date:20070705





http://www.crtc.gc.ca/archive/ENG/Orders/2007/o2007-56.pdf

some discouraging directions being taken by the idiots at the crtc.  
Essentially laying the groundwork to phase out bri completely in  
Canada, probably fcc has similar idiots making similar decisions as we  
discuss this.

Read the rationale for a good laugh - these bureaucrats are out to  
lunch on the technical aspects completely.

So this may all be moot, yeah we can get it to work but if we can't  
buy the service after the next year or so, so what 









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Re: [asterisk-users] North American voice BRI - Informal survey

Quoting Dan Austin <[EMAIL PROTECTED]>:


anyone interested take a read of this listing

http://cgi.ebay.com/4x-BRI-ISDN-2B-D-Asterisk-like-Digium-B410P-VOIP_W0QQitemZ180135963254QQihZ008QQcategoryZ61841QQrdZ1QQcmdZViewItem


I already did ask for details and mentioned that this thread was underway.





> David Wrote:
>> On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
>>> David Boyd wrote:
>>> >
>>> > I seem to remember that the wan Pipeline units supported BRI, and
> also
>>> > provided a couple of analog phone jacks.  I will dig around in the
>>> > basement and try to find the one that I had, if I find it, who
> wants it
>>> > for play?
>>>
>>> Well, whoever ends up with the simulator should get it.
>>>
>>> I'm not familiar with the Pipeline stuff. Got a link you can share?
>>>
>>> -Stephen-
>
>
>> No link, it was something I used 8+ years ago, so I am surprised i
>> pulled it out of my memory :)  I will dig around this weekend and see
> if
>> I can find it. Pretty easy to setup, used it for an ISP connection for
>> centrex purposes. Hopefully I am not mis-remembering it capabilities.
>
> Ascend Pipeline 50/75 units were great remote access devices long
> before ADSL killed em off.  Yes they could handle voice, with one
> or two FXS ports, and one to three BRI ports.
>
> I think I only recently threw away the units I had in the closet and
> maybe even the ones at work.
>
> Setup was not hard, at least the ISDN bits.  We still use BRIs for our
> VC systems, so they can be ordered at least for businesses.  I've also
> used BRI lines to setup small offices on out CCM installation (mostly
> outside of the U.S.)
>
> The hardest part of setting one up is getting the carrier to provide
> the provisioning details (switchtype, SPID or no SPID, PTP or MP).  If
> you can get those details, it is no harder to setup than a PRI.
>
> Dan
>
>
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Re: [asterisk-users] How many number of parallel calls can make through asterisk

2007-07-05 Thread Dimitri Volski

Hi,

You can see some sample configurations from the link below
http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+recommendations

It really depends on the hardware, the codecs, what you are going to use 
it for, etc,etc.

I, for example, have a Pentium IV 2.4Ghz, 2Gb RAM, using G729 codec and 
putting through up to 15 simultaneous calls, with CPU load under 10%.

Are you going to be connecting your Asterisk through phone lines or 
through a VoIP provider?

If it  is VoIP provider then you would need to consider your network 
path to the provider - bandwidth, latency, loads.

Dimitri

Santosh S Kumar wrote:
> Hi,
>
> We are planning to develop a product making asterisk as base, I love that
> asterisk is open source and eager to start working on it. But before 
> even we
> get into start working on asterisk we want to know how many number of
> parallel calls can be made from a single asterisk box, considering we
> install the latest stable version of asterisk (we are ready to buy the
> enterprise version if there is any) on a highly configured box.  So, how
> many number of parallel calls can we make through asterisk??
>
> Regards,
> 
>
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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Dan Austin

David Wrote:
> On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
>> David Boyd wrote:
>> > 
>> > I seem to remember that the wan Pipeline units supported BRI, and
also
>> > provided a couple of analog phone jacks.  I will dig around in the
>> > basement and try to find the one that I had, if I find it, who
wants it
>> > for play?
>> 
>> Well, whoever ends up with the simulator should get it.
>> 
>> I'm not familiar with the Pipeline stuff. Got a link you can share?
>> 
>> -Stephen-

> No link, it was something I used 8+ years ago, so I am surprised i
> pulled it out of my memory :)  I will dig around this weekend and see
if
> I can find it. Pretty easy to setup, used it for an ISP connection for
> centrex purposes. Hopefully I am not mis-remembering it capabilities.

Ascend Pipeline 50/75 units were great remote access devices long
before ADSL killed em off.  Yes they could handle voice, with one
or two FXS ports, and one to three BRI ports.

I think I only recently threw away the units I had in the closet and
maybe even the ones at work.

Setup was not hard, at least the ISDN bits.  We still use BRIs for our
VC systems, so they can be ordered at least for businesses.  I've also
used BRI lines to setup small offices on out CCM installation (mostly
outside of the U.S.)

The hardest part of setting one up is getting the carrier to provide
the provisioning details (switchtype, SPID or no SPID, PTP or MP).  If
you can get those details, it is no harder to setup than a PRI.

Dan

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Re: [asterisk-users] Configuring BLF or Asterisk presence/Hints feature

2007-07-05 Thread Paul Hales


Is the 'buddy' soft button on the bottom of the screen?

PaulH

On Tue, 2007-07-03 at 21:43 +1000, Farooq Ahmed wrote:
> Hi all,
> 
> I am working on 
> 
> asterisk 1.2.18
> zaptel 1.2.17
> Polycom 650
> polycom 430
> SIP version 2.0.3.0131 for IP 650
> SIP version for IP430 2.0.3.0127 
> freepbx 2.2.1
> 
> I am trying to configure BLF using asterisk but failed. I would be thankfull 
> if somebody help me.
> Regards
> FArooq
> 
> **
> 1
> **
> in my extension_additional.conf
> [ext-local]
> include => ext-local-custom
> exten => 501,1,Macro(exten-vm,501,501)
> exten => 501,n,Hangup
> exten => 501,hint,SIP/501
> exten => ${VM_PREFIX}501,1,Macro(vm,501,DIRECTDIAL)
> exten => ${VM_PREFIX}501,n,Hangup
> exten => 502,1,Macro(exten-vm,502,502)
> exten => 502,n,Hangup
> exten => 502,hint,SIP/502
> exten => ${VM_PREFIX}502,1,Macro(vm,502,DIRECTDIAL)
> exten => ${VM_PREFIX}502,n,Hangup
> exten => 503,1,Macro(exten-vm,503,503)
> exten => 503,n,Hangup
> exten => 503,hint,SIP/503
> exten => ${VM_PREFIX}503,1,Macro(vm,503,DIRECTDIAL)
> exten => ${VM_PREFIX}503,n,Hangup
> ; end of [ext-local]
> 
> ***
> 2
> **
> SIP_additional.conf
> one of my extension is configured as
> -- 
> [507]
> type=friend
> secret=1234
> record_out=Adhoc
> record_in=Adhoc
> qualify=yes
> port=5060
> nat=yes
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=rfc2833
> dial=SIP/507
> context=from-internal
> canreinvite=no
> subscribecontext = ext-local
> notifyringing = yes
> callerid=device <507>
> 
> 
> 3
> 
> ext 501 phone is configured with complete contact directory.
> Buddywatch was enabled in the polycom contact directory
> using config like below
> 
>  
> Doe
>  John 
> 507 
> 1 
> 1 
>  
> 0
>  0 
> 1 
> 0 
>  
> 
> **
> Results
> ***
> localhost*CLI> show hints
> localhost*CLI>
> -= Registered Asterisk Dial Plan Hints =-
>507 : SIP/507   State:Unavailable Watchers 
>  0
>506 : SIP/506   State:Unavailable Watchers 
>  0
>505 : SIP/505   State:Unavailable Watchers 
>  0
>504 : SIP/504   State:IdleWatchers 
>  0
>503 : SIP/503   State:Unavailable Watchers 
>  0
>502 : SIP/502   State:IdleWatchers 
>  0
>501 : SIP/501   State:IdleWatchers 
>  0
> 
> - 7 hints registered
> localhost*CLI>
> localhost*CLI> sip show subscriptions
> Peer UserCall ID  ExtensionLast state Type
> 0 active SIP subscriptions
> localhost*CLI>


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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread David Boyd

On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
> David Boyd wrote:
> > 
> > I seem to remember that the wan Pipeline units supported BRI, and also
> > provided a couple of analog phone jacks.  I will dig around in the
> > basement and try to find the one that I had, if I find it, who wants it
> > for play?
> 
> Well, whoever ends up with the simulator should get it.
> 
> I'm not familiar with the Pipeline stuff. Got a link you can share?
> 
> -Stephen-


No link, it was something I used 8+ years ago, so I am surprised i
pulled it out of my memory :)  I will dig around this weekend and see if
I can find it. Pretty easy to setup, used it for an ISP connection for
centrex purposes. Hopefully I am not mis-remembering it capabilities.

Dave


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Re: [asterisk-users] Asterisk E1 card support Q.SIG

2007-07-05 Thread Andrew Joakimsen


I highly recommend the Sangoma cards. They have good support for Asterisk
also for other systems as well :) Asterisk does support Q.SIG that is not an
issue.

On 7/5/07, satish patel <[EMAIL PROTECTED]> wrote:


Dear all
I have asterisk 1.2 and now i want to install E1 card with
support Q.SIG  singaling so which E1 card is best for my setup i need
single port E1/PRI card which support Q.SIG


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Re: [asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)

Hi Bilal -

>> In other words, how can I let the ued codec for the IP
>> Trunk between my Asterisk and the other IP PBX to be
>> g729 and not g711? Ofcourse, I am assuming that the
>> other side also supporting g729.
>>
>You can have multiple allow lines, i.e.
>
>allow=g729
>allow=ulaw

Be sure to also put in a disallow statement first:

disallow=all
allow=g729


- Noah

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[asterisk-users] Slow list

2007-07-05 Thread Doug Lytle

>>stand s a large probability that the list server is trying that address 
>>first.

We'll test your theory, I don't get anything but spam on my second 
server, so I've had it shut down for a good portion of this year.  I 
just went and started it back up.  If this was the case though, I'd 
expect the same delay for the dev list, which I don't have.

