Re: [asterisk-users] AMI able to call from known endpoint to unknown endpoint?

[benoit plessis]

On Thu, Jul 31, 2008 at 05:28:42PM -0700, Stephen Cattaneo wrote:
> Both are sitting behind a Linksys IP PBX (SPA9000).  On the Linksys IP
> PBX I have set the outside number 5000 to connect to 3001.  3002 does
> not have a similar external mapping (this would defeat the purpose of
> the test I am attempting).
> [...]> 
> Is it possible (and if yes, how can I do this) to use the AMI's
> "originate" to call from 5000 to 3001?

Don't you mean to 3002 ?
either that or you will make a loopback call ...

Anyway, if that's what i understood it's impossible. You can see
a PABX a little bit little a network NAT device (well in your specific
problem). Behind the Linksys there is a private network, with one
public adresse, which is a static map to one private adresse.

Well at least it's how your asterisk IPBX will see things.
Basically when using asterisk to connect (call) two extension,
asterisk will dial each extension, and then connect them. Since
he doesn't have any access to your 3002 internal extension, there
is no way this could work.

When calling from 3001 you can reach the 3002 extension, but only 
because you are within the same "network", to keep the metaphor

-- 
Benoit

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[asterisk-users] SIP registration

[Nhadie]

Hi,

I have this weird problem i cant explain.

i have two asterisk, i'm using realtime table for my sip/user accounts.
my database is on a mysql cluster.

my prob is if i register on phone on asterisk 1 it is ok, but on second 
asterisk it can't,

  Registration from '"122144" ' failed for 
'12.34.56.78' - Wrong password

but both asterisk talks to a single mysql cluster.

i defined this on my sip.conf

domain=10.10.10.130
domain=10.10.10.131
domain=my.domain.com

any ides? TIA

Regards,
Ron

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[asterisk-users] XMPP developers

[Dean Collins]

Are there are any xmpp developers on this list? 

 

I might have a small consulting project to build an XMPP chat
application/(or even better alter off the shelf application with desired
customizations)

 

Email me for details.

 

 

Regards,

Dean Collins
[EMAIL PROTECTED] 

+1-212-203-4357 (New York) 
+61-2-9016-5642 (Sydney)
http://www.Cognation.net  

 

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Re: [asterisk-users] IP door opening devices

[Julian Yap]

C F,

Does the 2nd port of the ATA with 2 FXS ports just work like a
'pass-through' that is connected to the DTMF Relay?  Or am I totally
off track?

Any ATA's with 2 FXS ports that you can recommend?

Thanks,
Julian

On Thu, Jul 24, 2008 at 3:27 AM, C F <[EMAIL PROTECTED]> wrote:
> leave the existing keypad there. as for integrating it with asterisk.
> use an ata with 2 FXS ports. one FXS port connect to a viking door box
> http://www.vikingelectronics.com/ and set the ATA to do hotline on it.
> that door box is a regular analog phone in the shape of a door box
> that when call is pressed it goes offhook hence the requirement of
> hotline mode. it also has auto answer that when you call the box it
> goes off hook automaticaly.
> then use a relay from http://www.mikesandman.com/ that gets activated
> on ring connect that to the second FXS and that will unlock the door.
>
>
> On 7/24/08, Chris Bagnall <[EMAIL PROTECTED]> wrote:
>> Greetings list,
>>
>> We have a client with an analogue door intercom/opening unit which we're
>> attempting to replace with an IP variant. The existing unit has the
>> following functionality:
>>
>> 1) Intercom - visitor hits "call", talks to operator
>> 2) Door opening - operator can open the door by dialling a 4-digit PIN
>> followed by * (the door unit interprets the DTMF tones)
>> 3) Door opening - the door unit has a numeric keypad to enable approved
>> persons to enter by entering the 4-digit PIN on the keypad
>>
>> We've tried getting the existing unit working with an ATA, but it's only
>> about 50% reliable (hangup not always detected, DTMF not always detected,
>> etc.), so it's probably time to look at fully IP alternatives.
>>
>> Any suggestions gratefully appreciated.
>>
>> Regards,
>>
>> Chris
>>
>>
>>
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Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Paul Hales]

That's a good question - the plantronics are available with 
interchangeable ends - which makes them easy to move between different 
phones.

