't always receive enough info to respond back
 (missing phone numbers or complete names are common)

 If you have any issue you can call my direct number at
 +1 505 998 0567.  Thats my desk, ring it.

 cheers,

 john

 On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
  My experience has been one of unresponsiveness to my e-mails.  I have
  ordered and received devices from other providers in the time I have
been
  waiting for Chagres.  As of now, based on my experiences and those of
others
  that I have heard from I would highly recommend avoiding Chagres and Mr.
  Brown.  All I want now is a refund.
 
  Mike
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
  Johansson
  Sent: Saturday, January 10, 2004 3:22 AM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
 
  Mail John Brown at Chagres. [EMAIL PROTECTED]
 
  He usually responds quickly and I get information about where my
products
  are.
  Yes, I also have rest orders, but I have acceptable responses on why and
  when
  they are expected to arrive in this snowy winterland...
 
  /O
 
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RE: [Asterisk-Users] Record all phone calls

[Jimmy Riley]

Here is what I have now. Where should the line  exten =
_.,1,macro(record-on,${EXTEN},${CALLERIDNUM}) go should it be under [sip]?
Right now if I call sip to sip monitoring starts and the calls connect but I
only get 44 byte files. If I call and iaxtel number monitoring starts but
call never gets placed and again 44byte files with nothing in them.
Thanks for the help.

[iaxtel]

exten =
_1700XXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1888NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1877NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1866NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1800NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])


[sip]
include = iaxtel
exten = _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})
exten = s,1,Dial(SIP/one|20|tr)
exten = 1001,1,Dial(SIP/one|20|tr)
exten = 1001,2,VoiceMail,u1001
exten = 1001,102,VocieMail,b1001
exten = 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001
exten = 1002,1,Dial(SIP/two|20|mtr)
exten = 1002,2,VoiceMail,u1002
exten = 1002,102,VoiceMail,b1002
exten = 6001,1,Ringing
exten = 6001,2,Wait(2)
exten = 6001,3,VoicemailMain



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Mann
Sent: January 10, 2004 12:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Record all phone calls

See Below
 
- Original Message - 
From: Jimmy Riley 
To: '[EMAIL PROTECTED]' 
Sent: Saturday, January 10, 2004 10:01 AM
Subject: [Asterisk-Users] Record all phone calls

I want to record all phone calls made inbound and outbound. I'm new so
having a hard time getting this started. Here is what I have so far but
isn't working. Can someone help me out? Thanks,

[macro-record-on]
exten = s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1})
exten = s,2,Monitor(wav,${CALLFILENAME})

[sip]
include = macro-record-on
include = iaxtel 
exten = _,1,macro(record-on,${EXTEN},${CALLERIDNUM})
exten = 1001,1,Dial(SIP/one|20|tr)
exten = 1001,2,VoiceMail,u1001
exten = 1001,102,VocieMail,b1001
exten = 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001
exten = 1002,1,Dial(SIP/two|20|mtr)
exten = 1002,2,VoiceMail,u1002
exten = 1002,102,VoiceMail,b1002
exten = 6001,1,Ringing
exten = 6001,2,Wait(2)
exten = 6001,3,VoicemailMain
 
There are a few issues I can see with this but your two big problems are as
follows.
 
You never want to include a macro.
include = macro-record-on
So remove that line altogether.
 
You show exten = _,1,macro(record-on,${EXTEN},${CALLERIDNUM})
the _ tells asterisk that you are going to want to match characters but then
you dont tell it what you want to match.
so exten = _.,1etc...  See the . after _ this tells * to match the rest of
the characters (digits)
 
Those are your two big issues with not getting the recording to start.

 
 

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Re: [Asterisk-Users] Chagres Technologies, Inc

[admin]

I would feel sympathetic to Chagres Technologies but I have read many many
posts to the same effect.  If you are going to take someone's money then
follow through on your service or product in a timely manner.  If you
cannot, close your business and stop taking people's money.


- Original Message - 
From:  John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:56 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


 Hi List,

 Matt hasn't contacted us directly about this.  I've
 responded to his previous statement that he hasn't
 recevied the last 20 units, and never heard back from
 him.

 Matt, again, if this is an issue please do contact us.
 Our CDR and SMTP logs show no such attempt.

 Our inventory records show 100 Grandstream Serial Numbers
 have been shipped to you, along with tracking numbers.

 +1 505 830 1200   Office Number, Auto Attendent answers
   Pressing 0 takes you to a operator
   M-F 9-5 MDT (GMT-7)

 orders at chagres dot net  gets email into the order admin
which replies within 1 biz day
and you should get a auto reply.

 our email system now auto replys to help verify that your
 email did reach us.  If you don't get an auto reply to
 the sales or order  role accounts then our SMTP box didn't
 get your email.



 On Sat, Jan 10, 2004 at 12:57:43AM -0500, mattf wrote:

  As I've said several times on this list[insert usual apology here], I
still
  haven't received the last 20  of 100 phones I ordered over 2 months ago.
If
  you get a hold of them please let me know
 
  MATT---
 
 
  -Original Message-
  From: mikeu [mailto:[EMAIL PROTECTED]
  Sent: Saturday, January 10, 2004 12:41 AM
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Chagres Technologies, Inc
 
 
 
  Anyone else having problems getting product from Chagres?  They took my
  payment almost two months ago and I still have not seen hardware.  They
have
  been horribly unresponsive to my e-mails.
 
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Re: [Asterisk-Users] Chagres Technologies, Inc

[admin]

Sorry, but how can you ID his inbound packets?


- Original Message - 
From: admin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 3:17 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


 Just refund the guy his money...
 - Original Message - 
 From:  John Brown (CV) [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 2:46 PM
 Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


  For the list,
 
  Mike received a partial order shipped 15-Dec, SN ending 4CD8.
 
  Mike received email replies on 3-Dec  and 17-Dec advising him
  on his order.
 
  Mike ack'd those emails.
 
  This is the first time we have heard anything (phone calls or email)
  from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
  or calls.
 
  Mike has been sent a private email and has been advised that
  we will be issuing him a refund on product not received.
 
  I can only say that there is a human that answers the phones
  at Chagres M-F 9-5 MDT (GMT-7).
 
  I think I'll change the Auto-Attendent so that it says
  For a Human press 0, instead of To reach an operator
  press 0.  Most people don't seem to press 0
 
  for order status:  orders AT chagres dot net,
 
  or call  +1 505 830 1200 and please do leave good
  information (name, phone number, what you ordered)
  we don't always receive enough info to respond back
  (missing phone numbers or complete names are common)
 
  If you have any issue you can call my direct number at
  +1 505 998 0567.  Thats my desk, ring it.
 
  cheers,
 
  john
 
  On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
   My experience has been one of unresponsiveness to my e-mails.  I have
   ordered and received devices from other providers in the time I have
 been
   waiting for Chagres.  As of now, based on my experiences and those of
 others
   that I have heard from I would highly recommend avoiding Chagres and
Mr.
   Brown.  All I want now is a refund.
  
   Mike
  
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
   Johansson
   Sent: Saturday, January 10, 2004 3:22 AM
   To: [EMAIL PROTECTED]
   Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
  
   Mail John Brown at Chagres. [EMAIL PROTECTED]
  
   He usually responds quickly and I get information about where my
 products
   are.
   Yes, I also have rest orders, but I have acceptable responses on why
and
   when
   they are expected to arrive in this snowy winterland...
  
   /O
  
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[Asterisk-Users] ADSI Configs

[Lee Redmayne]

Hi All
 
If I want to get my ADSI Phones (successfully connected off a Rhino Channel
Bank and TE410P) to connect to Asterisk to get their config downloaded, is
there something specific needed in extensions.conf for them to dial to get
this?

Thanks :)

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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[admin]

I work for an interconnect that sells 3com and NEC.  When I made this
project my own and followed through to show my boss, he said, this is going
to ruin our industry

If that is the case then so be it.  Same with mp3s and the music industry.
Had they embraced the technology, everyone could be making a living.  Now
they have to sue as a last fight on the way out.

Really, this is like a car that doesnt run on gas.
- Original Message - 
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 3:03 PM
Subject: Re: [Asterisk-Users] Free Software or not -- that's the question /*
New subject */


 [EMAIL PROTECTED] wrote:
 
  And why is this unnecessary cruft included in the source
  tree? So that Digium can leverage the Free Software
  community into developing proprietary software for
  them.
 
  Am I way off the mark?
 

 I think you're unfairly impugning Digium's motives.  And I also think
 you're--again--salting your post with enough innuendo that a reasonable
 person might suspect you of flame-baiting.

 I suscribe to the mailing lists of several OS VoIP solutions, as I'm
 sure do many others on this list.  There is nothing out there like
 asterisk, in terms of it functionality, or the body of minds that have
 collected to work on it.  I have recently found myself embarking on a
 mini-career doing fundamental-level VoIP training to network operators,
 technology freaks, and even some small-telco tech people. I take along a
 laptop with asterisk on it and do a little song-and-dance that shows off
 some of its gee-whiz features.

 It is not much of an exaggeration to say that almost always people's
 mouths drop open in amazement at what all that asterisk can do.  It's
 comical sometimes how affected people are.

 So I have all this functionality, and I have all the source code to it,
 and I can legally keep it forever at this (mostly happy) level of
 functionality, and if Digium drops off the face of the earth, I can
 start with what's there (we can start with what's there; I know I
 won't be alone) and keep going should that happen.

 So I can look at the same set of facts that you do, but in my mind
 Digium is not the nefarious would-be crook that you imply in your
 postings, but rather a brilliant and disruptive force upon the telco
 world.  And they are a *business,* and as many of the people reading
 this sentence are bound to know, one trick of the Open Source world is
 to figure out how to keep things open and free and at the same time how
 to keep bread on the table and enough cashflow to keep up with the
 technology (VoIP in this case) Joneses.

 I cannot guess your motives, but I'm pretty sure that I *do* know what
 Digium's motives are, and they are innocuous and altruistic instead of
 the way you portray them.

 Where are you trying to take this?

 B.
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RE: [Asterisk-Users] Chagres Technologies, Inc

[Sean Cheesman]

just drop it!  it is for them to iron out!  and for the record, I received my order 
within a week of placing the order.

-Original Message- 
From: admin [mailto:[EMAIL PROTECTED] 
Sent: Sat 1/10/2004 3:23 PM 
To: [EMAIL PROTECTED] 
Cc: 
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc



Sorry, but how can you ID his inbound packets?


- Original Message -
From: admin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 3:17 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


 Just refund the guy his money...
 - Original Message -
 From:  John Brown (CV) [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 2:46 PM
 Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


  For the list,
 
  Mike received a partial order shipped 15-Dec, SN ending 4CD8.
 
  Mike received email replies on 3-Dec  and 17-Dec advising him
  on his order.
 
  Mike ack'd those emails.
 
  This is the first time we have heard anything (phone calls or email)
  from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
  or calls.
 
  Mike has been sent a private email and has been advised that
  we will be issuing him a refund on product not received.
 
  I can only say that there is a human that answers the phones
  at Chagres M-F 9-5 MDT (GMT-7).
 