Doug



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Re: [asterisk-users] North American voice BRI - Informal survey

Quoting Stephen Bosch <[EMAIL PROTECTED]>:

> Jon Pounder wrote:
>> I have a bunch of old cisco stuff with BRI ports on it but it was
>> never meant for voice, just purely data, so I don't think its very
>> useful for this purpose, but some of the basic signalling could
>> probably be tested with it.
>>
>> is exploring some sort of back to back card setup worth looking into
>> without committing to a line ? Then at least if a pair of cards can
>> talk in the "new format" there is a better chance of them working on a
>> real line. This would also have the advantage of being able to see
>> both ends of the line for debugging purposes, or put the line in the
>> euro mode which should work out of the box just to make sure the
>> hardware setup is all valid before making changes.
>
> Maybe, but it will probably mean writing another driver just to provide
> telco-side signalling -- or is it the same on each end?
>
> What's the deal with PRI cards? Can you run those back-to-back?

t1 in general yes, not so sure about the isdn side of things, hardware  
wise all you need is to cross the pairs, but not familiar enough with  
signalling to know.

BRI isdn has been a while since I looked at the hardware aspect at  
all, I think you can run them back to back similar to the PRI/t1 - the  
raw line is not really BRI proper, its something else similar to the  
hdsl used on t1's and the network terminator makes this into an s/t or  
U interface, the s/t being a bus sort of setup where you can have more  
than 2 devices present ( at this point is actually 2pr just like a t1  
is) some cards though have the nt1's in them so you go right from raw  
copper to card internals so not sure how that all works out. Like I  
said its been a while since I touched this stuff first hand and I may  
be forgetting something or may be mistaken on part of it.

>
> -Stephen-
>
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Re: [asterisk-users] Call Screening Not Working

2007-07-05 Thread Mojo with Horan & Company, LLC

No idea if this is where your problems are coming from, but change:
   exten => s,n,Set(TIMEOUT(response=5))
to
   exten => s,n,Set(TIMEOUT(response)=5)

(the parenthesis moved a bit)


Peder @ NetworkOblivion wrote:
> I am using the Find-me/Follow-me example below with screening:
> 
> http://www.voip-info.org/wiki/view/Asterisk+tips+findme
> 
> Here is my actual config:
> 
> [macro-screen]
> exten => s,1,Wait(1)
> exten => s,n,Background(press-1-to-be-connected-to-the-caller)
> exten => s,n,Set(TIMEOUT(response=5))
> exten => 1,1,NoOp(Caller accepted)
> exten => i,1,Set(MACRO_RESULT=CONTINUE)
> exten => t,1,Set(MACRO_RESULT=CONTINUE)
> 
> [default]
> exten => office,1,Dial(SIP/609,30,M(screen))
> exten => office,2,Hangup
> 
> exten => mobile,1,Dial(SIP/608,30,M(screen))
> exten => mobile,2,Hangup
> 
> exten => 6084,1,NoOp
> exten => 6084,2,SetMusicOnHold(default)
> exten => 6084,3,Dial(LOCAL/office&LOCAL/mobile,40,m)
> 
> 
> 
> 
> I am running 1.4.5.  When I call the number, it rings the phones and 
> plays the message, but no matter what I do, the call gets bridged.  If I 
> hit 2, or nothing, or it times out, the call gets bridged to whoever 
> picks it up.  The script should continue with the other called numbers 
> until the timeout, but it doesn't seem to work that way.  Any ideas what 
> is wrong?  My guess is that something changed in 1.4 to make this fail, 
> but I don't really know what.
> 
> 
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Re: [asterisk-users] North American voice BRI - Informal survey

David Boyd wrote:
> 
> I seem to remember that the wan Pipeline units supported BRI, and also
> provided a couple of analog phone jacks.  I will dig around in the
> basement and try to find the one that I had, if I find it, who wants it
> for play?

Well, whoever ends up with the simulator should get it.

I'm not familiar with the Pipeline stuff. Got a link you can share?

-Stephen-

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Re: [asterisk-users] North American voice BRI - Informal survey

Jeff Davis wrote:
> I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the 
> line simulator and a phone would work. Then get a BRI line when there's 
> a driver that looks like it works.

You'd think it would -- otherwise the line simulator is somewhat
pointless, isn't it?

> I saw another, although more expensive, line simulator at:
> http://vconsole.com/client/?page=simulators&p_id=7
> 
> But it's current production and supported by the manufacturer.

I think currency is really important here. You don't want to be building
something against an old implementation only to find that the result
doesn't work with anything in actual deployment.

So -- I like that simulator. We could probably scrape together the funds
to do it.

Who's got driver coding experience?

-Stephen-

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Re: [asterisk-users] Missing TRANSFER event in queue log when using Local Channels

2007-07-05 Thread Anthony Francis

James FitzGibbon wrote:
> Has anyone observed a problem where using Local channels with 
> AddQueueMember results in missing TRANSFER events?
>
> Right now I'm using straight SIP channels when I call 
> AddQueueMember().  I'm contemplating moving to Local channels because 
> the non-state-based wrapuptime blows when you have a channel in 
> multiple queues (they can hang up and get a call immediately so long 
> as it's from a different queue).  My grand plan is to use the 'h' 
> extension in the context where app_queue calls my agents to invoke 
> PauseQueueMember instead.
>
> The problem is with the /n suffix to the channel name.  With it, I 
> lose TRANSFER events.  Without it, the 'h' extension gets invoked as 
> soon as the call is bridged to the agent.
>
> My agent context looks like this:
>
> [agents]
> exten => 491,1,Dial(SIP/491,20)
> exten => h,1,PauseQueueMember(|${CUT(CHANNEL,,1)})
>
> When I do something along the lines of:
>
> AddQueueMember(queuename,Local/[EMAIL PROTECTED])
>
> Then as soon as the call is bridged, my 'h' extension gets run:
>
> -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2", 
> "SIP/491") in new stack
> -- Called 491
> -- SIP/491-00aa22d0 is ringing
> -- Local/[EMAIL PROTECTED],1 is ringing
> -- SIP/491-00aa22d0 answered Local/[EMAIL PROTECTED],2
> -- Local/[EMAIL PROTECTED],1 answered SIP/427-9d849a90
>   == Spawn extension (agents, 491, 1) exited non-zero on 
> 'Local/[EMAIL PROTECTED],2'
> -- Executing [ [EMAIL PROTECTED]:1] 
> PauseQueueMember("Local/[EMAIL PROTECTED],2", "|Local/[EMAIL PROTECTED]") in 
> new stack
> -- Stopped music on hold on SIP/427-9d849a90
>
> Once the call is bridged, I transfer to  from the agent 
> softphone.  This is what queue log looks like for this type of call:
>
> 1183671934|1183671934.5745|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427
> 1183671937|NONE|NONE|Local/[EMAIL PROTECTED]/n|PAUSEALL|
> 1183671940|1183671934.5745|emerg_nccc_ld_ts|Local/[EMAIL 
> PROTECTED]|CONNECT|6|1183671934.5746
> 1183672005|1183671934.5745|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED] 
> |TRANSFER||from-somecontext|6|65
>
> If I use /n when adding the channel to the queue:
>
> AddQueueMember(queuename,Local/[EMAIL PROTECTED]/n)
>
> Then my 'h' extension is not executed until the bridged call is 
> actually over.  I do the same transfer, but it doesn't show up in the 
> queue log - the call appears to have been terminated by the caller.
>
> 1183672119|1183672119.5839|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427
> 1183672124|1183672119.5839|emerg_nccc_ld_ts|Local/[EMAIL 
> PROTECTED]/n|CONNECT|5|1183672119.5840
> 1183672135|1183672119.5839|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED] 
> /n|COMPLETECALLER|5|11|1
>
> Any ideas?  I need the Pause-on-agent-hangup behaviour (or something 
> like it, short of adding proper wrapup state to app_queue), but I 
> can't lose visibility of my transfers (especially not after I just 
> introduced the sales people to them after never having visibility of 
> this stat on a Nortel BCM)
>
> Much appreciated.
>
> -- 
> j.
> 
>
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I would suggest using the AMI to watch for the complete events and have 
your ami watcher fire in the pause, that way you could use actual device 
names or even dynamic members.

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Re: [asterisk-users] North American voice BRI - Informal survey

Jon Pounder wrote:
> I have a bunch of old cisco stuff with BRI ports on it but it was  
> never meant for voice, just purely data, so I don't think its very  
> useful for this purpose, but some of the basic signalling could  
> probably be tested with it.
> 
> is exploring some sort of back to back card setup worth looking into  
> without committing to a line ? Then at least if a pair of cards can  
> talk in the "new format" there is a better chance of them working on a  
> real line. This would also have the advantage of being able to see  
> both ends of the line for debugging purposes, or put the line in the  
> euro mode which should work out of the box just to make sure the  
> hardware setup is all valid before making changes.

Maybe, but it will probably mean writing another driver just to provide
telco-side signalling -- or is it the same on each end?

What's the deal with PRI cards? Can you run those back-to-back?

-Stephen-

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Re: [asterisk-users] Slow list

2007-07-05 Thread Anthony Francis

Doug Lytle wrote:
>>> Before poking Digium too much, I would look at exactly what YOUR mail
>>> servers are doing that may potentially be the real cause of the delays.
>>>   
>
>
> Already did that.  I use ASSP for filtering.  Digium and associated 
> mailing lists are white listed.  There was only 1 attempt for deliver 
> and there were no delays.  I subscribe to 10 mailing lists (Including 
> the dev list) and they are not having issues.
>
> By the way, the only reason I'm able to respond to your messages and I'm 
> watching the archives at lists.digium.com
>
>
> Doug
>
>   
You have two servers in your MX records.
drdos.info. 60  IN  MX  10 smtp.drdos.info.
drdos.info. 60  IN  MX  5 drdos.info.

The one weighted 10 refuses smtp connections, while it is the higher 
weight it is the first one that was listed when I did the dig. SO there 
stand s a large probability that the list server is trying that address 
first.