PaulH

Simon wrote:
> So any 2.5" headset will work with the SPA922?
>
> On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales <[EMAIL PROTECTED]> wrote:
>   
>> Plantronics.
>>
>> PaulH
>>
>>
>> Simon wrote:
>> 
>>> Hi there,
>>>
>>> Is anyone using a headset with one of these phones? If so, can you
>>> recommend any?
>>>
>>> Thanks
>>>
>>> Simon
>>>
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>> 
>
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Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Simon]

So any 2.5" headset will work with the SPA922?

On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales <[EMAIL PROTECTED]> wrote:
>
> Plantronics.
>
> PaulH
>
>
> Simon wrote:
>> Hi there,
>>
>> Is anyone using a headset with one of these phones? If so, can you
>> recommend any?
>>
>> Thanks
>>
>> Simon
>>
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>
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[asterisk-users] AMI able to call from known endpoint to unknown endpoint?

[Stephen Cattaneo]

Hi I am new to asterisk and to the AMI.

I have been automating calls using the AMI's originate, this has been
working fine for me.  I have been calling from one end point registered
with the asterisk to another endpoint registered with the asterisk.

Now I want to be able to call from a known extension to an unknown
extension (from the point of the asterisk).

I have two SIP phones with userids: 3001 and 3002.

Both are sitting behind a Linksys IP PBX (SPA9000).  On the Linksys IP
PBX I have set the outside number 5000 to connect to 3001.  3002 does
not have a similar external mapping (this would defeat the purpose of
the test I am attempting).

By telling asterisk to use 5000, I can call to 3001 from other asterisk
extensions.  Similarly I can call from 3001 to other asterisk
extensions, by again telling asterisk to use 5000.

I want to call from 5000/3001 to 3002.  If I manually dial 3002 from
5000/3001, 3001 and 3002 will become connected.

Is it possible (and if yes, how can I do this) to use the AMI's
"originate" to call from 5000 to 3001?

Something like:
channel=SIP/5000
exten = 3001
context= ??
priority=1
caller_id= test call

Thanks,

Steve


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Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Paul Hales]

Plantronics.

PaulH


Simon wrote:
> Hi there,
>
> Is anyone using a headset with one of these phones? If so, can you
> recommend any?
>
> Thanks
>
> Simon
>
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Re: [asterisk-users] Setting up ring group

[Kevin P. Fleming]

Tom Moore wrote:
> This works only half way.
> This gives the ring function I want, but doesn't take in to account the 30
> sec timer to send to voicemail if the line is not answered.

What you are looking for is a 'queue' in Asterisk terminology. These
already exist and can be built and managed using app_queue.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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[asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Simon]

Hi there,

Is anyone using a headset with one of these phones? If so, can you
recommend any?

Thanks

Simon

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[asterisk-users] Friday August 1st @ 12 Noon EDT

[randulo]

Happy August.

After two fiascos, let's try this again. I'm not positive John Todd
will be available, so we will play it by ear. If he is, we can talk
about Astricon news and Asterisk User Groups as planned. If John can't
make it, we'll talk about anything anyone wants to discuss. There is
plenty happening in VoIP at the moment.

http://VoipUsersConference.org has all the info to dial in to the call
via POTS, SIP or beheaded chicken.

The text channel is IRC on freenode.net # voip-users-conference

RSS feed is http://feeds.feedburner.com/AstUser to subscribe. You can
also use iTunes if that's your thing.

To call the SIP URI, you can dial [EMAIL PROTECTED]

To communicate the conference page to others you can use http://bit.ly/voip

For the rest, there's always http://google.com

Hope to see you all there!

r

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Re: [asterisk-users] comparing pots solutions for asterisk

[Jay R. Ashworth]

On Thu, Jul 31, 2008 at 01:20:00PM -0700, Eric Fort wrote:
>I've been looking at various solutions for getting FXS and FXO
>lines in and out of asterisk. one solution is using TDM-400 cards.
>Another solution is using the grandstream GXW400x and GXW410x
>gateways. Cost per port seems lower on the gateways and no pci slot
>is required. Why would one choose to use the TDM-400 cards? what
>would be the advantages and disadvantages of each approach?

How many lines do you need to move?