  I think I'll change the Auto-Attendent so that it says
  For a Human press 0, instead of To reach an operator
  press 0.  Most people don't seem to press 0
 
  for order status:  orders AT chagres dot net,
 
  or call  +1 505 830 1200 and please do leave good
  information (name, phone number, what you ordered)
  we don't always receive enough info to respond back
  (missing phone numbers or complete names are common)
 
  If you have any issue you can call my direct number at
  +1 505 998 0567.  Thats my desk, ring it.
 
  cheers,
 
  john
 
  On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
   My experience has been one of unresponsiveness to my e-mails.  I have
   ordered and received devices from other providers in the time I have
 been
   waiting for Chagres.  As of now, based on my experiences and those of
 others
   that I have heard from I would highly recommend avoiding Chagres and
Mr.
   Brown.  All I want now is a refund.
  
   Mike
  
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
   Johansson
   Sent: Saturday, January 10, 2004 3:22 AM
   To: [EMAIL PROTECTED]
   Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
  
   Mail John Brown at Chagres. [EMAIL PROTECTED]
  
   He usually responds quickly and I get information about where my
 products
   are.
   Yes, I also have rest orders, but I have acceptable responses on why
and
   when
   they are expected to arrive in this snowy winterland...
  
   /O
  
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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[Brian West]

w,
You also have to consider that if Asterisk used any GPL code we
would loose the ability to use/link to openh323, provide g729 of any sort.
We would also Dialogic support.  Now do you want to be the one to tell
everyong that depends on h323, g729 or Dialogic cards they are just SOL?

Asterisk is GPL and the way digium does their disclaimers doesn't make
Asterisk any less of a GPL project.

I require h323 and g729 support and use it daily.

bkw

On Sat, 10 Jan 2004 [EMAIL PROTECTED] wrote:

 (removed In-Reply-To header)

 On Sat, Jan 10, 2004 at 10:01:12AM +, WipeOut wrote:
  
  And make sure to send in a  disclaimer otherwise it will not even be
  looked at.. :)
  
  How do we know what is disclaimed or not disclaimed?
  /O
  
  Digium have all the Disclaimers and will not develop or include any code
  into the CVS without one.. Thats all I was saying.. :)

 And the disclaimers waive all of your rights to the code,
 allow Digium to include it in their proprietary product,
 and then they may or may not release it in the Asterisk
 public CVS under the GPL.

 Consider:

  A: Software licensed under the GPL is Free Software
  B: One of the freedoms relevant to Free Software is the
 ability to make use of other Free Software in such
 a way as to reduce duplication of effort.
  C: Digium will not include Free Software in the Asterisk
 CVS.

 So Digium releases Free Software while maintaining
 strong centralized control of the project, to the point
 of making dubious design decisions.

 First of all, I applaud the recent decision to start
 making more formal releases of the software. This is a
 big step forward.

 Now, a case in point to illustrate C. Asterisk includes
 a Berkeley DB implementation in its source tree. It lives
 in the db1-ast subdirectory. Now every modern UNIX has a
 Berkeley DB implementation included. These days it is
 usually DB3 from Sleepycat. Not the Sleepycat license under
 which DB3 is released is basically the standard BSD license
 with a bit of GPLish language added in.

 Though Digium supports Free Software to the point of releasing
 code under the GPL, they are afraid enough of the idea
 of Free Software, that they included an ancient (obsolete,
 deprecated) implementation of a standard part of most operating
 systems, in order to avoid GPL-like terms.

 And why is this unnecessary cruft included in the source
 tree? So that Digium can leverage the Free Software
 community into developing proprietary software for
 them.

 Am I way off the mark?

 -w
 --
 /~\  The ASCII Ribbon Campaign
 \ /No HTML/RTF in email
  X No Word docs in email
 / \  Respect for open standards
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[Asterisk-Users] far end disconnect supervision

[Lance Arbuckle]

I'm starting to shop for my first channel bank and one of the features
that eveyone seems to recommend is far end disconnect supervision.
What other terms do various manufactures use to describe this same
feature ?

Is calling party disconnect the same as far end disconnect
supervision ?

Thanks

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Re: [Asterisk-Users] Chagres Technologies, Inc

[John Brown (CV)]

Hi,

You have read a small sample of posts.  With over 300 customers
there are maybe 25 to 30 in Nov that had issues.  Thats less
than 10 percent.   And for Dec, our order lead times have
gotten back on track.  The average turn around time for an
order in Jan 2004 is 2.1 business days.  Compare that to over
18 days for Nov when we had internal issues.  Our goal is
1.0 business day for order shipment.  These are averages
and certainly some people had to wait longer than 18 days
for an order.

Cheers

On Sat, Jan 10, 2004 at 03:21:26PM -0500, admin wrote:
 I would feel sympathetic to Chagres Technologies but I have read many many
 posts to the same effect.  If you are going to take someone's money then
 follow through on your service or product in a timely manner.  If you
 cannot, close your business and stop taking people's money.
 
 
 - Original Message - 
 From:  John Brown (CV) [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 2:56 PM
 Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
 
 
  Hi List,
 
  Matt hasn't contacted us directly about this.  I've
  responded to his previous statement that he hasn't
  recevied the last 20 units, and never heard back from
  him.
 
  Matt, again, if this is an issue please do contact us.
  Our CDR and SMTP logs show no such attempt.
 
  Our inventory records show 100 Grandstream Serial Numbers
  have been shipped to you, along with tracking numbers.
 
  +1 505 830 1200   Office Number, Auto Attendent answers
Pressing 0 takes you to a operator
M-F 9-5 MDT (GMT-7)
 
  orders at chagres dot net  gets email into the order admin
 which replies within 1 biz day
 and you should get a auto reply.
 
  our email system now auto replys to help verify that your
  email did reach us.  If you don't get an auto reply to
  the sales or order  role accounts then our SMTP box didn't
  get your email.
 
 
 
  On Sat, Jan 10, 2004 at 12:57:43AM -0500, mattf wrote:
 
   As I've said several times on this list[insert usual apology here], I
 still
   haven't received the last 20  of 100 phones I ordered over 2 months ago.
 If
   you get a hold of them please let me know
  
   MATT---
  
  
   -Original Message-
   From: mikeu [mailto:[EMAIL PROTECTED]
   Sent: Saturday, January 10, 2004 12:41 AM
   To: [EMAIL PROTECTED]
   Subject: [Asterisk-Users] Chagres Technologies, Inc
  
  
  
   Anyone else having problems getting product from Chagres?  They took my
   payment almost two months ago and I still have not seen hardware.  They
 have
   been horribly unresponsive to my e-mails.
  
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Re: [Asterisk-Users] Chagres Technologies, Inc

[John Brown (CV)]

its simple,
i can lookup the MX for his zone, then look up the
A RR for each MX, and then search the logs for IP's
or I can even expand the search to look for CIDR prefixes.

I can also lookup in my private RBL, any query my SMTP
machine would have made to see if his IP(s) are spam
sources or not.

If I don't see packets from those IP(s), or from
his MX's, or from his domain, then I'm going to assume
no packets where received.

cheers


On Sat, Jan 10, 2004 at 03:23:13PM -0500, admin wrote:
 Sorry, but how can you ID his inbound packets?
 
 
 - Original Message - 
 From: admin [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 3:17 PM
 Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
 
 
  Just refund the guy his money...
  - Original Message - 
  From:  John Brown (CV) [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Saturday, January 10, 2004 2:46 PM
  Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
 
 
   For the list,
  
   Mike received a partial order shipped 15-Dec, SN ending 4CD8.
  
   Mike received email replies on 3-Dec  and 17-Dec advising him
   on his order.
  
   Mike ack'd those emails.
  
   This is the first time we have heard anything (phone calls or email)
   from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
   or calls.
  
   Mike has been sent a private email and has been advised that
   we will be issuing him a refund on product not received.
  
   I can only say that there is a human that answers the phones
   at Chagres M-F 9-5 MDT (GMT-7).
  
   I think I'll change the Auto-Attendent so that it says
   For a Human press 0, instead of To reach an operator
   press 0.  Most people don't seem to press 0
  
   for order status:  orders AT chagres dot net,
  
   or call  +1 505 830 1200 and please do leave good
   information (name, phone number, what you ordered)
   we don't always receive enough info to respond back
   (missing phone numbers or complete names are common)
  
   If you have any issue you can call my direct number at
   +1 505 998 0567.  Thats my desk, ring it.
  
   cheers,
  
   john
  
   On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
My experience has been one of unresponsiveness to my e-mails.  I have
ordered and received devices from other providers in the time I have
  been
waiting for Chagres.  As of now, based on my experiences and those of
  others
that I have heard from I would highly recommend avoiding Chagres and
 Mr.
Brown.  All I want now is a refund.
   
Mike
   
   
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Saturday, January 10, 2004 3:22 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
   
Mail John Brown at Chagres. [EMAIL PROTECTED]
   
He usually responds quickly and I get information about where my
  products
are.
Yes, I also have rest orders, but I have acceptable responses on why
 and
when
they are expected to arrive in this snowy winterland...
   
/O
   
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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[info-lists]

admin said:
 I work for an interconnect that sells 3com and NEC.  When I made this
 project my own and followed through to show my boss, he said, this is
 going
 to ruin our industry

 If that is the case then so be it.  Same with mp3s and the music industry.
 Had they embraced the technology, everyone could be making a living.  Now
 they have to sue as a last fight on the way out.

 Really, this is like a car that doesnt run on gas.


Seems like it isn't going to ruin your industry but will put a dent in
3Com and NEC !!  In fact it could improve your company's business model
since you  sell services to setup and configure Asterisk.
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[Asterisk-Users] drop calls with T100P / PRI

[John Brown (CV)]

Hi List,

a number of our customers are reporting dropped calls.

here is the config.

1   T100P  T1 Card
1   Asterisk  (Mid Nov build)

T1 is signalled as a PRI(National)

The card will only sync up if we clock, if
we line side clock the card goes into yellow alarm
and won't sync up.

the only errors we see are framing slips.

Around 2500 slips over a 18 hour period.
(this was reported from our T-Berd)


Any thoughts ???

--

Here is our /etc/zaptel.conf

span=1,0,0,esf,b8zs
bchan=1-4
dchan=24
fxols=25


--

here is  .../asterisk/zapata.conf

[channels]
context=espire-pri-in

switchtype=national
pridialplan=national

usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
callgroup=2
pickupgroup=2

immediate=no
busydetect=yes
callprogress=yes
musiconhold=default
signalling=pri_cpe
group=1
channel= 1-4

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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[Olle E. Johansson]

It's very hard to find a business model for working with Open Source Software in a 
for-profit
software company.
Mysql and Digium are success stories that work with a two-fold model that seems to 
work. Do not forget
that there are companies out there that wants to buy the software with a more 
business-minded license
than GPL and can't use the software with GPL license. As long as Digium continues to 
enhance
and give away code, I see no problem with letting them use my code. And as Tilghman 
will
point out if I don't do it, I can still have the copyright to my code, just let them 
use it
in their business. It's a form of coop-operation ;-)
If Digium seriously misbehave and start releasing lots of functionality on the side 
and not giving
new releases to the community, well then we might have to consider where the community 
want to go.
We're far away from that situation.
I think it's time to calm down and move forward, use the time to make sure we can 
relase
a stable 1.0 soon.
To do that, we need help debugging and testing all the patches in bugs.digium.com. Bug 
marshals
are working with the process, but we need many more people testing and reporting their 
findings,
good or bad, in bugs.digium.com
Thank you for helping us with this.