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Re: [asterisk-users] North American voice BRI - Informal survey

Dave Donovan wrote:
> On 7/5/07, *Stephen Bosch* <[EMAIL PROTECTED]
> > wrote:
> 
> > I would be willing to help out with a driver, but without a line and
> > card I am not sure how productive that would be.
> 
> As I've already said, I can get one, and it's not a big deal, so
> I'll be
> the test case.
> 
> -Stephen-
> 
> 
> Ok, how about this for a scenario:  We could order a BRI from Bell at
> Stephen's location.  We could get a Sangoma card (or some other piece of
> hardware that we determine) and put it in some vanilla box.  I guess it
> would be best to isolate that from Stephen's production network and then
> just let the developers tunnel into it by some means.
> 
> I'd be willing to bend some ears and see if Sangoma would put up the card.

Okay.

> I wonder if we need piece of reference hardware.  Imagine this scenario,
> we order the circuit and Bell wants to turn it up.  What do we turn up
> with?  It would be great if we had something that we could plug into it
> and get that working just to establish that the line is delivered
> correctly and functioning.
> 
> Another option is to buy something that would provide BRI service on
> site.  I'm thinking maybe an old PBX system or something.  I'm not sure
> what this stuff would be worth.  I saw this simulator on Ebay but I
> don't know enough to say whether it would be suitable for testing voice
> services:
> http://cgi.ebay.ca/Lucent-ISDN-simulator-8-BRI-4-T1-NICE-Document-CD_W0QQitemZ120136665085QQihZ002QQcategoryZ51268QQrdZ1QQcmdZViewItem
> 
> 
> I'm not in the biz but I'd be willing to contribute some $$$ toward the
> project.  I'll talk to some other folks in the Toronto Asterisk User
> Group. I know there are quite a few consultants/installers in the group
> and maybe some will see the benefit of supporting this kind of work.
> 
> Has anyone got a PBX with spare BRI ports in it?  Maybe that's a cheap
> way to get started.  We could just hook a box up to that and work out
> some of the early stage stuff.  I know that people with Polycom (and
> other) video/teleconferencing equipment often have BRI cards in their
> Nortel PBX or Avaya gear.
> 
> It looks like we have the nucleus of a project here.  Let's keep the
> ball rolling.

Brilliant, Dave.

I'm inclined to say that a real-world service from the telco is
important for testing, but it's also true that there would be a lot of
messing around in the beginning, so the first few months of service
would be a bit of a waste. If we can simulate the environment, that
would be wonderful -- as long as it can be done with current equipment.
I wouldn't want to be solving against hardware running an older standard.

I would also be interested in contributing to such a project. The caveat
is that I'm not a programmer -- we will need some programming talent on
this to make it succeed.

-Stephen-

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[asterisk-users] Missing TRANSFER event in queue log when using Local Channels

2007-07-05 Thread James FitzGibbon


Has anyone observed a problem where using Local channels with AddQueueMember
results in missing TRANSFER events?

Right now I'm using straight SIP channels when I call AddQueueMember().  I'm
contemplating moving to Local channels because the non-state-based
wrapuptime blows when you have a channel in multiple queues (they can hang
up and get a call immediately so long as it's from a different queue).  My
grand plan is to use the 'h' extension in the context where app_queue calls
my agents to invoke PauseQueueMember instead.

The problem is with the /n suffix to the channel name.  With it, I lose
TRANSFER events.  Without it, the 'h' extension gets invoked as soon as the
call is bridged to the agent.

My agent context looks like this:

[agents]
exten => 491,1,Dial(SIP/491,20)
exten => h,1,PauseQueueMember(|${CUT(CHANNEL,,1)})

When I do something along the lines of:

AddQueueMember(queuename,Local/[EMAIL PROTECTED])

Then as soon as the call is bridged, my 'h' extension gets run:

   -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2", 
"SIP/491")
in new stack
   -- Called 491
   -- SIP/491-00aa22d0 is ringing
   -- Local/[EMAIL PROTECTED],1 is ringing
   -- SIP/491-00aa22d0 answered Local/[EMAIL PROTECTED],2
   -- Local/[EMAIL PROTECTED],1 answered SIP/427-9d849a90
 == Spawn extension (agents, 491, 1) exited non-zero on '
Local/[EMAIL PROTECTED],2'
   -- Executing [EMAIL PROTECTED]:1] PauseQueueMember("Local/[EMAIL 
PROTECTED],2",
"|Local/[EMAIL PROTECTED]") in new stack
   -- Stopped music on hold on SIP/427-9d849a90

Once the call is bridged, I transfer to  from the agent softphone.  This
is what queue log looks like for this type of call:

1183671934|1183671934.5745|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427
1183671937|NONE|NONE|Local/[EMAIL PROTECTED]/n|PAUSEALL|
1183671940|1183671934.5745|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
|CONNECT|6|1183671934.5746
1183672005|1183671934.5745|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
|TRANSFER||from-somecontext|6|65

If I use /n when adding the channel to the queue:

AddQueueMember(queuename,Local/[EMAIL PROTECTED]/n)

Then my 'h' extension is not executed until the bridged call is actually
over.  I do the same transfer, but it doesn't show up in the queue log - the
call appears to have been terminated by the caller.

1183672119|1183672119.5839|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427
1183672124|1183672119.5839|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
/n|CONNECT|5|1183672119.5840
1183672135|1183672119.5839|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
/n|COMPLETECALLER|5|11|1

Any ideas?  I need the Pause-on-agent-hangup behaviour (or something like
it, short of adding proper wrapup state to app_queue), but I can't lose
visibility of my transfers (especially not after I just introduced the sales
people to them after never having visibility of this stat on a Nortel BCM)

Much appreciated.

--
j.
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[asterisk-users] sounds

2007-07-05 Thread Hans Witvliet

Just curious,

I noticed that with SetLanguage() you can change it into a lot of other
languages. Yes one can record them easily enough with "record", but
don't like to re-invent the wheel..

Browsed through a lot of google-pages but failed to find any other
languages (except for FR and ES on the digium site)

Any pointers for german, dutch, greek, italian,  prompts?

Hans
-- 
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Re: [asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)

2007-07-05 Thread Alex Balashov


Bilal,

   There are allow= options you can use on the peers in sip.conf to define
what codec capability Asterisk advertises toward them, and therefore, what
the negotiation on both call legs will ultimately settle upon.  When those
legs are bridged -- as Asterisk does unto them -- they will be of a codec
that is within that bounded set.

   You can have multiple allow lines, i.e.

   allow=g729
   allow=ulaw

   ... or just one.

   Does that help?

-- Alex

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)

2007-07-05 Thread bilal ghayyad

Dear Alex;

I am asking about:

What is the configuration that I can do it to let the
traffic between the two Asterisk PBX and another IP BX
to be g729 or G711 or g723?

In other words, how can I let the ued codec for the IP
Trunk between my Asterisk and the other IP PBX to be
g729 and not g711? Ofcourse, I am assuming that the
other side also supporting g729.

Regards
Bilal 

> Where I determine the codec to be used for the SIP
Trunk (between 
> Asterik and another SIP softswitch)?

   Are you asking positively how to determine which
codec is being 
negotiated between those two elements, or,
normatively, which one is
best to use?

   If the former question, you can look at the SDP
(Session Description
Protocol) payload in the INVITEs (and other messages
part of the INVITE
transaction).  Find the 'rtpmap' lines.  They look
like this:

 a=rtpmap:18 G729/8000

   There may be multiple such lines indicating that
endpoint's support
for all of them.  If so, the only way to determine
which one is
 actually
going to be used is to look at the RTP stream with a
protocol analyser.
It should be able to tell you.  Wireshark/Ethereal
certainly can.

   If you're asking which codec should be used, it
depends on the
 desired 
application, whether the trunk is running over a LAN,
WAN, or over the 
public Internet, whether QoS is involved, etc.



 

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Re: [asterisk-users] Call Queues

2007-07-05 Thread C. Chad Wallace

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Floyd wrote:
> Hi everyone:
> 
> I've searching for a while and haven't found what i
> need.
> The thing is that i have a tdm422p with the two fxo
> ports connected to the pstn. I want my sip users to be
> able to call other numbers(any number) in the pstn
> through my zap fxo channels. I have a big number of
> sip users so as you can imagine there will be
> congestion when some of them(more than two!!) want to
> call outside, that is why i want to be able to put
> those outgoing calls in a queue. For example if i want
> to call someone in the pstn and the fxo port is
> already in use, i want to be placed in a queue and
> when the channel is free my call is routed to the
> aproppiated destination. As far as i have read the
> queues are not for this kind of stuffs,  there are
> just agents or extensions that attend the calls in the
> queue and nothing more. am i wrong???
> Any help will be useful. 
> thanks in advance!!

You could probably do this using the Local channel.  You'd create a
context, say outbound, to take calls from the queue and connect them to
a Zap channel, with 2 extensions in that context--one for each channel.
 Then you add each of those extensions as members of the queue:

member => Local/[EMAIL PROTECTED]/n
member => Local/[EMAIL PROTECTED]/n

Make sure your dialplan in outbound returns Busy if the Zap channel is busy.

The tricky part would be passing the dialed number through...  But if
you set an inheriting channel var, it should go through the queue and
into the Local channel to your outbound extension.

Sorry I don't have any code for you... I haven't done it yet; I'm just
putting the idea out there.

Hope this helps!
Good luck.

- --

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0

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Re: [asterisk-users] Call Queues

2007-07-05 Thread Rob Schall


Noah Miller wrote:

Hi Eve -

  

The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of them(more than two!!) want to
call outside, that is why i want to be able to put
those outgoing calls in a queue. For example if i want
to call someone in the pstn and the fxo port is
already in use, i want to be placed in a queue and
when the channel is free my call is routed to the
aproppiated destination. As far as i have read the
queues are not for this kind of stuffs,  there are
just agents or extensions that attend the calls in the
queue and nothing more. am i wrong???