We tend to use, for hooking up our analog agent phones, the
discontinued Zhone zPlex-10 T-1 channel bank... though I suppose that's
only cost effective, even at the $250 used we pay for them, if you
already need to have a T-1 card in the machine.  Going from a single to
a dual or a dual to a quad T card is cost effective.  Putting in a
T-card from scratch, maybe not so much.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] AA50 Failover

[Dave]

 From my research, it seems that for FXOs you can use a siple RJ11  
splitter. A special splitter that gives priority to the backup split  
is preferred. These will sometimes be used for old answering machines  
where the handset can overuse the answering machine if it's picked up.

For the server end I was considering heartbeat plus dbrm. I might have  
to build my own box for that since I've seen it recommended you  
connect the two boxes by serial port as well as Ethernet.

---
Sent from my iPhone

On 31-Jul-08, at 4:57 PM, Tzafrir Cohen <[EMAIL PROTECTED]>  
wrote:

> On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote:
>> If I buy two AA50s can I set them up so that everything runs  
>> through the
>> first one, but the second one will take over if the first one goes  
>> down?
>>   I can see the extensions recovering, because they use ethernet, but
>> what about the FSO lines? Is there a way they can be spliced to both
>> AA50s so that no one need to do any emergency rewiring?
>
> Thinking aloud:
>
> what happens if you just connect th two units on the same phone line?
>
> You basically need a way to prevent the slave unit from answering  
> calls
> if the master is alive.
>
> The same should work with BRI PtMP, right?
>
> (But both are not used. And in the case of BRI, dedicated hardware is
> used instead. So I guess that there must be a good reason for that)
>
> -- 
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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[asterisk-users] Panasonic Door phone monitor to Asterisk box?

[Steve Prior]

I'm considering getting a Panasonic video door phone system (VL-GM301A) 
which can interface with a PBX and would like to connect it to my 
Asterisk box with an analog FXS port.  Of course the Panasonic 
documentation only talks about hooking it up to a Panasonic PBX which 
only talks about using Panasonic phones, so it's hard to tell whether 
the 2 wire connection from the door phone monitor is analog or digital. 
  The video door phone itself hooks to the door phone monitor with a 2 
conductor wire, so that part is clearly digital since video, audio, and 
button press all go through that wire, but since the door phone central 
station apparently plugs into a Panasonic PBX FXS port which possibly 
supports fax machines I'm thinking that this might be an ordinary analog 
connection.

Does anyone know if the door phone interface is analog or digital? 
Anyone have experience with interfacing with one?

Thanks
Steve

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Re: [asterisk-users] Whitepaper: How and to whom sell VoIP

[Grygoriy Dobrovolskyy]

i saw that billing iface somewhere else, maybe i am wrong...

2008/7/30 Mindaugas Kezys <[EMAIL PROTECTED]>

> Hello,
>
> Based on our own and our clients' experience we compiled short manual: How
> and to whom sell VoIP
>
> Hope it can be useful to some of you also.
>
> You can download it from our site: http://www.kolmisoft.com
>
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
>
>
>
>
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Re: [asterisk-users] sip registration timeout/expiration

[Grygoriy Dobrovolskyy]

you have this option on major phones also, try that.

2008/7/31 Vieri <[EMAIL PROTECTED]>

> Hi,
>
> If I set maxexpirey=60 in sip.conf and also set a "registration timeout=60"
> on client software, doesn't this mean that the SIP user (an ATA connected
> phone) should be "forced" to re-register every minute?
>
> If I look at the CLI when the SIP user registers I do see a statement
> regarding a 60 second timeout. However, after 1 minute I don't "see" it
> unregister and register again (debug is on).
>
> I'm asking this because in my LAN I have a DNS server which is dynamically
> updated (via a script) with both A and SRV records with very short TTLs.
> The idea is that the LAN SIP clients (both softphones and ATA-connected
> phones) switch from one failing (or "down for maintenance") server to
> another active box.
> This part seems to work fine. However, I'm having trouble getting the SIP
> registrations back to the first server when the latter is back on-line. The
> only way I found to do this within a minute is to kill asterisk on box 2 and
> all accounts will register on box 1 (even if the 5-second-TTL A records have
> been updated and/or the SRV entries give box1 a much higher priority).
>
> How can I make them "move" to box 1 without bringing down box 2?
>
> It seems as though "maxexpirey" is not taken into account. The extensions
> will stay on box 2 and will move to box 1 only if:
> - box 2 dies
> - or I wait around 30 minutes (I don't what this timeout could be)
>
> I've tried it on Asterisk 1.4.21.2 and 1.2.30.
>
> Any ideas?
>
> Thanks,
>
> Vieri
>
>
>
>
>
>
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Re: [asterisk-users] AA50 Failover

[Tzafrir Cohen]

On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote:
> If I buy two AA50s can I set them up so that everything runs through the 
> first one, but the second one will take over if the first one goes down? 
>I can see the extensions recovering, because they use ethernet, but 
> what about the FSO lines? Is there a way they can be spliced to both 
> AA50s so that no one need to do any emergency rewiring?