/O

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Re: [Asterisk-Users] Chagres Technologies, Inc

[Rich Adamson]

common on john, stop the bs. we all know email can be sent
from hundreds of different valid accounts that you can't
trace that way (yahoo and msn as just two), and those of
us that have been involved with security understand it
rather well.


 its simple,
 i can lookup the MX for his zone, then look up the
 A RR for each MX, and then search the logs for IP's
 or I can even expand the search to look for CIDR prefixes.
 
 I can also lookup in my private RBL, any query my SMTP
 machine would have made to see if his IP(s) are spam
 sources or not.
 
 If I don't see packets from those IP(s), or from
 his MX's, or from his domain, then I'm going to assume
 no packets where received.
 
 cheers
 
 
 On Sat, Jan 10, 2004 at 03:23:13PM -0500, admin wrote:
  Sorry, but how can you ID his inbound packets?
  
  
  - Original Message - 
  From: admin [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Saturday, January 10, 2004 3:17 PM
  Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
  
  
   Just refund the guy his money...
   - Original Message - 
   From:  John Brown (CV) [EMAIL PROTECTED]
   To: [EMAIL PROTECTED]
   Sent: Saturday, January 10, 2004 2:46 PM
   Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
  
  
For the list,
   
Mike received a partial order shipped 15-Dec, SN ending 4CD8.
   
Mike received email replies on 3-Dec  and 17-Dec advising him
on his order.
   
Mike ack'd those emails.
   
This is the first time we have heard anything (phone calls or email)
from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
or calls.
   
Mike has been sent a private email and has been advised that
we will be issuing him a refund on product not received.
   
I can only say that there is a human that answers the phones
at Chagres M-F 9-5 MDT (GMT-7).
   
I think I'll change the Auto-Attendent so that it says
For a Human press 0, instead of To reach an operator
press 0.  Most people don't seem to press 0
   
for order status:  orders AT chagres dot net,
   
or call  +1 505 830 1200 and please do leave good
information (name, phone number, what you ordered)
we don't always receive enough info to respond back
(missing phone numbers or complete names are common)
   
If you have any issue you can call my direct number at
+1 505 998 0567.  Thats my desk, ring it.
   
cheers,
   
john
   
On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
 My experience has been one of unresponsiveness to my e-mails.  I have
 ordered and received devices from other providers in the time I have
   been
 waiting for Chagres.  As of now, based on my experiences and those of
   others
 that I have heard from I would highly recommend avoiding Chagres and
  Mr.
 Brown.  All I want now is a refund.

 Mike


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
 Johansson
 Sent: Saturday, January 10, 2004 3:22 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Chagres Technologies, Inc

 Mail John Brown at Chagres. [EMAIL PROTECTED]

 He usually responds quickly and I get information about where my
   products
 are.
 Yes, I also have rest orders, but I have acceptable responses on why
  and
 when
 they are expected to arrive in this snowy winterland...

 /O

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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[Olle E. Johansson]

Brian West wrote:
w,
You also have to consider that if Asterisk used any GPL code we
would loose the ability to use/link to openh323, provide g729 of any sort.
We would also Dialogic support.  Now do you want to be the one to tell
everyong that depends on h323, g729 or Dialogic cards they are just SOL?
And, with clever interfaces, we can still interface to GPL code even though
we can't include it in the base CVS. The Mysql-addon is one example,
the festival interface may be another and Brians solution with ODBC to connect
to every other database, GPL or not, is another solution.
Let's go back to work :-)
/O
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[Asterisk-Users] My first E1 card is running :)

[Anton Tinchev]

Just happy.

hardware information:
--
Some small factor IBM
Celetron (coppermine) at 1100 (11*100FSB)
256 RAM
15GB Hard.
1 x Digium E100P - E1 Line from telco with 300 Dids

1 x TDM400P for local phones
---
Few small machines (mainly brand PII at 233Mhz with TDM400P Cards.
---
There is a lot of SIP equipment attached:
2 x Micronet SIP Gateways
1 x ata186
1 x AudioCodes 1004
2 x Cisco AS5350 Gateways ( now it seems Obsolete :) )
Not a sign of echo problem - is this becouse all my analog phones are 
connected with cat5e cables?

This is heavy production enviroment - Sofia's Metropolian Area Network 
Operation centre.
Now i'm playing with the ADSI scripts. If someone has cool ADSI scripts, 
please send me.

P.S.
Should we arrange different Successfull Stories Mailing list?
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Re: [Asterisk-Users] Chagres Technologies, Inc

[John Brown (CV)]

rick, i didn't say that they couldn't have sent email
from another location.  certainly yahoo and msn are
harder to deal with.

Yes, rick you can do some tracing the way I mentioned.

lets see:dig routers.com mx
routers.com.4H IN MX10 texas.routers.com.

;; AUTHORITY SECTION:
routers.com.4H IN NSdns.inetnebr.com.
routers.com.4H IN NStexas.routers.com.

;; ADDITIONAL SECTION:
texas.routers.com.  2D IN A 206.222.193.73
dns.inetnebr.com.   2D IN A 199.184.119.1


hmm, so i would expect to see email from  texas.routers.com
or from some device within  206.222.193.xxx at a min.
(which would cover your machine called vegas.)

I would expect that a  grep -i routers.com  mailbox
would produce output that showed mail from that domain.

I would expect that  a  grep 192.222.206  db.rbl.ct
would either show something in that block as being a 
locatlly flagged spam source, or show nothing, which means
we didn't block it

i would expect that a grep  206.222.192  /var/log/security
(freebsd ipfw logs)  would show something since we have a
rule called   permit log tcp from any to mailserver 25 

I would expect that a grep  routers.com  /var/log/maillog
which logs the smtp sessions to show something.

If I didnt' get a hit on any of those, I think its pretty 
safe to say I didn't get the email, something is broken.

All I'm saying is that based on the information we have
for that customer, I can and do check our logs to see if
something got dropped.


Almost half of the customers that had issues have their
IP's listed in multiple different RBL's  So instead of
dropping those emails, now we have to put them into 
a seperate folder and manually check them.  we get close
to 2800 spam messages a day into those folders.  

If they sent from hotmail or yahoo, then about the only
thing I can do is grep for there email addy string.

and yes rich, i'm involved with security issues as
well and have a clear understanding of how packets 
move, and what tools I have on my network that allow
me to see whats happening.

cheers


On Sat, Jan 10, 2004 at 03:37:15PM -0600, Rich Adamson wrote:
 common on john, stop the bs. we all know email can be sent
 from hundreds of different valid accounts that you can't
 trace that way (yahoo and msn as just two), and those of
 us that have been involved with security understand it
 rather well.
 
 
  its simple,
  i can lookup the MX for his zone, then look up the
  A RR for each MX, and then search the logs for IP's
  or I can even expand the search to look for CIDR prefixes.
  
  I can also lookup in my private RBL, any query my SMTP
  machine would have made to see if his IP(s) are spam
  sources or not.
  
  If I don't see packets from those IP(s), or from
  his MX's, or from his domain, then I'm going to assume
  no packets where received.
  
  cheers
  
  
  On Sat, Jan 10, 2004 at 03:23:13PM -0500, admin wrote:
   Sorry, but how can you ID his inbound packets?
   
   
   - Original Message - 
   From: admin [EMAIL PROTECTED]
   To: [EMAIL PROTECTED]
   Sent: Saturday, January 10, 2004 3:17 PM
   Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
   
   
Just refund the guy his money...
- Original Message - 
From:  John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:46 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
   
   
 For the list,

 Mike received a partial order shipped 15-Dec, SN ending 4CD8.

 Mike received email replies on 3-Dec  and 17-Dec advising him
 on his order.

 Mike ack'd those emails.

 This is the first time we have heard anything (phone calls or email)
 from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound packets
 or calls.

 Mike has been sent a private email and has been advised that
 we will be issuing him a refund on product not received.

 I can only say that there is a human that answers the phones
 at Chagres M-F 9-5 MDT (GMT-7).

 I think I'll change the Auto-Attendent so that it says
 For a Human press 0, instead of To reach an operator
 press 0.  Most people don't seem to press 0

 for order status:  orders AT chagres dot net,

 or call  +1 505 830 1200 and please do leave good
 information (name, phone number, what you ordered)
 we don't always receive enough info to respond back
 (missing phone numbers or complete names are common)

 If you have any issue you can call my direct number at
 +1 505 998 0567.  Thats my desk, ring it.

 cheers,

 john

 On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote:
  My experience has been one of unresponsiveness to my e-mails.  I have
  ordered and received devices from other providers in the time I have
been
  waiting for Chagres.  As of 

Re: [Asterisk-Users] far end disconnect supervision

[Rich Adamson]

 I'm starting to shop for my first channel bank and one of the features
 that eveyone seems to recommend is far end disconnect supervision.
 What other terms do various manufactures use to describe this same
 feature ?
 
 Is calling party disconnect the same as far end disconnect
 supervision ?

Yes, in most readers terms. However, in some cases marketing/sales 
people may have written stuff with no clue what they are talking
about.


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RE: [Asterisk-Users] drop calls with T100P / PRI

[Scott Stingel]

Maybe you could look in /var/log/asterisk/messages and see if there are any
errors that correspond to the times of dropped calls? If so, what kinds of
errors do you see there?

As far as the problems you report receiving emails from your customers,
maybe your provider is spam-filtering your mail, and accidentally deleting
mail from your customers?

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  scott at evtmedia.com   
URL:www.evtmedia.com 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV)
Sent: Saturday, January 10, 2004 9:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] drop calls with T100P / PRI



Hi List,

a number of our customers are reporting dropped calls.

here is the config.

1   T100P  T1 Card
1   Asterisk  (Mid Nov build)

T1 is signalled as a PRI(National)

The card will only sync up if we clock, if
we line side clock the card goes into yellow alarm
and won't sync up.

the only errors we see are framing slips.

Around 2500 slips over a 18 hour period.
(this was reported from our T-Berd)


Any thoughts ???

--

Here is our /etc/zaptel.conf

span=1,0,0,esf,b8zs
bchan=1-4
dchan=24
fxols=25


--

here is  .../asterisk/zapata.conf

[channels]
context=espire-pri-in

switchtype=national
pridialplan=national

usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
callgroup=2
pickupgroup=2

immediate=no
busydetect=yes
callprogress=yes
musiconhold=default
signalling=pri_cpe
group=1
channel= 1-4

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Re: [Asterisk-Users] Cisco Gear

[Nicolas Bougues]

Hi,

Would you mind giving me an idea of the price level for the 7970,
7960, 7940 and 7920 ?

Qty 5 or more. Shipping to France.

On Fri, Jan 09, 2004 at 06:00:29PM +, Adthrawn wrote:
 Hi,
 
 I know it's not really the place, but if anybody in the UK (or US) is 
 interested, I'm clearing out lots of new Cisco stock...
 
 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), 
 7935's (conference phone) and 3550-24-PWR switches.
 
 I also have boxes of 7914's, the single-7914 foot stand and double-7914 
 foot stand (these are required to connect a 7914 to a 7960G).
 