I think your suspicions may be correct.  You could add your ZAP
channels as members in queues.conf, maybe something like this: members
=> ZAP/1, and then use queue() on your outbound extensions.  The
problem is how will your agents, in this case your ZAP trunks, know to
"pick up the line" when they are not busy.  You'd have to get these
lines to somehow go offhook if they're not already busy.  Maybe you
can do this with an AGI script.  I don't know, I've never tried to
artificially control hook status.

Personally, I'd probably just skip the whole queue idea and get some
cheap SIP or IAX trunks and fall back to them when the ZAP lines are
busy.


- Noah

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Possibly do a combination of things. Check if those zap chans are in 
use/busy. If they care, then create a call file using a script. I 
haven't played too much with it, so I don't know if those will queue 
until they can complete or if it will just error and delete itself. If 
you really are determined, you might even be able to route all requests 
to a script. Then have it check if there are any open lines... if so, 
create the call file... if not, then put it in a queue (in python, 
etc... not an asterisk queue), and try again in a min and see if a 
channel has opened up.


Disclaimer - I have no idea if this idea will work. :)
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Re: [asterisk-users] Call Queues

Hi Eve -

> The thing is that i have a tdm422p with the two fxo
> ports connected to the pstn. I want my sip users to be
> able to call other numbers(any number) in the pstn
> through my zap fxo channels. I have a big number of
> sip users so as you can imagine there will be
> congestion when some of them(more than two!!) want to
> call outside, that is why i want to be able to put
> those outgoing calls in a queue. For example if i want
> to call someone in the pstn and the fxo port is
> already in use, i want to be placed in a queue and
> when the channel is free my call is routed to the
> aproppiated destination. As far as i have read the
> queues are not for this kind of stuffs,  there are
> just agents or extensions that attend the calls in the
> queue and nothing more. am i wrong???

I think your suspicions may be correct.  You could add your ZAP
channels as members in queues.conf, maybe something like this: members
=> ZAP/1, and then use queue() on your outbound extensions.  The
problem is how will your agents, in this case your ZAP trunks, know to
"pick up the line" when they are not busy.  You'd have to get these
lines to somehow go offhook if they're not already busy.  Maybe you
can do this with an AGI script.  I don't know, I've never tried to
artificially control hook status.

Personally, I'd probably just skip the whole queue idea and get some
cheap SIP or IAX trunks and fall back to them when the ZAP lines are
busy.


- Noah

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Re: [asterisk-users] Asterisk E1 card support Q.SIG

2007-07-05 Thread Matthew Fredrickson

satish patel wrote:
> Dear all
> I have asterisk 1.2 and now i want to install E1 card 
> with support Q.SIG  singaling so which E1 card is best for my setup i 
> need single port E1/PRI card which support Q.SIG
All T1/E1 cards using libpri have basic Q.SIG support.  I recommend the 
Digium cards, of course :-)

---
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread David Boyd

On Thu, 2007-07-05 at 15:08 -0400, Jon Pounder wrote:
> Quoting Jeff Davis <[EMAIL PROTECTED]>:
> 
> > Jon Pounder wrote:
> >> I have a bunch of old cisco stuff with BRI ports on it but it was
> >> never meant for voice, just purely data, so I don't think its very
> >> useful for this purpose, but some of the basic signalling could
> >> probably be tested with it.
> >>
> >> is exploring some sort of back to back card setup worth looking into
> >> without committing to a line ? Then at least if a pair of cards can
> >> talk in the "new format" there is a better chance of them working on a
> >> real line. This would also have the advantage of being able to see
> >> both ends of the line for debugging purposes, or put the line in the
> >> euro mode which should work out of the box just to make sure the
> >> hardware setup is all valid before making changes.
> >
> >
> > I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the
> > line simulator and a phone would work. Then get a BRI line when there's
> > a driver that looks like it works.
> 
> the signalling on the line simulator and phone would have to be  
> compatible - phone and card are going to have the same issues with  
> supporting the northamerican signalling.
> 
> probably depend mainly what country the hardware is being sold from  
> what the odds of working are.
SNIP



I seem to remember that the wan Pipeline units supported BRI, and also
provided a couple of analog phone jacks.  I will dig around in the
basement and try to find the one that I had, if I find it, who wants it
for play?

Dave


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Re: [asterisk-users] Call Screening Not Working

2007-07-05 Thread [EMAIL PROTECTED]

I know there is FMFM in 1.4, but I want to know why the macro isn't working.  I 
added a NoOp and Wait to the macro as lines 4 and 5 and neither gets executed.  
As soon as I hit any number such as 2, 3 or 4, I immediately get bridged to the 
call.  I may be wrong, but I'm pretty sure that shouldn't happen.


- Original Message -
From: "Bobby Crawford" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, July 5, 2007 2:27:05 PM (GMT-0600) America/Mexico_City
Subject: Re: [asterisk-users] Call Screening Not Working

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion
> Sent: Thursday, July 05, 2007 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Call Screening Not Working
> 
> I am using the Find-me/Follow-me example below with screening:
> 
> http://www.voip-info.org/wiki/view/Asterisk+tips+findme
> 
> Here is my actual config:
> 
> [macro-screen]
> exten => s,1,Wait(1)
> exten => s,n,Background(press-1-to-be-connected-to-the-caller)
> exten => s,n,Set(TIMEOUT(response=5))
> exten => 1,1,NoOp(Caller accepted)
> exten => i,1,Set(MACRO_RESULT=CONTINUE)
> exten => t,1,Set(MACRO_RESULT=CONTINUE)
> 
> [default]
> exten => office,1,Dial(SIP/609,30,M(screen))
> exten => office,2,Hangup
> 
> exten => mobile,1,Dial(SIP/608,30,M(screen))
> exten => mobile,2,Hangup
> 
> exten => 6084,1,NoOp
> exten => 6084,2,SetMusicOnHold(default)
> exten => 6084,3,Dial(LOCAL/office&LOCAL/mobile,40,m)
> 
> 
> 
> 
> I am running 1.4.5.  When I call the number, it rings the phones and
> plays the message, but no matter what I do, the call gets bridged.  If I
> hit 2, or nothing, or it times out, the call gets bridged to whoever
> picks it up.  The script should continue with the other called numbers
> until the timeout, but it doesn't seem to work that way.  Any ideas what
> is wrong?  My guess is that something changed in 1.4 to make this fail,
> but I don't really know what.
> 
> 
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Starting with 1.4, there is a built-in FollowMe application.  You can find
some docs on voip-info.org on how to use it.  Hopefully that fixes your
problem here.

Bobby


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[asterisk-users] Call Queues

2007-07-05 Thread Floyd

Hi everyone:

I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of them(more than two!!) want to
call outside, that is why i want to be able to put
those outgoing calls in a queue. For example if i want
to call someone in the pstn and the fxo port is
already in use, i want to be placed in a queue and
when the channel is free my call is routed to the
aproppiated destination. As far as i have read the
queues are not for this kind of stuffs,  there are
just agents or extensions that attend the calls in the
queue and nothing more. am i wrong???
Any help will be useful. 
thanks in advance!!

eve

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Re: [asterisk-users] Call Screening Not Working

2007-07-05 Thread Bobby Crawford

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion
> Sent: Thursday, July 05, 2007 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Call Screening Not Working
> 
> I am using the Find-me/Follow-me example below with screening:
> 
> http://www.voip-info.org/wiki/view/Asterisk+tips+findme
> 
> Here is my actual config:
> 
> [macro-screen]
> exten => s,1,Wait(1)
> exten => s,n,Background(press-1-to-be-connected-to-the-caller)
> exten => s,n,Set(TIMEOUT(response=5))
> exten => 1,1,NoOp(Caller accepted)
> exten => i,1,Set(MACRO_RESULT=CONTINUE)
> exten => t,1,Set(MACRO_RESULT=CONTINUE)
> 
> [default]
> exten => office,1,Dial(SIP/609,30,M(screen))
> exten => office,2,Hangup
> 
> exten => mobile,1,Dial(SIP/608,30,M(screen))
> exten => mobile,2,Hangup
> 
> exten => 6084,1,NoOp
> exten => 6084,2,SetMusicOnHold(default)
> exten => 6084,3,Dial(LOCAL/office&LOCAL/mobile,40,m)
> 
> 
> 
> 
> I am running 1.4.5.  When I call the number, it rings the phones and
> plays the message, but no matter what I do, the call gets bridged.  If I
> hit 2, or nothing, or it times out, the call gets bridged to whoever
> picks it up.  The script should continue with the other called numbers
> until the timeout, but it doesn't seem to work that way.  Any ideas what
> is wrong?  My guess is that something changed in 1.4 to make this fail,
> but I don't really know what.
> 
> 
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Starting with 1.4, there is a built-in FollowMe application.  You can find
some docs on voip-info.org on how to use it.  Hopefully that fixes your
problem here.

Bobby


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[asterisk-users] sometimes half audio on 7960

2007-07-05 Thread Jerry Geis

Hi,

I am getting half channel audio on cisco 7960?
Any idea why? Details below.

Jerry


This same phone has been working for MONTHS using a TDM2400p
with no issues.
Today we got a T1 installed coming into Box A with all incoming calls 
going to Box B (the TDM2400P box).
The TDM2400 is no longer in use. All incoming and outgoing calls are 
done with the T1 and Box A and Box B
use SIP between them (will change later).

I have 6 polycom phones that work perfectly in this arrangement. I have 
1 cisco 7960 that gets half channel audio.
The calling person can hear the called person but that is all.

Where do I start looking for something like this?

Again the 7960 has been fine for 2 years using the TDM2400p.

Doesnt seem to matter if calls are coming into the 7960 or if this 
person is calling out. The same half channel exists.

THanks for hte help.