Thinking aloud:

what happens if you just connect th two units on the same phone line?

You basically need a way to prevent the slave unit from answering calls
if the master is alive.

The same should work with BRI PtMP, right?

(But both are not used. And in the case of BRI, dedicated hardware is
used instead. So I guess that there must be a good reason for that)

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] comparing pots solutions for asterisk

[Grygoriy Dobrovolskyy]

With use of tdm you can get 0 lag when using faxes & modems over FXO FXS
With use of fxo & fx gateways it is easyer to build redundancy with
heartbeat for example.

2008/7/31 Dean Collins <[EMAIL PROTECTED]>

>  Might not be lower in cost but when you take into account the cost of the
> server it would be – how about checking out the Vdex-40 appliance if you
> need 4 pots lines or less.
>
>
>
> http://www.taa.com/products-vdex-40.html
>
>
>
>
> Cheers,
>
> Dean
>   --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Eric Fort
> *Sent:* Thursday, 31 July 2008 4:20 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] comparing pots solutions for asterisk
>
>
>
> I've been looking at various solutions for getting FXS and FXO lines in and
> out of asterisk. one solution is using TDM-400 cards.  Another solution is
> using the grandstream GXW400x and GXW410x gateways.  Cost per port seems
> lower on the gateways and no pci slot is required.  Why would one choose to
> use the TDM-400 cards?  what would be the advantages and disadvantages of
> each approach?
>
> Eric
>
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Re: [asterisk-users] Setting up ring group

[Tom Moore]

This is true.
Probably is a hunt group.
Different systems use different terminology for the same thing sometimes.
 
Tom

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito
Sent: Thursday, July 31, 2008 11:19 AM
To: Ruddy G.
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Setting up ring group

Sounds more like a hunt group than a ring group.

Bruce Komito
WPTI Telecom
(775) 236-5815


On Thu, 31 Jul 2008, Ruddy G. wrote:

> Why don't you just call the Dial application for each user, one after
> another ??
> The ones that are busy will just go through. So, on the next priority,
> you dial another one.
>
>
> Tom Moore wrote:
> > Hi guys,
> > What's the best way to setup a ring group that contains 6 extensions so
that
> > when a call comes in there starts a 30 second timer and the first
available
> > device is rang instead of ringing all extensions at the same time?
> > What I want it to do is cycle through the extensions and have the system
> > ignore the ones that are busy and if there are not any free extensions
in
> > the ring group to have the system drop the caller to voicemail.
> > If none of the extensions are present in the group I'd like to also drop
to
> > voicemail.
> > Basically what I'm looking for is a multiple extensions version of the
> > standard extension macro with multiple devices and the exten busy state
> > ignored.
> >
> > Tom
> >
> >
> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > 
> >
> >
> > Internal Virus Database is out of date.
> > Checked by AVG.
> > Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date:
5/16/2008 7:42 PM
> >
>
>
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Re: [asterisk-users] AA50 Failover

[Grygoriy Dobrovolskyy]

There is 2 possibilities to do failover with asterisk:
First use openser with failover, but still you need to switch cables fxo fxs
& pri bri

Second is simplier and i would choose this one for small&medium
installations where T1/E1 is not needed:
It consists of externaising of all fxs fxo pri & bri wih sip gateways, and
installation 2 servers with heartbeat, you dont need openser in this case,
also think of registration time reducing, it helps to swich fast. So why not
T1/E1 ? it is possible no problem, but price...3K$ for audiocodes. Of course
this is relative, and choice is yours.

2008/7/31 Dave Welsh <[EMAIL PROTECTED]>

> If I buy two AA50s can I set them up so that everything runs through the
> first one, but the second one will take over if the first one goes down?
>   I can see the extensions recovering, because they use ethernet, but
> what about the FSO lines? Is there a way they can be spliced to both
> AA50s so that no one need to do any emergency rewiring?
>
> --
> Dave Welsh
> Quality of Course
> (613) 749-8248
>
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Re: [asterisk-users] comparing pots solutions for asterisk

[Dean Collins]

Might not be lower in cost but when you take into account the cost of
the server it would be - how about checking out the Vdex-40 appliance if
you need 4 pots lines or less.