 And some useful locking and non-locking wallmount brackets for 79xx 
 range.
 
 We also have lots of PSU's for the whole 79xx range.
 
 I'll now feel ashamed, and sink into my seat :-)
 
 Best,
 Ad.
 
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-- 
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] Cisco Gear

[Nicolas Bougues]

Whoops...

echo set ignore_list_reply_to = yes .muttrc

Sorry. I believe that the Reply-To setting on this list must have
been discussed here a few times here, so I won't start :)

On Fri, Jan 09, 2004 at 06:00:29PM +, Adthrawn wrote:
 Hi,
 
 I know it's not really the place, but if anybody in the UK (or US) is 
 interested, I'm clearing out lots of new Cisco stock...
 
 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), 
 7935's (conference phone) and 3550-24-PWR switches.
 
 I also have boxes of 7914's, the single-7914 foot stand and double-7914 
 foot stand (these are required to connect a 7914 to a 7960G).
 
 And some useful locking and non-locking wallmount brackets for 79xx 
 range.
 
 We also have lots of PSU's for the whole 79xx range.
 
 I'll now feel ashamed, and sink into my seat :-)
 
 Best,
 Ad.
 
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-- 
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] Chagres Technologies, Inc

[info-lists]

John,
Take your discussion off list... It is way off topic. I think you
do yourself more harm than good by responding to these issues on list.
If you want to build confidence in your company then ask your satisfied
customers to reccommend you and give their testimonials regarding your
speedy service and support.  BUT don't get into these arguments.

Respectifully
Robert


John Brown (CV) said:
 rick, i didn't say that they couldn't have sent email
 from another location.  certainly yahoo and msn are
 harder to deal with.

 Yes, rick you can do some tracing the way I mentioned.

 lets see:dig routers.com mx
 routers.com.4H IN MX10 texas.routers.com.

 ;; AUTHORITY SECTION:
 routers.com.4H IN NSdns.inetnebr.com.
 routers.com.4H IN NStexas.routers.com.

 ;; ADDITIONAL SECTION:
 texas.routers.com.  2D IN A 206.222.193.73
 dns.inetnebr.com.   2D IN A 199.184.119.1


 hmm, so i would expect to see email from  texas.routers.com
 or from some device within  206.222.193.xxx at a min.
 (which would cover your machine called vegas.)

 I would expect that a  grep -i routers.com  mailbox
 would produce output that showed mail from that domain.

 I would expect that  a  grep 192.222.206  db.rbl.ct
 would either show something in that block as being a
 locatlly flagged spam source, or show nothing, which means
 we didn't block it

 i would expect that a grep  206.222.192  /var/log/security
 (freebsd ipfw logs)  would show something since we have a
 rule called   permit log tcp from any to mailserver 25

 I would expect that a grep  routers.com  /var/log/maillog
 which logs the smtp sessions to show something.

 If I didnt' get a hit on any of those, I think its pretty
 safe to say I didn't get the email, something is broken.

 All I'm saying is that based on the information we have
 for that customer, I can and do check our logs to see if
 something got dropped.


 Almost half of the customers that had issues have their
 IP's listed in multiple different RBL's  So instead of
 dropping those emails, now we have to put them into
 a seperate folder and manually check them.  we get close
 to 2800 spam messages a day into those folders.

 If they sent from hotmail or yahoo, then about the only
 thing I can do is grep for there email addy string.

 and yes rich, i'm involved with security issues as
 well and have a clear understanding of how packets
 move, and what tools I have on my network that allow
 me to see whats happening.

 cheers


 On Sat, Jan 10, 2004 at 03:37:15PM -0600, Rich Adamson wrote:
 common on john, stop the bs. we all know email can be sent
 from hundreds of different valid accounts that you can't
 trace that way (yahoo and msn as just two), and those of
 us that have been involved with security understand it
 rather well.

 
  its simple,
  i can lookup the MX for his zone, then look up the
  A RR for each MX, and then search the logs for IP's
  or I can even expand the search to look for CIDR prefixes.
 
  I can also lookup in my private RBL, any query my SMTP
  machine would have made to see if his IP(s) are spam
  sources or not.
 
  If I don't see packets from those IP(s), or from
  his MX's, or from his domain, then I'm going to assume
  no packets where received.
 
  cheers
 
 
  On Sat, Jan 10, 2004 at 03:23:13PM -0500, admin wrote:
   Sorry, but how can you ID his inbound packets?
  
  
   - Original Message -
   From: admin [EMAIL PROTECTED]
   To: [EMAIL PROTECTED]
   Sent: Saturday, January 10, 2004 3:17 PM
   Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
  
  
Just refund the guy his money...
- Original Message -
From:  John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:46 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
   
   
 For the list,

 Mike received a partial order shipped 15-Dec, SN ending 4CD8.

 Mike received email replies on 3-Dec  and 17-Dec advising him
 on his order.

 Mike ack'd those emails.

 This is the first time we have heard anything (phone calls or
 email)
 from Mike since 17-Dec.  Our CDR and SMTP logs show no inbound
 packets
 or calls.

 Mike has been sent a private email and has been advised that
 we will be issuing him a refund on product not received.

 I can only say that there is a human that answers the phones
 at Chagres M-F 9-5 MDT (GMT-7).

 I think I'll change the Auto-Attendent so that it says
 For a Human press 0, instead of To reach an operator
 press 0.  Most people don't seem to press 0

 for order status:  orders AT chagres dot net,

 or call  +1 505 830 1200 and please do leave good
 information (name, phone number, what you ordered)
 we don't always receive enough info to respond back
 (missing phone numbers or 

[Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Owen Kelso]

I'm using Asterisk on a open server (no firewall or NAT) and trying to
communicate with a Grandstream BudgeTone 102 SIP phone which is behind
NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS
about a week ago.  My problem is that I'm only getting half-duplex
communication -- I can hear voice from the Asterisk server but the server
does not understand any voice from me.  From the console sip debug shows
that the SIP part is working fine and DTMF via SIP INFO works.

I've struggled with this for a few days now and can't figure out the
cause.  The only symptoms I've found are:

(1) When I make a call the console spits out the following errors several
times per minute:
WARNING[-1220854864]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable

(2) An ethereal trace reveals that incoming RTP packets have failed UDP
checksums (all packets have the same checksum of 0xb38f).  I don't see
anything else irregular, like unreachable ports.

My sip.conf contains:
[test]
type=friend
username=test
secret=12345
host=dynamic
nat=yes
qualify=1000
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
canreinvite=no

On the NAT'ed side I have the BudgetTone set up to use STUN and ports 5060
for SIP and 19000 for RTP.  The firewall that performs NAT forwards ports
5060 and 19000-19100 UDP to the phone.

An ethereal snapshot looks like:

1.1.1.1 = Asterisk server
2.2.2.2 = Public IP where the BudgeTone is
10.0.3.205 = Private IP of BudgeTone

Frame 211 (214 bytes on wire, 214 bytes captured)
Ethernet II, Src: 00:06:29:ce:5f:f2, Dst: 00:01:c7:0b:70:22
Destination: 00:01:c7:0b:70:22 (Cisco_0b:70:22)
Source: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2)
Type: IP (0x0800)
Internet Protocol, Src Addr: 1.1.1.1 (1.1.1.1), Dst Addr: 2.2.2.2 (2.2.2.2)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
Total Length: 200
Identification: 0x (0)
Flags: 0x04
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0x2538 (correct)
Source: 1.1.1.1 (1.1.1.1)
Destination: 2.2.2.2 (2.2.2.2)
User Datagram Protocol, Src Port: 13364 (13364), Dst Port: 19000 (19000)
Source port: 13364 (13364)
Destination port: 19000 (19000)
Length: 180
Checksum: 0xdf43 (correct)
Real-Time Transport Protocol
10..  = Version: RFC 1889 Version (2)
..0.  = Padding: False
...0  = Extension: False
  = Contributing source identifiers count: 0
0...  = Marker: False
.000 1000 = Payload type: ITU-T G.711 PCMA (8)
Sequence number: 45554
Timestamp: 16480
Synchronization Source identifier: 1847249288
Payload: E4E4E5FAF9FDF0F6F5C2C5DFD0575D58...

Frame 212 (214 bytes on wire, 214 bytes captured)
Ethernet II, Src: 00:01:c7:0b:70:22, Dst: 00:06:29:ce:5f:f2
Destination: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2)
Source: 00:01:c7:0b:70:22 (Cisco_0b:70:22)
Type: IP (0x0800)
Internet Protocol, Src Addr: 2.2.2.2 (2.2.2.2), Dst Addr: 1.1.1.1 (1.1.1.1)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
Total Length: 200
Identification: 0xe398 (58264)
Flags: 0x00
Fragment offset: 0
Time to live: 233
Protocol: UDP (0x11)
Header checksum: 0xd89e (correct)
Source: 2.2.2.2 (2.2.2.2)
Destination: 1.1.1.1 (1.1.1.1)
User Datagram Protocol, Src Port: 19000 (19000), Dst Port: 13364 (13364)
Source port: 19000 (19000)
Destination port: 13364 (13364)
Length: 180
Checksum: 0xb38f (incorrect, should be 0x1dc4)
Real-Time Transport Protocol
10..  = Version: RFC 1889 Version (2)
..0.  = Padding: False
...0  = Extension: False
  = Contributing source identifiers count: 0
0...  = Marker: False
.000 1000 = Payload type: ITU-T G.711 PCMA (8)
Sequence number: 53058
Timestamp: 3449661727
Synchronization Source identifier: 3820906983
Payload: D4D4D5D5D555D5D555D4D5D5D5D4D4D4...

Frame 213 (214 bytes on wire, 214 bytes captured)
Ethernet II, Src: 00:06:29:ce:5f:f2, Dst: 00:01:c7:0b:70:22
Destination: 00:01:c7:0b:70:22 (Cisco_0b:70:22)
Source: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2)
Type: IP (0x0800)
Internet Protocol, Src Addr: 1.1.1.1 (1.1.1.1), Dst Addr: 2.2.2.2 (2.2.2.2)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
Total Length: 200
Identification: 0x (0)
Flags: 0x04
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0x2538 (correct)
Source: 1.1.1.1 (1.1.1.1)
Destination: 2.2.2.2 (2.2.2.2)
User Datagram Protocol, Src Port: 13364 (13364), Dst Port: 19000 (19000)
Source port: 13364 (13364)
Destination port: 19000 (19000)
Length: 180
Checksum: 0xa9d4 (correct)
Real-Time Transport Protocol
10..  = Version: RFC 1889 Version (2)
..0.  

Re: [Asterisk-Users] far end disconnect supervision

[James Sharp]

 If some channel banks don't support this, how on earth do they know when
 the telco side of the call has hung up ?

They don't.  They rely on either a timeout or the called party hanging up
to disconnect the call.
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[Asterisk-Users] Record calls where to put line?

[Jimmy Riley]

Here is what I have now. Where should the line  exten =
_.,1,macro(record-on,${EXTEN},${CALLERIDNUM}) go should it be under [sip]?
Right now if I call sip to sip monitoring starts and the calls connect but I
only get 44 byte files. If I call and iaxtel number monitoring starts but
call never gets placed and again 44byte files with nothing in them.
Thanks for the help.