Jerry


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Re: [asterisk-users] North American voice BRI - Informal survey

Quoting Jeff Davis <[EMAIL PROTECTED]>:

> Jon Pounder wrote:
>> I have a bunch of old cisco stuff with BRI ports on it but it was
>> never meant for voice, just purely data, so I don't think its very
>> useful for this purpose, but some of the basic signalling could
>> probably be tested with it.
>>
>> is exploring some sort of back to back card setup worth looking into
>> without committing to a line ? Then at least if a pair of cards can
>> talk in the "new format" there is a better chance of them working on a
>> real line. This would also have the advantage of being able to see
>> both ends of the line for debugging purposes, or put the line in the
>> euro mode which should work out of the box just to make sure the
>> hardware setup is all valid before making changes.
>
>
> I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the
> line simulator and a phone would work. Then get a BRI line when there's
> a driver that looks like it works.

the signalling on the line simulator and phone would have to be  
compatible - phone and card are going to have the same issues with  
supporting the northamerican signalling.

probably depend mainly what country the hardware is being sold from  
what the odds of working are.



>
> I saw another, although more expensive, line simulator at:
> http://vconsole.com/client/?page=simulators&p_id=7
>
> But it's current production and supported by the manufacturer.
>
> --
> Jeff Davis
> Netsource Consulting
> Richmond, VA
>
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Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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Re: [asterisk-users] Caller ID Spoofing to be banned in the USA

2007-07-05 Thread Rob Schall


Dovid B wrote:
You are right but my concerns is the ITSP's may stop allowing it 
because they don't want to get in to trouble. They may request a list 
of all the DID's that I have and limit me setting my CID to the list 
that I gave them.


I doubt this will ever be an issue. The telco companies and certainly 
not one for major changes, nor are they likely to enforce any laws and 
put any time into something they aren't responsible for. My understand 
of this new law is the punish those users who abuse the ability to set 
the CID. If the government was really bored and wanted to waste a lot of 
time and money in court, I guess they could try to take AT&T or 
Broadwing to court, but my guess is they wouldn't get far. It isn't 
their fault that a customer of theirs broke the law, and they haven't 
(and probably won't) be required to keep people from breaking the law. 
Just because you make and sell guns doesn't mean you can go after them 
for using your guns and bullets to kill people. Ya know?


Long story short, unless the government has nothing else to go after, 
I'd think you'll be perfectly fine. Like others have said on this 
subject... Unless you do it to fool somebody and it has a mean spirit to 
it, you should be in the clear. Masking a caller-id to be a cell phone 
of an employee, etc, should be fine, as long as every one knows what is 
going on.


Rob
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Re: [asterisk-users] Slow list

2007-07-05 Thread Andrew Kohlsmith

On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote:
> Already did that.  I use ASSP for filtering.  Digium and associated
> mailing lists are white listed.  There was only 1 attempt for deliver
> and there were no delays.  I subscribe to 10 mailing lists (Including
> the dev list) and they are not having issues.
>
> By the way, the only reason I'm able to respond to your messages and I'm
> watching the archives at lists.digium.com

I am having no issues with Digium's lists.  They get a little laggy at times, 
but generally are fast enough.

-A.

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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Jeff Davis

Jon Pounder wrote:
> I have a bunch of old cisco stuff with BRI ports on it but it was  
> never meant for voice, just purely data, so I don't think its very  
> useful for this purpose, but some of the basic signalling could  
> probably be tested with it.
> 
> is exploring some sort of back to back card setup worth looking into  
> without committing to a line ? Then at least if a pair of cards can  
> talk in the "new format" there is a better chance of them working on a  
> real line. This would also have the advantage of being able to see  
> both ends of the line for debugging purposes, or put the line in the  
> euro mode which should work out of the box just to make sure the  
> hardware setup is all valid before making changes.


I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the 
line simulator and a phone would work. Then get a BRI line when there's 
a driver that looks like it works.

I saw another, although more expensive, line simulator at:
http://vconsole.com/client/?page=simulators&p_id=7

But it's current production and supported by the manufacturer.

--
Jeff Davis
Netsource Consulting
Richmond, VA

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[asterisk-users] Slow list

2007-07-05 Thread Doug Lytle

>> Before poking Digium too much, I would look at exactly what YOUR mail
>> servers are doing that may potentially be the real cause of the delays.


Already did that.  I use ASSP for filtering.  Digium and associated 
mailing lists are white listed.  There was only 1 attempt for deliver 
and there were no delays.  I subscribe to 10 mailing lists (Including 
the dev list) and they are not having issues.

By the way, the only reason I'm able to respond to your messages and I'm 
watching the archives at lists.digium.com


Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] REGEX expression for NXXNXXXXXX?

2007-07-05 Thread Mik Cheez

My mistake...you're correct...should have tested it.

Mojo with Horan & Company, LLC wrote:
> But those are not REGEX expressions, those are asterisk dialplan 
> pattern-matching expressions.  great for the X in:
> 
> exten => _X.,1,blah
> 
> but not for use with REGEX() function.
> 
> I think it would be close to what Michael said, but like this:
> 1{0,1}[2-9][0-9]{2}[2-9][0-9]{6}
> 
> Michael, your expression would be satisfied by "122000" which has a 
> 0 in the first part of the prefix, no good, that one must be within 2-9 
> range also. "1 220 *0*00 "
> 
> 
> Moj
> 
> Noah Miller wrote:
>> Hi Again Brent -
>>
 What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
 NXXNXX?
>>> I think you've got it right already.  What do you need to do?
>> If you wanted to get more specific and identify ONLY NANP, you may
>> have to break it out into more than just one rule:
>>
>> _800NXX
>> _888NXX
>> _877NXX
>> _866NXX
>> 
>>
>> etc.
>>
>> BTW: I just read up and discovered that 855, 844, 833, and 822 are
>> reserved but not yet used.
>>
>>
>> - Noah
>>
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> 


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Re: [asterisk-users] Slow list

2007-07-05 Thread Walt Reed


On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said:
> >>Well, this is now the third active thread on this subject, but I guess
> >>you won't see this message for a while.  Has anyone dissected the
> >>headers of a delayed message yet?  We should be able to tell for sure
> >>where the holdup is.  All of the messages are coming through on time
> >>for me, so it won't do much good for me to look.
> 
> 
> Looks like mail is getting held up between INXS.digium.internal and 
> lists.digium.com
> 
> INXS.digium.internal received it the first of July, lists.digium.com 
> received it on the 4th.
> 
> drdos.info (ME) received it from lists.digium.com on that same day (Today).

What you can't see without looking at the mail server logs on both ends
is delivery attempts. Greylisting for example can totally hose you over
depending on the implementation. Greylisting without whitelisting is
irresponsible.  How many tries did the digium server make before the
message finally got through??? That's what we need to know. Only Digium
can say.

Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.

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[asterisk-users] Call Screening Not Working

2007-07-05 Thread Peder @ NetworkOblivion

I am using the Find-me/Follow-me example below with screening:

http://www.voip-info.org/wiki/view/Asterisk+tips+findme

Here is my actual config:

[macro-screen]
exten => s,1,Wait(1)
exten => s,n,Background(press-1-to-be-connected-to-the-caller)
exten => s,n,Set(TIMEOUT(response=5))
exten => 1,1,NoOp(Caller accepted)
exten => i,1,Set(MACRO_RESULT=CONTINUE)
exten => t,1,Set(MACRO_RESULT=CONTINUE)

[default]
exten => office,1,Dial(SIP/609,30,M(screen))
exten => office,2,Hangup

exten => mobile,1,Dial(SIP/608,30,M(screen))
exten => mobile,2,Hangup

exten => 6084,1,NoOp
exten => 6084,2,SetMusicOnHold(default)
exten => 6084,3,Dial(LOCAL/office&LOCAL/mobile,40,m)




I am running 1.4.5.  When I call the number, it rings the phones and 
plays the message, but no matter what I do, the call gets bridged.  If I 
hit 2, or nothing, or it times out, the call gets bridged to whoever 
picks it up.  The script should continue with the other called numbers 
until the timeout, but it doesn't seem to work that way.  Any ideas what 
is wrong?  My guess is that something changed in 1.4 to make this fail, 
but I don't really know what.


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Re: [asterisk-users] North American voice BRI - Informal survey

Quoting Dave Donovan <[EMAIL PROTECTED]>:

> On 7/5/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
>>
>>> I would be willing to help out with a driver, but without a line and
>>> card I am not sure how productive that would be.
>>
>> As I've already said, I can get one, and it's not a big deal, so I'll be
>> the test case.
>>
>> -Stephen-
>
>
> Ok, how about this for a scenario:  We could order a BRI from Bell at
> Stephen's location.  We could get a Sangoma card (or some other piece of
> hardware that we determine) and put it in some vanilla box.  I guess it
> would be best to isolate that from Stephen's production network and then
> just let the developers tunnel into it by some means.
>
> I'd be willing to bend some ears and see if Sangoma would put up the card.
>
> I wonder if we need piece of reference hardware.  Imagine this scenario, we
> order the circuit and Bell wants to turn it up.  What do we turn up with?
> It would be great if we had something that we could plug into it and get
> that working just to establish that the line is delivered correctly and
> functioning.
>
> Another option is to buy something that would provide BRI service on site.
> I'm thinking maybe an old PBX system or something.  I'm not sure what this
> stuff would be worth.  I saw this simulator on Ebay but I don't know enough
> to say whether it would be suitable for testing voice services:
> http://cgi.ebay.ca/Lucent-ISDN-simulator-8-BRI-4-T1-NICE-Document-CD_W0QQitemZ120136665085QQihZ002QQcategoryZ51268QQrdZ1QQcmdZViewItem
>
> I'm not in the biz but I'd be willing to contribute some $$$ toward the
> project.  I'll talk to some other folks in the Toronto Asterisk User Group.
> I know there are quite a few consultants/installers in the group and maybe
> some will see the benefit of supporting this kind of work.
>
> Has anyone got a PBX with spare BRI ports in it?  Maybe that's a cheap way
> to get started.  We could just hook a box up to that and work out some of
> the early stage stuff.  I know that people with Polycom (and other)
> video/teleconferencing equipment often have BRI cards in their Nortel PBX or
> Avaya gear.

I have a bunch of old cisco stuff with BRI ports on it but it was  
never meant for voice, just purely data, so I don't think its very  
useful for this purpose, but some of the basic signalling could  
probably be tested with it.

is exploring some sort of back to back card setup worth looking into  
without committing to a line ? Then at least if a pair of cards can  
talk in the "new format" there is a better chance of them working on a  
real line. This would also have the advantage of being able to see  
both ends of the line for debugging purposes, or put the line in the  
euro mode which should work out of the box just to make sure the  
hardware setup is all valid before making changes.