 

http://www.taa.com/products-vdex-40.html 

 


Cheers,

Dean



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort
Sent: Thursday, 31 July 2008 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] comparing pots solutions for asterisk

 

I've been looking at various solutions for getting FXS and FXO lines in
and out of asterisk. one solution is using TDM-400 cards.  Another
solution is using the grandstream GXW400x and GXW410x gateways.  Cost
per port seems lower on the gateways and no pci slot is required.  Why
would one choose to use the TDM-400 cards?  what would be the advantages
and disadvantages of each approach?

Eric 

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Re: [asterisk-users] need creative solutions for number portability

[Alex Balashov]

Eric Fort wrote:
> I'm presently working on an office move and evaluation of 
> telecommunications services needed at the new location.  I'm presently 
> wrastling with an issue related to portability and geography between 
> landline carriers.  Presently certain people within the organization are 
> hopelessly in love with our 909-822- number(provided by 
> pacbell/att).  As that number is presently provisioned it rings to a 
> location geographically within 909-822 and forwards all calls to another 
> number 909-944- (Verizon)  Because of this toll is paid on all 
> incomming calls. )  

This is strange, because according to the tariffs, 909-822 and 909-944 
are in a local calling area with respect to each other.  It's not LD, so 
you should not be paying any tolls.  909-822 and 909-944 are blocks 
belonging to different ILECs, true, but they rate traffic as local 
between each other.

The source for this information is http://www.localcallingguide.com/ -> 
Area Code/Prefix/OCN.  Put in 909-822, click on it in the results, and 
you will get a list of all other rate centers local to it.

The information is not 100% accurate - it's published from commercial 
tariff research, and I am not sure exactly on what terms.

> The office is moving to another verizon area 
> (909-899, actually north fontana) and is just feet from the ATT/Verizon 
> border.  Verizon tells me the 909-822- number being held by ATT can 
> not be moved to ring direct into the new location, so toll charges for 
> inbound must still be paid.  I was hoping to avoid that.

909-899 is a local rate center to 909-822 as well.

Perhaps the issue has to do with the way they rate and provision "call 
forwarding" services vs. what is actually rated as a local call.

> What are the issues involved here?  Technically with SS7 it would seem a 
> number could ring anywhere.  my 909 npa cell phone works just fine when 
> on vacation in 941 or 808 and my 206 VoIP line finds me anywhere I have 
> a connection to the net.  What prevents this from being true with 
> landlines?  If this is a geograpic vs non-geographic issue then where 
> can I find street level maps of what wire center serves what area 
> thereby finding where to locate to be within a specific npa-nxx?  Other 
> than porting the number to VoIP, what solutions are available so inbound 
> calls incur no toll charges to the called party?

LocalCallingGuide.com is your best bet for this type of information. 
The real information is buried in thousands of pages of tariffs that the 
ILECs file and is sold in commercial guidebooks and databases by various 
consultancies that do this sort of thing, including some from Telcordia.

BTW, run away from anyone that advises you to get the LERG (Local 
Exchange Routing Guide) from Telcordia;  it's very useful, but has no 
information on local rate center coverage.

Regarding the "geographic" issue - none of the reasons for this is 
technical or geographic, but rather regulation and billing related:

Local carrier serving areas are split up into LATAs (Local Access and 
Transport Areas). [1]  LATAs are areas within which an ILEC operates, 
and within which any CLEC must be interconnected with the ILEC in order 
to provide service.  Traffic that traverses between LATA boundaries is 
known as IXC (Inter-Exchange Carrier) traffic, and this is a different 
type of carrier than a LEC.  However, certain types of intra-LATA 
traffic (traffic that stays in one LATA) can be rated long-distance too 
-- this is up to the fixed-line carrier.  Originally, LATAs were 
intended to be complete local calling areas, I think, but it never 
worked out that way, so long distance rating areas exist within LATAs as 
well as between them.  They can be very arbitrary.