[iaxtel]

exten =
_1700XXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1888NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1877NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1866NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten =
_1800NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED])


[sip]
include = iaxtel
exten = _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})
exten = s,1,Dial(SIP/one|20|tr)
exten = 1001,1,Dial(SIP/one|20|tr)
exten = 1001,2,VoiceMail,u1001
exten = 1001,102,VocieMail,b1001
exten = 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001
exten = 1002,1,Dial(SIP/two|20|mtr)
exten = 1002,2,VoiceMail,u1002
exten = 1002,102,VoiceMail,b1002
exten = 6001,1,Ringing
exten = 6001,2,Wait(2)
exten = 6001,3,VoicemailMain



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Mann
Sent: January 10, 2004 12:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Record all phone calls

See Below
 
- Original Message - 
From: Jimmy Riley 
To: '[EMAIL PROTECTED]' 
Sent: Saturday, January 10, 2004 10:01 AM
Subject: [Asterisk-Users] Record all phone calls

I want to record all phone calls made inbound and outbound. I'm new so
having a hard time getting this started. Here is what I have so far but
isn't working. Can someone help me out? Thanks,

[macro-record-on]
exten = s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1})
exten = s,2,Monitor(wav,${CALLFILENAME})

[sip]
include = macro-record-on
include = iaxtel 
exten = _,1,macro(record-on,${EXTEN},${CALLERIDNUM})
exten = 1001,1,Dial(SIP/one|20|tr)
exten = 1001,2,VoiceMail,u1001
exten = 1001,102,VocieMail,b1001
exten = 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001
exten = 1002,1,Dial(SIP/two|20|mtr)
exten = 1002,2,VoiceMail,u1002
exten = 1002,102,VoiceMail,b1002
exten = 6001,1,Ringing
exten = 6001,2,Wait(2)
exten = 6001,3,VoicemailMain
 
There are a few issues I can see with this but your two big problems are as
follows.
 
You never want to include a macro.
include = macro-record-on
So remove that line altogether.
 
You show exten = _,1,macro(record-on,${EXTEN},${CALLERIDNUM})
the _ tells asterisk that you are going to want to match characters but then
you dont tell it what you want to match.
so exten = _.,1etc...  See the . after _ this tells * to match the rest of
the characters (digits)
 
Those are your two big issues with not getting the recording to start.

 
 

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Re: [Asterisk-Users] far end disconnect supervision

[Rich Adamson]

   I'm starting to shop for my first channel bank and one of the features
   that eveyone seems to recommend is far end disconnect supervision.
   What other terms do various manufactures use to describe this same
   feature ?
  
   Is calling party disconnect the same as far end disconnect
   supervision ?
  
  Yes, in most readers terms. However, in some cases marketing/sales
  people may have written stuff with no clue what they are talking
  about.
 
 
 Is far end disconnect supervision BOTH a service/feature/line
 signaling provided by the Telco AND a feature of some channel banks ?
 
 If some channel banks don't support this, how on earth do they know when
 the telco side of the call has hung up ?

If you go way back in history, channel banks were only used by telcos and
at least initially were only required to pass signaling between central 
office switches. It wasn't until fx cards were added that channel banks
had to be concerned with calling and called party disconnects. In some
states, the regulatory agencies governed what could (or could not) be
deployed and under what conditions. Called party disconnect was frequently
used by court order for police verification on certain calls, while
calling party disconnect was the norm. At that time, customer lines were
directly connected to the central office switch, and it was functions within
the switch that controlled calling/called party disconnects.

If the telco deployed a channel bank with fx-type customer interfaces, the 
channel bank would need to support calling and called party disconnect in 
order to inform the central office switch of call status.

If the telco deployed a channel bank with interfaces to a customer's pbx
where signaling used tones (as an example), the channel bank would not 
need the added electronics to support disconnect supervision.

Disconnect supervision refers to opening/closing the 2-wire circuit (as in
hanging up a telephone), and in some cases, reversing tip/ring (48 volt
polarity change).

(There are a number of other interfaces available for channel banks beside 
those designed for two-wire fx's.)

Since there are lots of old (and new) channel banks being sold on ebay, etc,
that may have been designed for different purposes, some will support
disconnect supervision while others do not, some are two-wire while others
are four-wire, some support E  M signaling (extra wires per channel), some
supply 100vac ringing voltage while others do not, some run on only 48 volt
DC power while others are 110 vac power, etc.

If you're looking for a channel bank to interface phones with asterisk, then
keywords would include 2-wire, disconnect supervision, fx lines, etc.
Might also ensure it can supply the needed 100 vac ringing voltage
(historically referred to as a ring generator).



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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[asterisk]

On Sat, Jan 10, 2004 at 03:03:23PM -0500, Brian Capouch wrote:
 
 I think you're unfairly impugning Digium's motives.  And I also think 
 you're--again--salting your post with enough innuendo that a reasonable 
 person might suspect you of flame-baiting.

Baiting, perhaps, but not flames. If there is some devil's advocate
flavour, call it tactical hyperbole.

Sometimes one has to take an extreme position to get things done --
remember that the previous thread resulted in a commitment to release
more often, and branch CVS for stable and development versions, and
the scheduling of a long overdue release for this Monday.

 I suscribe to the mailing lists of several OS VoIP solutions, as I'm 
 sure do many others on this list.  There is nothing out there like 
 asterisk, in terms of it functionality, or the body of minds that have 
 collected to work on it.  I have recently found myself embarking on a 
 mini-career doing fundamental-level VoIP training to network operators, 
 technology freaks, and even some small-telco tech people. I take along a 
 laptop with asterisk on it and do a little song-and-dance that shows off 
 some of its gee-whiz features.

I have found myself doing similar things...

 It is not much of an exaggeration to say that almost always people's 
 mouths drop open in amazement at what all that asterisk can do.  It's 
 comical sometimes how affected people are.

with similar experience...

 So I have all this functionality, and I have all the source code to it, 
 and I can legally keep it forever at this (mostly happy) level of 
 functionality, and if Digium drops off the face of the earth, I can 
 start with what's there (we can start with what's there; I know I 
 won't be alone) and keep going should that happen.

True, and i credit mark with foresight in releasing at least some
of the code as Free Software.

 So I can look at the same set of facts that you do, but in my mind 
 Digium is not the nefarious would-be crook that you imply in your 
 postings, but rather a brilliant and disruptive force upon the telco 
 world.  And they are a *business,* and as many of the people reading 
 this sentence are bound to know, one trick of the Open Source world is 
 to figure out how to keep things open and free and at the same time how 
 to keep bread on the table and enough cashflow to keep up with the 
 technology (VoIP in this case) Joneses.

I myself am a veteran of the packet vs. circuit, data vs. voice
wars of the mid-late 90s, having built networks for several
merged ISP/Telco entities. And from time to time I have worried
about how to keep bread on the table while at the same time
producing only Free Software.

I want to draw a distinction between Open Source software and
Free Software. Open Source is an attempt to strike some middle
ground between Intellectual Proprietorship and Intellectual
Freedom. Digium has chosen the middle ground that offers them
the advantage of asserting Intellectual Property Rights and 
granting others Freedom as they deem fit. And they have to
go through all sorts of contortions in order to be able to do
that -- to the point where it affects code quality. Decisions
are made for what amount to political reasons rather than 
technical ones. This, I believe, is damaging, and indicates 
that the wrong balance has been struck.

When I first encountered Asterisk about a year ago, my impression
was that Digium was a hardware vendor that produced Free Software
as a way to drive hardware purchases, and that they offered 
support as a way to augment their revenue stream. Then I learned
that this was not the case, and they also produced proprietary
versions of the software, and I was disappointed.

If Digium had released Asterisk under a BSD-like license, this
would not be much of an issue -- if anybody could have their own
proprietary Asterisk, I would not begrudge Digium that ability.
But since they are the only ones who can do that...

 I cannot guess your motives, but I'm pretty sure that I *do* know what 
 Digium's motives are, and they are innocuous and altruistic instead of 
 the way you portray them.

My motives are to encourage and maximize the Free flow of ideas.
I pursue this on several levels. I contribute code only to Free
Software. I advocate the use and development of Free Software. 
I build networks over which ideas can be exchanged unhindered.
(I sometimes use the terms idea and software interchangeably
since the latter is an explicit manifestation of the former in
machine readable form.)

I have always been suspicious of centralized control and dictatorship,
benevolent or otherwise. After thinking for some time about the 
licensing structure of code for Asterisk, I am not sure that
their motives are so innocuous and altrusitic, or at least
this is not reflected so well in the fine print. After learning
that all code must pass through Mark, I am even less sure.
It means that Digium remains in a position of control and 
dominance over what is ostensibly 

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Arnd Vehling]

Hi,

thats very probably a NAT problem. Your NAT box is probaly blocking
the incoming UDP voice stream. 

If asteriks supports a RTP Proxy you can try that.

best regards,

  Arnd
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Re: [Asterisk-Users] Oops!

[Steven Critchfield]

On Sat, 2004-01-10 at 11:22, Terence Parker wrote:
 Didn't realise that replies are still tagged to specific threads in the 
 mail headers. Oops!
 
 A few of my postings so far have been replies (to save me retyping the 
 list address) - but aren't really replies (they are completely off 
 topic).
 
 Hope this doesn't cause too many problems in the archives!
 
 But... at least now I know!

I guess lazyness applies to learning too since it is possible to just
click on the mailing list address in the message and get a new message
that isn't linked to the old message. Seems to me that is easier to do
than erasing the quoted message and old subject. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] far end disconnect supervision

[Lance Arbuckle]

Rich Adamson wrote:
 
I'm starting to shop for my first channel bank and one of the features
that eveyone seems to recommend is far end disconnect supervision.
What other terms do various manufactures use to describe this same
feature ?
   
Is calling party disconnect the same as far end disconnect
supervision ?
  
   Yes, in most readers terms. However, in some cases marketing/sales
   people may have written stuff with no clue what they are talking
   about.
 
 
  Is far end disconnect supervision BOTH a service/feature/line
  signaling provided by the Telco AND a feature of some channel banks ?
 
  If some channel banks don't support this, how on earth do they know when
  the telco side of the call has hung up ?
 
 If you go way back in history, channel banks were only used by telcos and
 at least initially were only required to pass signaling between central
 office switches. It wasn't until fx cards were added that channel banks
 had to be concerned with calling and called party disconnects. In some
 states, the regulatory agencies governed what could (or could not) be
 deployed and under what conditions. Called party disconnect was frequently
 used by court order for police verification on certain calls, while
 calling party disconnect was the norm. At that time, customer lines were
 directly connected to the central office switch, and it was functions within
 the switch that controlled calling/called party disconnects.
 
 If the telco deployed a channel bank with fx-type customer interfaces, the
 channel bank would need to support calling and called party disconnect in
 order to inform the central office switch of call status.
 
 If the telco deployed a channel bank with interfaces to a customer's pbx
 where signaling used tones (as an example), the channel bank would not
 need the added electronics to support disconnect supervision.
 
 Disconnect supervision refers to opening/closing the 2-wire circuit (as in
 hanging up a telephone), and in some cases, reversing tip/ring (48 volt
 polarity change).
 
 (There are a number of other interfaces available for channel banks beside
 those designed for two-wire fx's.)
 