>
> It looks like we have the nucleus of a project here.  Let's keep the ball
> rolling.
>
> Dave Donovan



Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Dave Donovan


On 7/5/07, Dave Donovan <[EMAIL PROTECTED]> wrote:


I wonder if we need piece of reference hardware.  Imagine this scenario,
we order the circuit and Bell wants to turn it up.  What do we turn up
with?  It would be great if we had something that we could plug into it and
get that working just to establish that the line is delivered correctly and
functioning.



I just saw a BRI card for a Lucent/Avaya G3 (which would probably still work
in the 8x00 series).  listed on Ebay for <$300. I have some experience with
Avaya switches.  We could save the cost of the line and installation by
putting this card in a switch.  The added benefit is we would control both
sides of the link and could test different configs.

Does anyone in the GTA have a G3 that they'd be willing to put the card
into?  I would be willing to pick up the cost of the card.

Dave
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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Dave Donovan

On 7/5/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:

> I would be willing to help out with a driver, but without a line and
> card I am not sure how productive that would be.

As I've already said, I can get one, and it's not a big deal, so I'll be
the test case.

-Stephen-

Ok, how about this for a scenario: We could order a BRI from Bell at
Stephen's location. We could get a Sangoma card (or some other piece of
hardware that we determine) and put it in some vanilla box. I guess it
would be best to isolate that from Stephen's production network and then
just let the developers tunnel into it by some means.

I'd be willing to bend some ears and see if Sangoma would put up the card.

I wonder if we need piece of reference hardware. Imagine this scenario, we
order the circuit and Bell wants to turn it up. What do we turn up with?
It would be great if we had something that we could plug into it and get
that working just to establish that the line is delivered correctly and
functioning.

Another option is to buy something that would provide BRI service on site.
I'm thinking maybe an old PBX system or something. I'm not sure what this
stuff would be worth. I saw this simulator on Ebay but I don't know enough
to say whether it would be suitable for testing voice services:
http://cgi.ebay.ca/Lucent-ISDN-simulator-8-BRI-4-T1-NICE-Document-CD_W0QQitemZ120136665085QQihZ002QQcategoryZ51268QQrdZ1QQcmdZViewItem

I'm not in the biz but I'd be willing to contribute some $$$ toward the
project. I'll talk to some other folks in the Toronto Asterisk User Group.
I know there are quite a few consultants/installers in the group and maybe
some will see the benefit of supporting this kind of work.

Has anyone got a PBX with spare BRI ports in it? Maybe that's a cheap way
to get started. We could just hook a box up to that and work out some of
the early stage stuff. I know that people with Polycom (and other)
video/teleconferencing equipment often have BRI cards in their Nortel PBX or
Avaya gear.

It looks like we have the nucleus of a project here. Let's keep the ball
rolling.

Dave Donovan
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Re: [asterisk-users] Simple CDRs w/Asterisk/OpenSER.

2007-07-05 Thread Jaswinder Singh


Asterisk is poor with codec negotiation . It does not check if it can avoid
transcoding  by forcing codec available to both sides .. instead it will
read it's config file and will select first allowed codec that  is also
available on other device on each leg of call and happily transcode between
them .There was a patch on digium submitted by someone for asterisk 1.2.12
or so but it isnt updated from long time .  I am sure guys at digium are
aware about it and working on it . It's not  a bug  since asterisk is not a
sip proxy and tries to keep media path through it to offer its pbx features
but it would be a great feature nonetheless if implemented .

On 05/07/07, Alex Balashov <[EMAIL PROTECTED]> wrote:



Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based
proxy / call routing setup?  I need to get simple CDRs;  not for detailed
settlement/rating, but just for reconciliation with an ultimate TDM
carrier just to make sure we only get billed for what we're actually
using.

I'd use the often-heralded approach of dumping a call from OpenSER into
Asterisk and having it bounce right back out toward the proxy by way of
REINVITEs.  I don't want the media running through Asterisk or Asterisk
being a limiting factor in that regard.

The problem is I don't have native G.729 support - we have no need for
it because neither the customer's network elements nor ours lack an
implementation of their own they can negotiate on just fine.  But
unfortunately Asterisk insists on natively homogenising the SDP from
both sides even if it subsequently removes itself from the media path!

So, I end up with situations where on the one side, I get, say:

Customer MGW --> OpenSER --> Asterisk - sends call as G.729.

Asterisk --> OpenSER --> Our MGW - our MGW prefers G.711a.

Now, if customer MGW <-> Our MGW were talking directly, as they do
when the deal is brokered through the OpenSER proxy, they would simply
negotiate upon what they agree.  But for some reason with Asterisk
this does not seem to be working as advertised;  we get lots of failed
calls if we pass them through Asterisk because one leg is one codec
and the other is another.  I am not sure how it arrives at that
conclusion despite the overlap of shared codecs (G.729 on both sides,
I would expect it to pass thru licence-free), and to be honest, I
don't particularly care if it's a bug or a feature, I just need it
not to introduce codec issues if I use it as a billing target.

Any help or insight would be greatly appreciated.

Thanks,

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] IAX-Voicemail

2007-07-05 Thread Andrea Bencini

I have asterisk 1.2.18
Each my extention number has a mailbox; I have SIP extentions and IAX 
extentions.

When  I call an unregister extention number (SIP/IAX) from SIP 
hard/soft-phone, asterisk actives the voicemail (I can put a message in the 
mailbox).

Instead when I call an unregister extention number (SIP/IAX) from IAX 
hard/soft-phone I get an immediate hangup.

Can you help me
thank
Andrea 


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Re: [asterisk-users] REGEX expression for NXXNXXXXXX?

2007-07-05 Thread Mojo with Horan & Company, LLC

But those are not REGEX expressions, those are asterisk dialplan 
pattern-matching expressions.  great for the X in:

exten => _X.,1,blah

but not for use with REGEX() function.

I think it would be close to what Michael said, but like this:
1{0,1}[2-9][0-9]{2}[2-9][0-9]{6}

Michael, your expression would be satisfied by "122000" which has a 
0 in the first part of the prefix, no good, that one must be within 2-9 
range also. "1 220 *0*00 "


Moj

Noah Miller wrote:
> Hi Again Brent -
> 
>>> What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
>>> NXXNXX?
>> I think you've got it right already.  What do you need to do?
> 
> If you wanted to get more specific and identify ONLY NANP, you may
> have to break it out into more than just one rule:
> 
> _800NXX
> _888NXX
> _877NXX
> _866NXX
> 
> 
> etc.
> 
> BTW: I just read up and discovered that 855, 844, 833, and 822 are
> reserved but not yet used.
> 
> 
> - Noah
> 
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[asterisk-users] IAX additional call-data

2007-07-05 Thread Steve Davies

Hi,

Just a quick question. Is there a way when making an IAX call to
transmit some additional call-data, perhaps in a variable? I could
overload callerid-name, but that is nasty and ugly :)

Thanks for any suggestions.

Regards,
Steve

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Re: [asterisk-users] REGEX expression for NXXNXXXXXX?

Hi Again Brent -

> > What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
> > NXXNXX?
>
> I think you've got it right already.  What do you need to do?

If you wanted to get more specific and identify ONLY NANP, you may
have to break it out into more than just one rule:

_800NXX
_888NXX
_877NXX
_866NXX


etc.

BTW: I just read up and discovered that 855, 844, 833, and 822 are
reserved but not yet used.


- Noah

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Re: [asterisk-users] North American voice BRI - Informal survey

Jon Pounder wrote:
 > most of the first level reps I have ever talked to in the last 10
> years don't even know it exists, higher level people claim they don't  
> offer it, still higher level people know what you are talking about  
> when you say its tariffed and finally cave in to what you want.

It was not hard over here.

> similar situations with the telco:
> - outsource vendor exclusions (don't call me)
> - no line test exclusion (don't test my line without me asking first)
> 
> etc etc
> 
> all "top secret" but does exist and you have to persist to get it (at  
> no charge)
> 
> the secret is get one reference phone number with this stuff setup  
> right, and then insist they look at the configuration for that line  
> when they tell you it doesn't exist.
> 
> 
> I would be willing to help out with a driver, but without a line and  
> card I am not sure how productive that would be.

As I've already said, I can get one, and it's not a big deal, so I'll be
the test case.

-Stephen-

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Re: [asterisk-users] REGEX expression for NXXNXXXXXX?

Hi Brent -

> What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
> NXXNXX?

I think you've got it right already.  What do you need to do?


- Noah

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Re: [asterisk-users] North American voice BRI - Informal survey

Quoting Stephen Bosch <[EMAIL PROTECTED]>:

> Jeff Davis wrote:
>> Stephen Bosch wrote:
>>> Your rep at Sangoma? Or your reseller?
>>
>> That wasn't very clear. Sorry. It was Sangoma.
>> (I would be more verbose, but I don't want to spam the list)
>
> I just wanted to make sure it wasn't stale information.
>
>>> This is a real chicken-and-egg problem. More people would get BRI if
>>> there were affordable hardware for it.
>>>
>>> I would like to see them write a NAm driver for it. To get them to take
>>> the chance, there have to be enough people willing to purchase the card
>>> to make them consider it seriously.
>>>
>>> The other option is a bounty or community support to get it done. The
>>> hardware already exists.
>>>
>>> The more people make noise about this, the better the chances of that
>>> happening.
>>
>>
>> If there was a driver available, I'm still not sure how many installs I
>> could sell. Verizon wants to pretend the service doesn't exist, and the
>> largest CLEC in my area doesn't even sell it. (I even offered to buy my
>> CLEC rep dinner and she wouldn't sell it to me.)
>
> This is not at all surprising -- they'd be re-selling Verizon's service.
> We've already heard from another poster how eager CLECs are to resell
> the incumbent's service. A lot of mutual sabotage goes on (and I have
> this from insiders).
>
> There is the theory of the "deregulated, competitive market" and then
> there is the practice.
>
>> Without telco support I
>> think that the only real market for this is the DIY crowd.
>
> ISDN had the bad luck of entering adoption right around the AT&T
> breakup. BRI does cost as much to provision as PRI for a smaller
> revenue, so, with maximizing profits on the brain, they're just not keen.
>
> It's also small thinking.
>
> But it is also the law. This stuff is supposed to be available.

most of the first level reps I have ever talked to in the last 10  
years don't even know it exists, higher level people claim they don't  
offer it, still higher level people know what you are talking about  
when you say its tariffed and finally cave in to what you want.

similar situations with the telco:
- outsource vendor exclusions (don't call me)
- no line test exclusion (don't test my line without me asking first)

etc etc

all "top secret" but does exist and you have to persist to get it (at  
no charge)

the secret is get one reference phone number with this stuff setup  
right, and then insist they look at the configuration for that line  
when they tell you it doesn't exist.