All the rate centers and NPA-NXXs mentioned above are in LATA 730, and 
should be local calling areas according to the information given above. 
  Therefore, I am not sure what the basis of the carrier's suggestion 
that you should be paying LD tolls is, unless it is something that 
applies to their ring/forwarding service but not to straight-up calls.

If Verizon is an ILEC on the other side of that AT&T/Verizon border you 
mentioned, because it was a formerly GTE territory, then you can't port 
numbers across that border.  You are likely to be able to find a CLEC 
that interconnects with both carriers' tandems, though, and have them 
hang a number in whatever rate center you need.  But this should not be 
necessary since all these calls are local.

-- Alex

[1]  Enjoy a LATA map:  http://www.robotics.net/clec/LATA_Map.html

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] comparing pots solutions for asterisk

[Eric Fort]

I've been looking at various solutions for getting FXS and FXO lines in and
out of asterisk. one solution is using TDM-400 cards.  Another solution is
using the grandstream GXW400x and GXW410x gateways.  Cost per port seems
lower on the gateways and no pci slot is required.  Why would one choose to
use the TDM-400 cards?  what would be the advantages and disadvantages of
each approach?

Eric
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[asterisk-users] need creative solutions for number portability

[Eric Fort]

I'm presently working on an office move and evaluation of telecommunications
services needed at the new location.  I'm presently wrastling with an issue
related to portability and geography between landline carriers.  Presently
certain people within the organization are hopelessly in love with our
909-822- number(provided by pacbell/att).  As that number is presently
provisioned it rings to a location geographically within 909-822 and
forwards all calls to another number 909-944- (Verizon)  Because of this
toll is paid on all incomming calls. )  The office is moving to another
verizon area (909-899, actually north fontana) and is just feet from the
ATT/Verizon border.  Verizon tells me the 909-822- number being held by
ATT can not be moved to ring direct into the new location, so toll charges
for inbound must still be paid.  I was hoping to avoid that.

What are the issues involved here?  Technically with SS7 it would seem a
number could ring anywhere.  my 909 npa cell phone works just fine when on
vacation in 941 or 808 and my 206 VoIP line finds me anywhere I have a
connection to the net.  What prevents this from being true with landlines?
If this is a geograpic vs non-geographic issue then where can I find street
level maps of what wire center serves what area thereby finding where to
locate to be within a specific npa-nxx?  Other than porting the number to
VoIP, what solutions are available so inbound calls incur no toll charges to
the called party?

Eric
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[asterisk-users] AA50 Failover

[Dave Welsh]

If I buy two AA50s can I set them up so that everything runs through the 
first one, but the second one will take over if the first one goes down? 
   I can see the extensions recovering, because they use ethernet, but 
what about the FSO lines? Is there a way they can be spliced to both 
AA50s so that no one need to do any emergency rewiring?

-- 
Dave Welsh
Quality of Course
(613) 749-8248

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Re: [asterisk-users] list of minutes spent on SIP phone calls?! any advice?!

[Ruddy Gbaguidi]

You can check asterisk CDR (call detail records).
You should have a csv file in /var/log/asterisk/cdr-csv/Master.csv
You can also configure it to write the CDR in a database
http://www.voip-info.org/wiki-Asterisk+cdr+mysql

Then you can just write a script that will look at your database and 
send you a report every x day

RoLaNd RoLaNd wrote:
> Hi All,
>
> i have asterisk with 9 SIP accounts on it.
> i was wondering if theres a way to setup asterisk, to send the amount 
> of minutes each SIP account have spent incoming as well as outgoing 
> and if possible the number it called!
>
> any advice?!
>
> any help would truly be appreciated..!
>
> thanks in advance and best regards,
>
> 
> Connect to the next generation of MSN Messenger  Get it now! 
> 
>  
>
> 
>
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>
>
> Internal Virus Database is out of date.
> Checked by AVG. 
> Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 
> 7:42 PM
>   


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[asterisk-users] Astricon 2008 updates: keynotes, content, contests

[John Todd]

Astricon is only 54 days away!   If you're not 
booked, please take a moment to register for the 
conference, get your hotel room, and get your 
plane tickets before things fill up and/or get 
expensive.  This is a great opportunity to meet 
other developers, users, and members of the 
Asterisk  ecosystem, and I encourage everyone to 
attend.  While there are great things to be said 
about the mailing lists, IRC channels, and other 
forums throughout the year, it is still the 
face-to-face meetings that get the most done for 
new code ideas, business deals, and getting the 
"big picture" of what is happening in the world 
of Asterisk and VoIP in general.