 Since there are lots of old (and new) channel banks being sold on ebay, etc,
 that may have been designed for different purposes, some will support
 disconnect supervision while others do not, some are two-wire while others
 are four-wire, some support E  M signaling (extra wires per channel), some
 supply 100vac ringing voltage while others do not, some run on only 48 volt
 DC power while others are 110 vac power, etc.
 
 If you're looking for a channel bank to interface phones with asterisk, then
 keywords would include 2-wire, disconnect supervision, fx lines, etc.
 Might also ensure it can supply the needed 100 vac ringing voltage
 (historically referred to as a ring generator).


Yes, I'm looking for a channel bank to interface analog phones and pstn
lines to asterisk.  I've got a simple test system setup with the TDM10B
and X100p and want to continue learning asterisk with the T1 card.

I've been reading the list archives and searching Ebay for channel
banks.  There are lots of Carrier Access Corp (CAC) Access Bank I and
II's available but I don't explicitly see anything about calling party
disconnect in the user manuals.  The CAC Adit 600 manual, on the other
hand, states that calling party disconnect is supported on the FXO
interfaces.  That's great, but I haven't seen any on Ebay and I can't
justify the expense of a new unit.  Acording to the list archives the
Adtran 750/850 units work well but I haven't read the manuals for those
units yet.

Basically, I was hoping to purchase a functional, used, older channel
bank for a few hundred dollars so that I could continue learning
Asterisk and to also have a system to demonstrate to potential clients. 
Any suggestions welcome.

-Lance
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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[Brian West]

I'm going to keep this short and to the point.

Nobody is twisting your arm to use Asterisk...

we didn't find you.. you found us.

NEXT!!!

bkw
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Re: [Asterisk-Users] drop calls with T100P / PRI

[Steven Critchfield]

On Sat, 2004-01-10 at 15:19, John Brown (CV) wrote:
 busydetect=yes
 callprogress=yes
 musiconhold=default
 signalling=pri_cpe
 group=1
 channel= 1-4

Well seems you haven't been on the list, or maybe you haven't been
paying attention since we have been covering that problem for a while
lately. PRI has busydetect and callprogress built into the D channel and
is absolutely known. Those 2 options are for analog links where the
signaling is not always accurate. 
 

-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Chandra]

i also had the same problem temporarily i solved my problem with both
outside NAT. u can also do it if both inside NAT. * outside NAT and
Budgetone behind NAT simply doesn't seem to work. if u ever solve this
problem please let me know too.

thanks

cm

- Original Message -
From: Owen Kelso [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 11, 2004 4:52 AM
Subject: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)


 I'm using Asterisk on a open server (no firewall or NAT) and trying to
 communicate with a Grandstream BudgeTone 102 SIP phone which is behind
 NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS
 about a week ago.  My problem is that I'm only getting half-duplex
 communication -- I can hear voice from the Asterisk server but the server
 does not understand any voice from me.  From the console sip debug shows
 that the SIP part is working fine and DTMF via SIP INFO works.

 I've struggled with this for a few days now and can't figure out the
 cause.  The only symptoms I've found are:

 (1) When I make a call the console spits out the following errors several
 times per minute:
 WARNING[-1220854864]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
 Resource temporarily unavailable

 (2) An ethereal trace reveals that incoming RTP packets have failed UDP
 checksums (all packets have the same checksum of 0xb38f).  I don't see
 anything else irregular, like unreachable ports.

 My sip.conf contains:
 [test]
 type=friend
 username=test
 secret=12345
 host=dynamic
 nat=yes
 qualify=1000
 dtmfmode=info
 disallow=all
 allow=ulaw
 allow=alaw
 canreinvite=no

 On the NAT'ed side I have the BudgetTone set up to use STUN and ports 5060
 for SIP and 19000 for RTP.  The firewall that performs NAT forwards ports
 5060 and 19000-19100 UDP to the phone.

 An ethereal snapshot looks like:

 1.1.1.1 = Asterisk server
 2.2.2.2 = Public IP where the BudgeTone is
 10.0.3.205 = Private IP of BudgeTone

 Frame 211 (214 bytes on wire, 214 bytes captured)
 Ethernet II, Src: 00:06:29:ce:5f:f2, Dst: 00:01:c7:0b:70:22
 Destination: 00:01:c7:0b:70:22 (Cisco_0b:70:22)
 Source: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2)
 Type: IP (0x0800)
 Internet Protocol, Src Addr: 1.1.1.1 (1.1.1.1), Dst Addr: 2.2.2.2
(2.2.2.2)
 Version: 4
 Header length: 20 bytes
 Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
 Total Length: 200
 Identification: 0x (0)
 Flags: 0x04
 Fragment offset: 0
 Time to live: 64
 Protocol: UDP (0x11)
 Header checksum: 0x2538 (correct)
 Source: 1.1.1.1 (1.1.1.1)
 Destination: 2.2.2.2 (2.2.2.2)
 User Datagram Protocol, Src Port: 13364 (13364), Dst Port: 19000 (19000)
 Source port: 13364 (13364)
 Destination port: 19000 (19000)
 Length: 180
 Checksum: 0xdf43 (correct)
 Real-Time Transport Protocol
 10..  = Version: RFC 1889 Version (2)
 ..0.  = Padding: False
 ...0  = Extension: False
   = Contributing source identifiers count: 0
 0...  = Marker: False
 .000 1000 = Payload type: ITU-T G.711 PCMA (8)
 Sequence number: 45554
 Timestamp: 16480
 Synchronization Source identifier: 1847249288
 Payload: E4E4E5FAF9FDF0F6F5C2C5DFD0575D58...

 Frame 212 (214 bytes on wire, 214 bytes captured)
 Ethernet II, Src: 00:01:c7:0b:70:22, Dst: 00:06:29:ce:5f:f2
 Destination: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2)
 Source: 00:01:c7:0b:70:22 (Cisco_0b:70:22)
 Type: IP (0x0800)
 Internet Protocol, Src Addr: 2.2.2.2 (2.2.2.2), Dst Addr: 1.1.1.1
(1.1.1.1)
 Version: 4
 Header length: 20 bytes
 Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
 Total Length: 200
 Identification: 0xe398 (58264)
 Flags: 0x00
 Fragment offset: 0
 Time to live: 233
 Protocol: UDP (0x11)
 Header checksum: 0xd89e (correct)
 Source: 2.2.2.2 (2.2.2.2)
 Destination: 1.1.1.1 (1.1.1.1)
 User Datagram Protocol, Src Port: 19000 (19000), Dst Port: 13364 (13364)
 Source port: 19000 (19000)
 Destination port: 13364 (13364)
 Length: 180
 Checksum: 0xb38f (incorrect, should be 0x1dc4)
 Real-Time Transport Protocol
 10..  = Version: RFC 1889 Version (2)
 ..0.  = Padding: False
 ...0  = Extension: False
   = Contributing source identifiers count: 0
 0...  = Marker: False
 .000 1000 = Payload type: ITU-T G.711 PCMA (8)
 Sequence number: 53058
 Timestamp: 3449661727
 Synchronization Source identifier: 3820906983
 Payload: D4D4D5D5D555D5D555D4D5D5D5D4D4D4...

 Frame 213 (214 bytes on wire, 214 bytes captured)
 Ethernet II, Src: 00:06:29:ce:5f:f2, Dst: 00:01:c7:0b:70:22
 Destination: 00:01:c7:0b:70:22 (Cisco_0b:70:22)
 Source: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2)
 Type: IP (0x0800)
 Internet Protocol, Src Addr: 1.1.1.1 (1.1.1.1), Dst Addr: 2.2.2.2
(2.2.2.2)
 Version: 4
 Header length: 20 bytes
 Differentiated Services 

[Asterisk-Users] default music source for SIP channel

[Lance Arbuckle]

The wiki says this about the MusicOnHold command:

Plays hold music specified by class. If omitted, the default music
source for the channel will be used.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold

How do I set the default music on hold class for the SIP channel ?  I
tried adding musiconhold=test to my sip.conf.
musiconhold.conf looks like this:
  [classes]
  default = quietmp3:/var/lib/asterisk/mohmp3
  loud = mp3:/var/lib/asterisk/mohmp3
  random = quietmp3:/var/lib/asterisk/mohmp3,-z
  test = quietmp3:/var/lib/asterisk/mohmp3,-z

in extensions.conf I did:
  exten = 6000,1,Answer 
  exten = 6000,2,MusicOnHold

When I dial 6000 from a SIP phone ( xlite), musiconhold starts to play,
but from the 'default' class.
What am I screwing up ?

-Lance
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[Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Anton Tinchev]

Just spended ~ hour googling - all boards are based on GC-XX or I750X 
Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 
800Mhz FSB.
Thanks
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RE: [Asterisk-Users] default music source for SIP channel

[ml]

 The wiki says this about the MusicOnHold command:
 
 Plays hold music specified by class. If omitted, the default music
 source for the channel will be used.
 http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold
 
 How do I set the default music on hold class for the SIP channel ?  I
 tried adding musiconhold=test to my sip.conf.
 musiconhold.conf looks like this:
   [classes]
   default = quietmp3:/var/lib/asterisk/mohmp3
   loud = mp3:/var/lib/asterisk/mohmp3
   random = quietmp3:/var/lib/asterisk/mohmp3,-z
   test = quietmp3:/var/lib/asterisk/mohmp3,-z
 
 in extensions.conf I did:
   exten = 6000,1,Answer 
   exten = 6000,2,MusicOnHold
 
 When I dial 6000 from a SIP phone ( xlite), musiconhold starts to
 play,
 but from the 'default' class.
 What am I screwing up ?
 
 -Lance

  -= Info about application 'SetMusicOnHold' =- 

[Synopsis]:
Set default Music On Hold class

[Description]:
SetMusicOnHold(class): Sets the default class for music on hold for a given channel.  
When
music on hold is activated, this class will be used to select which
music is played.

Kevin
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Re: [Asterisk-Users] default music source for SIP channel

[Lance Arbuckle]

[EMAIL PROTECTED] wrote:
 
  The wiki says this about the MusicOnHold command:
 
  Plays hold music specified by class. If omitted, the default music
  source for the channel will be used.
  http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold
 
  How do I set the default music on hold class for the SIP channel ?  I
  tried adding musiconhold=test to my sip.conf.
  musiconhold.conf looks like this:
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
loud = mp3:/var/lib/asterisk/mohmp3
random = quietmp3:/var/lib/asterisk/mohmp3,-z
test = quietmp3:/var/lib/asterisk/mohmp3,-z
 
  in extensions.conf I did:
exten = 6000,1,Answer
exten = 6000,2,MusicOnHold
 
  When I dial 6000 from a SIP phone ( xlite), musiconhold starts to
  play,
  but from the 'default' class.
  What am I screwing up ?
 
  -Lance
 
   -= Info about application 'SetMusicOnHold' =-
 
 [Synopsis]:
 Set default Music On Hold class
 
 [Description]:
 SetMusicOnHold(class): Sets the default class for music on hold for a given channel. 
  When
 music on hold is activated, this class will be used to select which
 music is played.
 
 Kevin

Thanks Kevin, but boy, do I feel dumb.  Maybe someone could update the
MusicOnHold wiki page and add SetMusicOnHold to the Also See section.