I would be willing to help out with a driver, but without a line and  
card I am not sure how productive that would be.


>
>> Of course, as you point out, we'll never know how big the market is
>> without a driver.
>
> The marginal effort of another driver is comparatively low. The card
> already exists.
>
>> I think that the only real incentive for Sangoma to write a driver for
>> an unproven market would be if there were a community driver available,
>> and the cards start selling. The addition of a manufacturer supplied and
>> supported driver would likely increase sales.
>
> This is the source of my other suggestion, that we put up a bounty or
> launch a community driver project. I've always felt there was latent
> demand -- this is an unserved need:
>
> - smaller installations need advanced features like call progress
> control, that they can only get in a BRI
> - the advantages in sound quality are substantial when compared with analog
> - unmanaged VoIP lines are just not reliable enough for serious businesses
> - North America is totally behind on this, we look like chumps
>
> I'm prepared to sacrifice myself for this and get a BRI for our office
> if we can get a driver that supports the signalling. That's a
> contribution I'm happy to make.
>
> -Stephen-
>
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Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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Re: [asterisk-users] REGEX expression for NXXNXXXXXX?

2007-07-05 Thread Mik Cheez

This would let you include/ignore a leading 1

1{0,1}[2-9]{2}[0-9]{8}

Brent Torrenga wrote:
> Hola,
> 
> What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
> NXXNXX?
> 
> Sincerely,
> 
> Brent A. Torrenga
> 
> Torrenga Engineering, Inc.
> 907 Ridge Road
> Munster, Indiana 46321-1771
> 
> tel:+1 219 836 8918 x325
> fax:+1 219 836 1138
> email:[EMAIL PROTECTED]
> web:www.torrenga.com
> 
> 
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 


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[asterisk-users] Visually impaired employees

2007-07-05 Thread Al Bochter

I have a customer asking about the type of equipment there is for
visually impaired employees working in a call center for inbound sales.

-- 

Best regards,

Al Bochter
http://www.BochterServices.com

---
See what we are selling at auction 
http://www.epier.com/auctions.asp?bochterservices
---
Take a look at our online store
http://www.bochterservices.com/onlinestore/
---


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Re: [asterisk-users] North American voice BRI - Informal survey


Maybe I am looking at things too naively, but if all thats different  
is the signalling standard is this really a monumental effort to make  
a driver work with more than one standard ?

I assume you would have various line encodings to deal with just like  
t1, so you have a driver layer that handles that, then you would have  
various "commands" to put the driver into various states, eg: next  
piece of data is a whatever, decode the whatever, wait next command,  
or call them frame types or headers or whatever the right terminology  
is. Shouldn't a well written driver be pretty simple to just  
substitute the new "commands" into the existing state machine ?




Quoting Dave Donovan <[EMAIL PROTECTED]>:

> On 7/4/07, Jon Pounder <[EMAIL PROTECTED]> wrote:
>>
>> Quoting Darren Wright <[EMAIL PROTECTED]>:
>>
>>> I wonder if this is issue is largely limited to to Canada.  (thus
>>> limiting the market)  In the states I think you can get PRI for around
>>> $250.  Am I right?  In Canada, you have to have about 9 or 10 lines to
>>> justify a PRI.  At $250, the cost and added features could justify PRI
>>> at around 4 lines.  Mind you, that still leaves a whole tonne of systems
>>> at the 4 lines and under mark.
>>>
>>
>> how do you come up with that ? (what are you assuming for line and pri
>> costs ?)
>>
>> when we had a bunch of lines (10+) through an at&t reseller in toronto
>> we were paying $35 each with all the features and I have never seen
>> any sort of t1 less than about $700 in Ontario, so that works out to
>> about 20 lines for breakeven - at that point what are you really
>> gaining except making it easier on the telco to deliver, yet you have
>> all your eggs in one basket and if there is a hardware or physical
>> plant issue you are completely down.
>
>
> I was wrong on a few counts here.  You're right.  I was using a very
> optimistic price for PRI service of around $550 which you can get in
> Toronto.  And I was using list pricing for Bell's featureful line that's
> like centrex lite, the name is something like 'direct line'.  That is around
> $55 at full price.  So you're right, I wasn't comparing apples to apples.
> If you negotiate your analog line price as agressively as your PRI price
> then you need more channels to make sense.
>
> The other thing that I was thinking is that I prefer PRI to analog so much
> that I even if it cost a hundred bucks more a month, it's still attractive
> to me.
>
> All that tends to support our contention that there should be a market for
> NA BRI support.  You'd think many installations would benefit.
>
>
>> No way.message rates lines hover at $350, and flat rate's run
>>> $450-$500 or so.
>
>
> Mea Culpa on that count too.  I'll stop guessing at pricing now.  :-)
>
> Dave



Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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Re: [asterisk-users] North American voice BRI - Informal survey

Jeff Davis wrote:
> Stephen Bosch wrote:
>> Your rep at Sangoma? Or your reseller?
> 
> That wasn't very clear. Sorry. It was Sangoma.
> (I would be more verbose, but I don't want to spam the list)

I just wanted to make sure it wasn't stale information.

>> This is a real chicken-and-egg problem. More people would get BRI if
>> there were affordable hardware for it.
>>
>> I would like to see them write a NAm driver for it. To get them to take
>> the chance, there have to be enough people willing to purchase the card
>> to make them consider it seriously.
>>
>> The other option is a bounty or community support to get it done. The
>> hardware already exists.
>>
>> The more people make noise about this, the better the chances of that
>> happening.
> 
> 
> If there was a driver available, I'm still not sure how many installs I 
> could sell. Verizon wants to pretend the service doesn't exist, and the 
> largest CLEC in my area doesn't even sell it. (I even offered to buy my 
> CLEC rep dinner and she wouldn't sell it to me.)

This is not at all surprising -- they'd be re-selling Verizon's service.
We've already heard from another poster how eager CLECs are to resell
the incumbent's service. A lot of mutual sabotage goes on (and I have
this from insiders).

There is the theory of the "deregulated, competitive market" and then
there is the practice.

> Without telco support I 
> think that the only real market for this is the DIY crowd.

ISDN had the bad luck of entering adoption right around the AT&T
breakup. BRI does cost as much to provision as PRI for a smaller
revenue, so, with maximizing profits on the brain, they're just not keen.

It's also small thinking.

But it is also the law. This stuff is supposed to be available.

> Of course, as you point out, we'll never know how big the market is 
> without a driver.

The marginal effort of another driver is comparatively low. The card
already exists.

> I think that the only real incentive for Sangoma to write a driver for 
> an unproven market would be if there were a community driver available, 
> and the cards start selling. The addition of a manufacturer supplied and 
> supported driver would likely increase sales.

This is the source of my other suggestion, that we put up a bounty or
launch a community driver project. I've always felt there was latent
demand -- this is an unserved need:

- smaller installations need advanced features like call progress
control, that they can only get in a BRI
- the advantages in sound quality are substantial when compared with analog
- unmanaged VoIP lines are just not reliable enough for serious businesses
- North America is totally behind on this, we look like chumps

I'm prepared to sacrifice myself for this and get a BRI for our office
if we can get a driver that supports the signalling. That's a
contribution I'm happy to make.

-Stephen-

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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Jeff Davis

Noah Miller wrote:
>> Verizon wants to pretend the service doesn't exist
> 
> I don't know, they're advertising it:
> 
> http://www.verizonbusiness.com/us/voice/local/compare.xml#isdnbri

Sure, but when you call someone up to buy it it's a different story. Or 
perhaps it was just Verizon's usual good customer service.

Also, to get to that page from verizon.com you must first select "Large 
Business". Nowhere in small or medium business is the product offered or 
mentioned.

The discussion in this thread has been primarily about using BRI as a 
replacement for POTS in a small/medium business environment. Other than 
connecting remote offices or teleworkers, most large businesses will 
simply get PRI or larger. I had a medium sized customer of mine put in a 
full T1 just so one person could have VoIP and Internet back to the main 
office.

In light of that let me ammend my earlier statement, though I thought 
the context was clear; BRI is a great solution for small businesses, but 
  Verizon wants to pretend the service doesn't exist for them.

--
Jeff Davis
Netsource Consulting
Richmond, VA

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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Dave Donovan

On 7/4/07, Jon Pounder <[EMAIL PROTECTED]> wrote:

Quoting Darren Wright <[EMAIL PROTECTED]>:

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

how do you come up with that ? (what are you assuming for line and pri
costs ?)

when we had a bunch of lines (10+) through an at&t reseller in toronto
we were paying $35 each with all the features and I have never seen
any sort of t1 less than about $700 in Ontario, so that works out to
about 20 lines for breakeven - at that point what are you really
gaining except making it easier on the telco to deliver, yet you have
all your eggs in one basket and if there is a hardware or physical
plant issue you are completely down.

I was wrong on a few counts here.  You're right.  I was using a very
optimistic price for PRI service of around $550 which you can get in
Toronto.  And I was using list pricing for Bell's featureful line that's
like centrex lite, the name is something like 'direct line'.  That is around
$55 at full price.  So you're right, I wasn't comparing apples to apples.
If you negotiate your analog line price as agressively as your PRI price
then you need more channels to make sense.