With the sad demise of the VON conferences, there 
has been a loss of one of the forums in which the 
Asterisk community members would personally meet. 
While we are disappointed that VON no longer 
exists, we are encouraging anyone who would have 
otherwise attended VON to come to Astricon as a 
venue for the same purposes of face-to-face 
meetings and technology review. (But we're a lot 
lighter on the quantity of "marketing 'droids" 
than VON was in it's last few years, which 
perhaps is a good thing.)


Keynotes


While we'll have talk details in a future 
message, we do have the list of keynote speakers 
for Astricon 2008.  We're happy to be able to 
announce the following speakers:

   Brian Aker - MySQL/Sun Microsystems Chief 
Architect.  Brian is one of the core architects 
of the Open Source MySQL database, as well as 
being an Asterisk enthusiast.  His has a deep 
understanding of the nature and process of open 
source development in large projects, and 
regularly speaks on the topics of open source 
ideology, large-scale design, and how open 
networks and open software can benefit users and 
enterprises.  Sun's role as a sponsor of 
open-source software has increased dramatically 
in the past several years, especially with the 
acquisition of MySQL within the last year, and 
Brian's insights into the open-source/commercial 
worlds will be a fascinating overview.

   Stefan Öberg - General Manager and Vice 
President of Skype Telecom.  Stefan has 
previously served in different positions at 
Tele2, the Swedish telecom operator, initially in 
Sweden and then Estonia, and has an excellent 
understanding of carrier, SMB, and residential 
aspects of telephony.  I am excited that Stefan 
and Skype accepted our offer of a keynote spot, 
as his company's perspectives on being the 
world's most widely-used VoIP client and P2P 
network will interest those in the Asterisk 
community who are trying to parallel that success 
with different methodologies and tools.  I'm 
certain that Stefan's talk and views will 
generate good discussion throughout the 
conference.


Content
===

Again I'm pleased to say that this year is a 
banner year for technical talks.  We have over 60 
talks in total, and three full tracks of 
"advanced" technical discussions.  In-depth case 
studies on Unified Communications, discussions of 
the new R2 stack, advanced PRI and SS7 sessions, 
carrier topics, call center strategies, voice 
recognition, STUN/TURN/ICE implementations - it's 
a wide list of topics:

http://www.astricon.net/2008/glendale/web/confSchedule.php

There is also an entire track on business issues 
- how to sell Asterisk against other platforms, 
regulatory case studies, more open source 
discussions, intellectual property and trademark 
talk, and may more.


Contests


Jared Smith is going to be running two contests 
at Astricon!  I won't divulge too many details to 
keep people from practicing too much, but the 
concepts are: "Asterisk Quick-Draw" and 
"Debug-Off".   The first is a test to see who can 
install Asterisk and get a given dialplan working 
the fastest.  The second contest will test your 
debugging skills to find subtle and common 
problems with *NIX systems, Asterisk, or VoIP 
components that prevent calls from working 
correctly.  I'm sure that the crowd has plenty of 
ringers for these two contests - bring your 
Asterisk ninja outfits to intimidate  your 
opponents.

The prizes are pretty nice - Nokia n810 phones. 
They run Linux - maybe someone can get * running 
on it (if it's not already) once we award them.


What YOU can do to make Astricon better
===

Are you attending?  If so, please feel free to 
put one of these graphics on your website, in 
your blog postings, or anywhere else you think it 
might be useful to collect interest and get 
people to know that Astricon is coming up.

Attending:
http://www.astricon.net/2008/glendale/boxes/box_attending.jpg

Speaking:
http://www.astricon.net/2008/glendale/boxes/box_speaking.jpg

Exhibiting:
http://www.astricon.net/2008/glendale/boxes/box_exhibitor.jpg


See you there!

JT

-- 
--
John Todd
[EMAIL PROTECTED]+1-256-428-6083
Asterisk Open Source Community Director

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[asterisk-users] list of minutes spent on SIP phone calls?! any advice?!

[RoLaNd RoLaNd]

Hi All,

i have asterisk with 9 SIP accounts on it.
i was wondering if theres a way to setup asterisk, to send the amount of 
minutes each SIP account have spent incoming as well as outgoing and if 
possible the number it called! 

any advice?! 

any help would truly be appreciated..! 

thanks in advance and best regards,

_
Connect to the next generation of MSN Messenger 
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[asterisk-users] Announcing the release of Web-MeetMe 3.0.4

[Dan Austin]

This release primarily focuses on security.