-Lance
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RE: [Asterisk-Users] Chagres Technologies, Inc

[mattf]

Hello,

I have the shipping numbers for the first 2 shipments of 40 phones but I do
not for the last 20 can you please send that to me as well as the serial
numbers of all 100 phones. and I have tried calling you, the week before
Christmas. I left a message and received no call back.

MATT---


-Original Message-
From: John Brown (CV) [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


Hi List,

Matt hasn't contacted us directly about this.  I've
responded to his previous statement that he hasn't 
recevied the last 20 units, and never heard back from
him.

Matt, again, if this is an issue please do contact us.
Our CDR and SMTP logs show no such attempt.

Our inventory records show 100 Grandstream Serial Numbers
have been shipped to you, along with tracking numbers.

+1 505 830 1200   Office Number, Auto Attendent answers 
  Pressing 0 takes you to a operator
  M-F 9-5 MDT (GMT-7)

orders at chagres dot net  gets email into the order admin
   which replies within 1 biz day
   and you should get a auto reply.

our email system now auto replys to help verify that your
email did reach us.  If you don't get an auto reply to 
the sales or order  role accounts then our SMTP box didn't
get your email.



On Sat, Jan 10, 2004 at 12:57:43AM -0500, mattf wrote:

 As I've said several times on this list[insert usual apology here], I
still
 haven't received the last 20  of 100 phones I ordered over 2 months ago.
If
 you get a hold of them please let me know
 
 MATT---
 
 
 -Original Message-
 From: mikeu [mailto:[EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 12:41 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Chagres Technologies, Inc
 
 
 
 Anyone else having problems getting product from Chagres?  They took my
 payment almost two months ago and I still have not seen hardware.  They
have
 been horribly unresponsive to my e-mails.
 
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Re: [Asterisk-Users] far end disconnect supervision

[Rich Adamson]

Lance,

 Yes, I'm looking for a channel bank to interface analog phones and pstn
 lines to asterisk.  I've got a simple test system setup with the TDM10B
 and X100p and want to continue learning asterisk with the T1 card.
 
 I've been reading the list archives and searching Ebay for channel
 banks.  There are lots of Carrier Access Corp (CAC) Access Bank I and
 II's available but I don't explicitly see anything about calling party
 disconnect in the user manuals.  The CAC Adit 600 manual, on the other
 hand, states that calling party disconnect is supported on the FXO
 interfaces.  That's great, but I haven't seen any on Ebay and I can't
 justify the expense of a new unit.  Acording to the list archives the
 Adtran 750/850 units work well but I haven't read the manuals for those
 units yet.
 
 Basically, I was hoping to purchase a functional, used, older channel
 bank for a few hundred dollars so that I could continue learning
 Asterisk and to also have a system to demonstrate to potential clients. 
 Any suggestions welcome.

I've not implemented any form of channel bank with *, so can't offer much
help on specific vendor/models. Since there are a fair number of folks
using them on the list, try posting a new thread with channel bank in
the subject, and summarize the responses in the wiki.

Personal opinion is that throwing a channel bank at * (other then for
better echo cancellation on trunks) is like taking a shower with your
socks on; don't see any practical use. But, I'm sure some do.


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Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Steve Underwood]

Anton Tinchev wrote:

Just spended ~ hour googling - all boards are based on GC-XX or I750X 
Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 
with 800Mhz FSB.
Thanks
Unless one has appeared in the last couple of weeks, there are none. In 
fact, the only one I know of for any kind of non-Xeon Pentium is the 
Dell 600SC. That one isn't an 800MHz bus machine.

Regards,
Steve
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RE: [Asterisk-Users] Chagres Technologies, Inc

[Sean Cheesman]

time to take this off-list.

-Original Message-
From: mattf [mailto:[EMAIL PROTECTED] 
Sent: Saturday, January 10, 2004 10:05 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Chagres Technologies, Inc


Hello,

I have the shipping numbers for the first 2 shipments of 40 phones but I
do not for the last 20 can you please send that to me as well as the
serial numbers of all 100 phones. and I have tried calling you, the week
before Christmas. I left a message and received no call back.

MATT---


-Original Message-
From: John Brown (CV) [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc


Hi List,

Matt hasn't contacted us directly about this.  I've
responded to his previous statement that he hasn't 
recevied the last 20 units, and never heard back from
him.

Matt, again, if this is an issue please do contact us.
Our CDR and SMTP logs show no such attempt.

Our inventory records show 100 Grandstream Serial Numbers
have been shipped to you, along with tracking numbers.

+1 505 830 1200   Office Number, Auto Attendent answers 
  Pressing 0 takes you to a operator
  M-F 9-5 MDT (GMT-7)

orders at chagres dot net  gets email into the order admin
   which replies within 1 biz day
   and you should get a auto reply.

our email system now auto replys to help verify that your
email did reach us.  If you don't get an auto reply to 
the sales or order  role accounts then our SMTP box didn't
get your email.



On Sat, Jan 10, 2004 at 12:57:43AM -0500, mattf wrote:

 As I've said several times on this list[insert usual apology here], I
still
 haven't received the last 20  of 100 phones I ordered over 2 months 
 ago.
If
 you get a hold of them please let me know
 
 MATT---
 
 
 -Original Message-
 From: mikeu [mailto:[EMAIL PROTECTED]
 Sent: Saturday, January 10, 2004 12:41 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Chagres Technologies, Inc
 
 
 
 Anyone else having problems getting product from Chagres?  They took 
 my payment almost two months ago and I still have not seen hardware.  
 They
have
 been horribly unresponsive to my e-mails.
 
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Re: [Asterisk-Users] R2 Digital - Brazil

[Steve Underwood]

Hi Daniel,

You will find libr2 is only about 10% of an implementation, and a bad 
one at that. I now have 95% of a good implementation, but its not yet 
released.

Regards,
Steve
Daniel Bichara wrote:

Hi all,

I will start testing libr2 for brazilian R2. Any clue?

Daniel


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RE: [Asterisk-Users] far end disconnect supervision

[Sean Cheesman]

Am I missing something?  Is there another way to pipe large quantities
of analog lines (FXS or FXO) into *?  Seriously, is there another way?

Sean

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: Saturday, January 10, 2004 9:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] far end disconnect supervision


Lance,

 Yes, I'm looking for a channel bank to interface analog phones and 
 pstn lines to asterisk.  I've got a simple test system setup with the 
 TDM10B and X100p and want to continue learning asterisk with the T1 
 card.
 
 I've been reading the list archives and searching Ebay for channel 
 banks.  There are lots of Carrier Access Corp (CAC) Access Bank I and 
 II's available but I don't explicitly see anything about calling 
 party disconnect in the user manuals.  The CAC Adit 600 manual, on 
 the other hand, states that calling party disconnect is supported on

 the FXO interfaces.  That's great, but I haven't seen any on Ebay and 
 I can't justify the expense of a new unit.  Acording to the list 
 archives the Adtran 750/850 units work well but I haven't read the 
 manuals for those units yet.
 
 Basically, I was hoping to purchase a functional, used, older channel 
 bank for a few hundred dollars so that I could continue learning 
 Asterisk and to also have a system to demonstrate to potential 
 clients. Any suggestions welcome.

I've not implemented any form of channel bank with *, so can't offer
much help on specific vendor/models. Since there are a fair number of
folks using them on the list, try posting a new thread with channel bank
in the subject, and summarize the responses in the wiki.

Personal opinion is that throwing a channel bank at * (other then for
better echo cancellation on trunks) is like taking a shower with your
socks on; don't see any practical use. But, I'm sure some do.


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Re: [Asterisk-Users] far end disconnect supervision

[Jonathan Moore]

I found this supplier on ebay and they seem to have a regular supply of Adit
600s. I spent $800 for one with 8 fxo/8 fxs. This may be out of your price
range, but I think it is a pretty good deal for this model. I really recommend
the 600. It did amazing things to eleminate our echo problems. My experience so
far with the ebay thing is you may wait a while to find something with all the
features you want and with fxo cards and fxo cards are pricy when bought
seperately/new. The way I finally got a unit was to use ebay to find vendors
selling Adits and then contect them directly for the configuration I wanted.

I made the wrong initial purchase and ended up with an AB1 (with broken ring
generator to boot). They definetely do not support disconnect supervision.

http://www.sunteldata.com/

-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Lance Arbuckle [EMAIL PROTECTED]:

 
 
 Rich Adamson wrote:
  
 I'm starting to shop for my first channel bank and one of the
 features
 that eveyone seems to recommend is far end disconnect supervision.
 What other terms do various manufactures use to describe this same
 feature ?

 Is calling party disconnect the same as far end disconnect
 supervision ?
   
Yes, in most readers terms. However, in some cases marketing/sales
people may have written stuff with no clue what they are talking
about.
  
  
   Is far end disconnect supervision BOTH a service/feature/line
   signaling provided by the Telco AND a feature of some channel banks ?
  
   If some channel banks don't support this, how on earth do they know when
   the telco side of the call has hung up ?
  
  If you go way back in history, channel banks were only used by telcos and
  at least initially were only required to pass signaling between central
  office switches. It wasn't until fx cards were added that channel banks
  had to be concerned with calling and called party disconnects. In some
  states, the regulatory agencies governed what could (or could not) be
  deployed and under what conditions. Called party disconnect was
 frequently
  used by court order for police verification on certain calls, while
  calling party disconnect was the norm. At that time, customer lines were
  directly connected to the central office switch, and it was functions
 within
  the switch that controlled calling/called party disconnects.
  
  If the telco deployed a channel bank with fx-type customer interfaces, the
  channel bank would need to support calling and called party disconnect
 in
  order to inform the central office switch of call status.
  
  If the telco deployed a channel bank with interfaces to a customer's pbx
  where signaling used tones (as an example), the channel bank would not
  need the added electronics to support disconnect supervision.
  
  Disconnect supervision refers to opening/closing the 2-wire circuit (as in
  hanging up a telephone), and in some cases, reversing tip/ring (48 volt
  polarity change).
  
  (There are a number of other interfaces available for channel banks beside
  those designed for two-wire fx's.)
  
  Since there are lots of old (and new) channel banks being sold on ebay,
 etc,
  that may have been designed for different purposes, some will support
  disconnect supervision while others do not, some are two-wire while others
  are four-wire, some support E  M signaling (extra wires per channel),
 some
  supply 100vac ringing voltage while others do not, some run on only 48
 volt
  DC power while others are 110 vac power, etc.
  
  If you're looking for a channel bank to interface phones with asterisk,
 then
  keywords would include 2-wire, disconnect supervision, fx lines,
 etc.
  Might also ensure it can supply the needed 100 vac ringing voltage
  (historically referred to as a ring generator).
 
 
 Yes, I'm looking for a channel bank to interface analog phones and pstn
 lines to asterisk.  I've got a simple test system setup with the TDM10B
 and X100p and want to continue learning asterisk with the T1 card.
 
 I've been reading the list archives and searching Ebay for channel
 banks.  There are lots of Carrier Access Corp (CAC) Access Bank I and
 II's available but I don't explicitly see anything about calling party
 disconnect in the user manuals.  The CAC Adit 600 manual, on the other
 hand, states that calling party disconnect is supported on the FXO
 interfaces.  That's great, but I haven't seen any on Ebay and I can't
 justify the expense of a new unit.  Acording to the list archives the
 Adtran 750/850 units work well but I haven't read the manuals for those
 units yet.
 