The other thing that I was thinking is that I prefer PRI to analog so much
that I even if it cost a hundred bucks more a month, it's still attractive
to me.

All that tends to support our contention that there should be a market for
NA BRI support.  You'd think many installations would benefit.

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

Mea Culpa on that count too.  I'll stop guessing at pricing now.  :-)

Dave
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Re: [asterisk-users] sometimes calls drop during attended transfer

Hi Giorgio -

> 1 - my asterisk version is 1.2.18
> 2 - my SIP devices are SNOM phones
> 3 - no SIP provider is involved...they are connected to my
> Asterisk...this is the strangest thing.
> This happens sometimesI think it could be a network overload...can
> it be?

Well, that's possible, but there would have to be quite a bit of
traffic.  You could check to see if any of your switches have recorded
any packet collisions.

I've also read about attended transfer bugs on some versions of Snom
firmware.  I've only ever used Snoms with blind transfers, so I've
never seen any of the bugs myself.  What model(s) of Snom do you use,
and what firmware version(s)?


- Noah

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Re: [asterisk-users] garbled calls

2007-07-05 Thread Anthony Francis

Joe acquisto wrote:
> . . .
>   
>> QOS across the internet is pointless and further more doesnt really 
>> exist, I would suggest setting qualify=200 in sip.conf so that asterisk 
>> will not send a call to the remote end if they are more than 200 
>> milliseconds away.
>>
>>
>> 
>
> "Away", in what sense?  Are you referring to packet "latency"?  How does 
> Asterisk measure this?  Ping response?
>
> joe a.
>
>
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Yes latency, and I believe it uses a sip message and waits for an ack to 
determine latency.

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[asterisk-users] REGEX expression for NXXNXXXXXX?

2007-07-05 Thread Brent Torrenga

Hola,

What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?

Sincerely,

Brent A. Torrenga

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com


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Re: [asterisk-users] exits in NJ

2007-07-05 Thread Bill Michaelson

Hooyoo kiddin? Exit 34, I-80.

And betta Inglish, myass...

Bill, Exit 8, NJTP

Date: Tue, 03 Jul 2007 18:13:47 -0400
From: Mark Phillips <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse

Damn!!! Beat me to it ;-}

As an Englishman now living in New Jersey (strangely nowhere near an
exit) I have to say that the local idiom and accent leaves a significant
amount to be desired.

Terms like "New Joisey", "Shuwa" ,"wadder", "badderies",
congradulations" etc make me wonder if I'm in an English speaking
country at all. 

I've heard better English spoken in Nigeria.

Mark

smime.p7s
Description: S/MIME Cryptographic Signature
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[asterisk-users] connecting 1.2 and 1.4 using SIP

2007-07-05 Thread Jerry Geis

Is there something special in connecting a 1.2 and 1.4 systems???
I have connected a number of 1.2 systems using SIP between them and had 
no issue.
In this case I am getting "407 Proxy Authentication Required"

I did the same steps I did for 1.2

Basically I have entries in sip.conf on both machines with the IP 
pointing to each other.

[x_to_y]
type=friend
username=x_to_y
secret=yes
host=25.x.x.x
context=x_to_y
allow=ulaw
allow=alaw

<--- Reliably Transmitting (no NAT) to 25.x.x.x:5060 --->
SIP/2.0 407 Proxy Authentication Required^M
Via: SIP/2.0/UDP 
25.x.x.x:5060;branch=z9hG4bK7d55a70d;received=25.x.x.x;rport=5060^M
From: "528 528" ;tag=as7855961b^M
To: ;tag=as0ab59997^M
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE^M
User-Agent: Asterisk PBX^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
Supported: replaces^M
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
nonce="49cd7fc3"^M
Content-Length: 0^M

THanks,

Jerry



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[asterisk-users] Slow list

2007-07-05 Thread Philipp Kempgen

Since the list was switched over to API-Digital almost
every message I get is older than a week. Coincidence?
Is anyone else having trouble?

Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] sometimes calls drop during attended transfer

2007-07-05 Thread gincantalupo

Hi Noah,
1 - my asterisk version is 1.2.18
2 - my SIP devices are SNOM phones
3 - no SIP provider is involved...they are connected to my 
Asterisk...this is the strangest thing.
This happens sometimesI think it could be a network overload...can 
it be?

TIA

Giorgio


Noah Miller wrote:
> Hi Giorgio -
>
>   
>> I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
>> my transfers make the call drop and I get this on my log:
>> 
>
> Some questions:
>
> 1. What asterisk version are you using?
> 2. What are your SIP devices?
> 3. Who is your SIP provider? (Judging by your CLI output, I'm guessing
> you're using one.)
>
>
> - Noah
>
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>   


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Re: [asterisk-users] North American voice BRI - Informal survey

> Verizon wants to pretend the service doesn't exist

I don't know, they're advertising it:

http://www.verizonbusiness.com/us/voice/local/compare.xml#isdnbri


- Noah

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Re: [asterisk-users] North American voice BRI - Informal survey

Quoting Frank Ochmann <[EMAIL PROTECTED]>:

> List,
>
> just my two cents here on BRI cards for Asterisk - sorry if the
> following info was posted before/is redundant.
>
> You will find the following BRI cards for Asterisk. Depending on the
> chipset/Asterisk module they will/will not support NT mode, have
> different numbers of ports, scale well when cards are combined, offer
> PCI/PCIe interfaces, support other protocolls like Q.Sig and so on.
>
> The most commom (here in Europe) would be:
>
> Digium B410 - supports NA BRI
> (http://www.digium.com/en/products/hardware/b410p.php)

the link you posted specifically says it doesn't support it.



>
> Sangoma A500
> (http://sangoma.com/datasheets/A500BRI)
>
> Junghanns Duo/Quad/OctoBRI
> (http://www.junghanns.net/en/produkte.html)
>
> BeroNet BN2S0/BN4S0/BN8S0
> (http://www.beronet.com/index.php?option=com_content&task=category§ionid=5&id=26&Itemid=28&lang=en)
>
> Sirrix PCI4S0
> (http://www.sirrix.com/content/pages/pci4s0en.htm)
>
> Dialogic/Eicon DIVA boards
> (www. dialogic.com)
>
> Other cards: Fritz!Card PCI (http://www.avm.de/en) or their C2/C4 cards,
> and basically clones of the cards mentioned above from China (OpenVox
> et. al). Basically check for the HFC-S chipset (the icon outline of the
> Cologne Cathedral is printed on these chips) on the net.
>
> All of them work with Asterisk, but settings/combinations with server
> hardware, analogue cards etc. can be an issue, so the easiest way to use
> BRI with Asterisk (for me) is to use a Patton SmartNode.
>
> BR,
>
> Frank
>
> --
> LocaNet oHG - http://www.loca.net
> Lindemannstrasse 81, D-44137 Dortmund
> tel +49 231 91596-23, mobil +49 172 2120354
> sip:[EMAIL PROTECTED]
>
> Registergericht Amtsgericht Dortmund HRA 14208
> Geschäftsführer Sven Haufe, Henning Holtschneider
>
>
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Jon Pounder

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Re: [asterisk-users] SIP / STUN / Network - Help!!

Hi Gary -

> What I want to do is take one of my SIP devices to my office (which is ALSO
> behind another NAT) and try to connect with my home Asterisk box with it.
>
> For port forwarding, my AsteriskNOW box has a static IP on the inside of my
> NAT and I've configured the LinkSys router to port-forward ports 5060 (TCP &
> UDP) and all the RTP port range used (UDP only) to the static IP of the
> AsteriskNOW box. - Was this the right thing to do?

Yes.  Just one thing, for the Sipura that you took to your office, did
you set "nat=yes" in sip.conf?

If the ports are forwarded, and Asterisk recognizes that this device
is NAT'ed, you shouldn't need to have a STUN server.


- Noah

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Re: [asterisk-users] Upgrade Asterisk

> It is recommended
> to stop asterisk b4r doing "make install" of new version .

It should work to do a "make install" of a new version while the
previous version is running.  At least, I've never had any issues
doing it.


- Noah

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Re: [asterisk-users] North American voice BRI - Informal survey

2007-07-05 Thread Frank Ochmann

List,

just my two cents here on BRI cards for Asterisk - sorry if the
following info was posted before/is redundant.

You will find the following BRI cards for Asterisk. Depending on the
chipset/Asterisk module they will/will not support NT mode, have
different numbers of ports, scale well when cards are combined, offer
PCI/PCIe interfaces, support other protocolls like Q.Sig and so on.

The most commom (here in Europe) would be:

Digium B410 - supports NA BRI
(http://www.digium.com/en/products/hardware/b410p.php)

Sangoma A500
(http://sangoma.com/datasheets/A500BRI)

Junghanns Duo/Quad/OctoBRI
(http://www.junghanns.net/en/produkte.html)

BeroNet BN2S0/BN4S0/BN8S0
(http://www.beronet.com/index.php?option=com_content&task=category§ionid=5&id=26&Itemid=28&lang=en)

Sirrix PCI4S0
(http://www.sirrix.com/content/pages/pci4s0en.htm)

Dialogic/Eicon DIVA boards
(www. dialogic.com)

Other cards: Fritz!Card PCI (http://www.avm.de/en) or their C2/C4 cards,
and basically clones of the cards mentioned above from China (OpenVox
et. al). Basically check for the HFC-S chipset (the icon outline of the
Cologne Cathedral is printed on these chips) on the net.

All of them work with Asterisk, but settings/combinations with server
hardware, analogue cards etc. can be an issue, so the easiest way to use
BRI with Asterisk (for me) is to use a Patton SmartNode.

BR,

Frank

-- 
LocaNet oHG - http://www.loca.net
Lindemannstrasse 81, D-44137 Dortmund
tel +49 231 91596-23, mobil +49 172 2120354
sip:[EMAIL PROTECTED]

Registergericht Amtsgericht Dortmund HRA 14208
Geschäftsführer Sven Haufe, Henning Holtschneider


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Re: [asterisk-users] sometimes calls drop during attended transfer