 Basically, I was hoping to purchase a functional, used, older channel
 bank for a few hundred dollars so that I could continue learning
 Asterisk and to also have a system to demonstrate to potential clients. 
 Any suggestions welcome.
 
 -Lance
 ___
 

Re: [Asterisk-Users] ADSI Configs

[C. Maj]

On Sat, 10 Jan 2004, Lee Redmayne waxed:

 Hi All
  
 If I want to get my ADSI Phones (successfully connected off a Rhino Channel
 Bank and TE410P) to connect to Asterisk to get their config downloaded, is
 there something specific needed in extensions.conf for them to dial to get
 this?
 
 Thanks :)

You'll need to set them up with adsi=yes in zapata.conf,
then try making an extension for VoiceMailMain and dial into
it from your ADSI phone.  I think that's a good start.  But
if your phone is locked, you might run into snags.  Check
the list archives for locked ADSI if that's the case.

--Chris


-- 

Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

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Re: [Asterisk-Users] Asterisk Development Updates

[Greg Boehnlein]

On Thu, 8 Jan 2004, Mark Spencer wrote:

 Prompted by the recent discussion on the mailing list regarding the
 Asterisk development and release process (or lack thereof), John Todd,
 Thorsten Lockert, Brian K. West, and myself have put together a plan to
 address the most significant two legitimate concerns that have been
 expressed regarding these processes.  Specifically:
 
 Concern #1: Asterisk release schedules and path to 1.0.0
 
 Asterisk version 0.7.0 will be released by Monday Jan 12, 2004.  Later
 that week, we will create a stable branch from which eventually 1.0.0 will
 be tagged.  Only bug fixes will go into the release branch, while feature
 requests and bug fixes will continue to go into the head branch.  If you
 are currently using CVS asterisk on a production server, we suggest that
 you move to the new stable CVS branch when it becomes available.
 Instructions for using the new stable CVS branch will be made available on
 asterisk.org next week.  Snapshots of the stable branch will also be made
 available periodically as Asterisk 0.9.x for those not using CVS.  If you
 wish to remain on the cutting edge, you may leave your system using the
 head CVS as it is currently.

Awesome! I'm game to create Asterisk RPMS when the stable branch comes 
out!
 
 Concern #2: Slow integration of bug fixes and feature requests into CVS
 
 With the assistance of John Todd and Brian West, we have added
 documentation about how the bug tracker operates, available at
 http://www.digium.com/bugtracker.html.  This document should help new
 users understand how the process of submitting bugs works, how to properly
 follow up on bugs to be sure they get applied, and how to contribute to
 the bug tracking process as a Bug Marshal, thus accellerating the process.
 In addition, I am commiting 5-10 dedicated hours of my own time per week
 to work with Bug Marshals on reviewing bugs, patches and feature requests.
 
 Conclusion:
 
 Hopefully these steps will help improve the quality and stability of the
 Asterisk code, and make it easier for people who wish to contribute to
 Asterisk to do so, while maintaining Asterisk's availibility to continue
 to advance new features and applications.
 
 Mark, John, Thorsten, and Brian
 
 
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-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] far end disconnect supervision

[Jonathan Moore]

Quoting Rich Adamson [EMAIL PROTECTED]:
 
 Personal opinion is that throwing a channel bank at * (other then for
 better echo cancellation on trunks) is like taking a shower with your
 socks on; don't see any practical use. But, I'm sure some do.
 
This statement assumes a lot. 

1. That you don't want to use analog phones.
2. That you need enough lines for a T1/PRI circuit. In this case the most
economical way to deploy a setup of say 5-16 voice lines is with a channel bank
and a T1 card in the * server. If there is some less expensive option please
share :-). 


Visit Winfield Public Schools at http://usd465.com
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Re: [Asterisk-Users] drop calls with T100P / PRI

[C. Maj]

On Sat, 10 Jan 2004, Steven Critchfield waxed:

 On Sat, 2004-01-10 at 15:19, John Brown (CV) wrote:
  busydetect=yes
  callprogress=yes
  musiconhold=default
  signalling=pri_cpe
  group=1
  channel= 1-4
 
 Well seems you haven't been on the list, or maybe you haven't been
 paying attention since we have been covering that problem for a while
 lately. PRI has busydetect and callprogress built into the D channel and
 is absolutely known. Those 2 options are for analog links where the
 signaling is not always accurate. 

Easy, now.  I just added another T1, and I have noticed the
D-Channel dropping for a few seconds then coming right back,
fortunately in the wee hours when no one's on the lines.
But I have both busydetect and callprogress off, with a T400
tho not a T100.

Here's some zaptel.conf:

span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,1,0,esf,b8zs
span=4,0,0,esf,b8zs

First 2 spans are channel banks, span 3 is local T1, and
span 4 is long distance T1.  The D-Channel only appears to
drop on span 4, although I can't yet get the time to play
wire/card-swap much due to the machine being in production.
Both T1's are from the same switch, so I'm told, so it
should be the same clock.

Here's some log:

Jan  9 04:22:05 WARNING[5126]: File chan_zap.c, Line 5759 (zt_pri_error): PRI: Read on 
106 failed: Unknown error 500
Jan  9 04:22:05 WARNING[6151]: File chan_zap.c, Line 4708 (handle_init_event): 
Detected alarm on channel 73: Red Alarm
Jan  9 04:22:05 WARNING[6151]: File chan_zap.c, Line 1101 (zt_disable_ec): Unable to 
disable echo cancellation on channel 73
Jan  9 04:22:05 WARNING[6151]: File chan_zap.c, Line 4708 (handle_init_event): 
Detected alarm on channel 74: Red Alarm
Jan  9 04:22:05 WARNING[6151]: File chan_zap.c, Line 1101 (zt_disable_ec): Unable to 
disable echo cancellation on channel 74
...last 2 lines repeated for each channel on span 4 (up to channel 95) all at the 
same time
Jan  9 04:22:11 NOTICE[6151]: File chan_zap.c, Line 4703 (handle_init_event): Alarm 
cleared on channel 73
Jan  9 04:22:11 NOTICE[6151]: File chan_zap.c, Line 4703 (handle_init_event): Alarm 
cleared on channel 74
...again repeated at the same time for every channel on span 4
Jan  9 04:22:11 WARNING[5126]: File chan_zap.c, Line 5759 (zt_pri_error): PRI: Read on 
106 failed: Unknown error 500
Jan  9 04:22:14 VERBOSE[5126]:   == D-Channel on span 4 down
Jan  9 04:22:20 VERBOSE[5126]:   == D-Channel on span 4 up
...and then the B-Channels start coming back up

I'm trying to blame it on the mobo and the telco, but now
that someone else is seeing it, maybe it's * ?  It wasn't a
problem when I had one T1, and span 3 doesn't seem to ever
drop, so maybe it's something with the span 4 port on the
T400 card ?  Or is this some side-effect of the known
buffering problem ?

--Chris


-- 

Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

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[Asterisk-Users] Using ACD functionality for main number answer and music on hold

[Lenny Tropiano / asterisk.org Mailing list]

I'm considering using the Agent login/logoff function to add to a queue
that will be our main number during the day to answer.  Periodically
our receptionist is not at her desk and would be useful for her to 
login elsewhere and get the main number calls to transfer as she sees 
fit.  If the agent's don't pick up in a specific amount of time, it's 
transferred to our main IVR...

I have the functionality working, but right now when you dial the main
number you get the musiconhold that is defined for that queue.  Is
there a way (short of recording a mp3 of a ringing phone) for the person
to get a ringing sound instead of the MOH?

Thanks,
Lenny
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Re: [Asterisk-Users] Asterisk Development Updates

[Jared Smith]

On Sat, 2004-01-10 at 20:25, Greg Boehnlein wrote:

 Awesome! I'm game to create Asterisk RPMS when the stable branch comes 
 out!

Great... I was going to do the same... maybe we should join forces and
make better RPMS!  (I've already got a semi-decent .spec file done.)

Jared Smith

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Re: [Asterisk-Users] default music source for SIP channel

[Lance Arbuckle]

Lance Arbuckle wrote:
 
 [EMAIL PROTECTED] wrote:
 
   The wiki says this about the MusicOnHold command:
  
   Plays hold music specified by class. If omitted, the default music
   source for the channel will be used.
   http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold


ok, I read the above statement from the wiki to mean a channel type like
ZAP or SIP or whatever.  Is this correct ?

Now, over here in the wiki
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf it says:
Editing your files to enable MusicOnHold 
In /etc/asterisk/zapata.conf, add the line musiconhold=default under
[channels] context 

To me this means that the default MusicOnHold (MOH) class can be set for
all Zap channels.  And this seams to work.  I can do musiconhold=random
and all calls from zap channels get MOH class random.  If I change
musiconhold=random to musiconhold=test all Zap calls get the new class
test.

For calls initiated from a SIP client, the SIP client always gets the
MOH class default.

I even reworked my sip.conf to send all sip calls to [from-sip] in
extensions.conf
[from-sip]
exten = s,1,setmusiconhold(test)
include = from-internal

So, why does zapata.conf accept musiconhold=class yet sip.conf ignores a
similar statement ?  Can anyone give me an example of how to control the
MOH class for a SIP channel ?

Thanks
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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Paul Liew]

From: Owen Kelso [EMAIL PROTECTED]
Sent: Sunday, January 11, 2004 10:07 AM
Subject: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)



 On the NAT'ed side I have the BudgetTone set up to use STUN and ports 5060
 for SIP and 19000 for RTP.  The firewall that performs NAT forwards ports
 5060 and 19000-19100 UDP to the phone.

Hi Owen,

Even though your GS is behind a NAT, it shouldn't be set for STUN unless
you're actually using a STUN server. I have GS installed in many locations
behind a NAT but without the STUN option set.

Paul

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[Asterisk-Users] how do i make this happen [macro-record-cleanup]

[Jimmy Riley]

[macro-record-on]
exten = s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1})
exten = s,2,Monitor(wav,${CALLFILENAME})

[macro-record-cleanup]

exten = s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) 
exten = s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3) 
exten = s,3,System(soxmix ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/${CALLFILENAME}-out.wav  ${MONITORDIR}/${CALLFILENAME}.gsm) 
exten = s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/${CALLFILENAME}-out.wav) 
exten = s,5,NoOp



[sip]
exten = 1001,1,macro(record-on,${EXTEN},${CALLERIDNUM})
exten = 1001,2,Dial(SIP/one|20|tr)


Jimmy Riley
Network Administrator
VeriCore
985-626-1701 X1103

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[Asterisk-Users] WTB / WTS Voip hardware

[Kent Williams]

I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has
been working fine for me now for a while.
These have been pulled out of a working Asterisk installation (as they
were no longer required) to use at home only to find that the fan noise
is too loud. As such I'm looking to sell off this hardware and replace
it with some combination of fanless hardware that will allow me to have
4 handsets and 2 incoming lines.

I don't really want to spend any more than the money I'd make from
selling the T100P / Zplex.

So, is anyone interested in buying this gear / selling some of their old
gear / trading?
Any other suggestions?

Cheers,
Kent
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