A number of problems involving SQL injection
and XSS were identified and reported by Jean-Michel
Besnard.

Jean-Michel was kind enough to help with the testing
as each vulnerability was addressed.

The new release is available in the downloads section
of http://sourceforge.net/projects/web-meetme

Thank you and enjoy.

Dan

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[asterisk-users] Unregistered indication country

[J . M .]

When I do a "reload" in the Asterisk CLI I get a long list "Unregistered
indication country" lines during the parsing of the features.conf file.
Then, when parsing the indications.conf file, they seem to all get
re-registered (lines saying "Registered indication country" are displayed).

What do these lines mean and why are they unregistered and then registered?

Thanks.
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Re: [asterisk-users] Setting up ring group

[Tom Moore]

This works only half way.
This gives the ring function I want, but doesn't take in to account the 30
sec timer to send to voicemail if the line is not answered.

Tom
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ruddy G.
Sent: Thursday, July 31, 2008 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Setting up ring group

Why don't you just call the Dial application for each user, one after 
another ??
The ones that are busy will just go through. So, on the next priority, 
you dial another one.


Tom Moore wrote:
> Hi guys,
> What's the best way to setup a ring group that contains 6 extensions so
that
> when a call comes in there starts a 30 second timer and the first
available
> device is rang instead of ringing all extensions at the same time?
> What I want it to do is cycle through the extensions and have the system
> ignore the ones that are busy and if there are not any free extensions in
> the ring group to have the system drop the caller to voicemail.
> If none of the extensions are present in the group I'd like to also drop
to
> voicemail.
> Basically what I'm looking for is a multiple extensions version of the
> standard extension macro with multiple devices and the exten busy state
> ignored.
>
> Tom
>
>
>
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> 
>
>
> Internal Virus Database is out of date.
> Checked by AVG. 
> Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date:
5/16/2008 7:42 PM
>   


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Re: [asterisk-users] Setting up ring group

[Bruce Komito]

Sounds more like a hunt group than a ring group.

Bruce Komito
WPTI Telecom
(775) 236-5815


On Thu, 31 Jul 2008, Ruddy G. wrote:

> Why don't you just call the Dial application for each user, one after
> another ??
> The ones that are busy will just go through. So, on the next priority,
> you dial another one.
>
>
> Tom Moore wrote:
> > Hi guys,
> > What's the best way to setup a ring group that contains 6 extensions so that
> > when a call comes in there starts a 30 second timer and the first available
> > device is rang instead of ringing all extensions at the same time?
> > What I want it to do is cycle through the extensions and have the system
> > ignore the ones that are busy and if there are not any free extensions in
> > the ring group to have the system drop the caller to voicemail.
> > If none of the extensions are present in the group I'd like to also drop to
> > voicemail.
> > Basically what I'm looking for is a multiple extensions version of the
> > standard extension macro with multiple devices and the exten busy state
> > ignored.
> >
> > Tom
> >
> >
> >
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> >
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> > Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 
> > 7:42 PM
> >
>
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[asterisk-users] Asterisk CDR "**Unknow**" as channel name

[Ruddy G.]

Hi all
I have been looking at my asterisk CDR in the mysql database and
some channel names are set to "**Unknown**" string.
When I look at the code, everybody when calling ast_channel_alloc set a 
channel format
like SIP/%s or Zap/%s
Only app_voicemail.c doesn't when sending emails and I don't use voicemail.
Why app_voicemail needs to allocate a channel to send emails ?
And in which case I have "**Unknown**" in the CDR
Thanks

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Re: [asterisk-users] Asterisk Realtime still reads from .conf files

[Rob Hillis]

J.M. wrote:
> I've followed the instructions here 
> (http://www.voip-info.org/wiki-Asterisk+RealTime) and other places, 
> however, Asterisk still reads information from the .conf files.  How 
> can I get Asterisk to read from the database and not from the .conf files?
>
> I realize the information above is sparse, but I do not know what 
> other information is relevant.

Asterisk will /always/ read from the config files, even if you have 
RealTime configured.  To the best of my knowledge, you can't disable 
Asterisk reading the .conf files, though this shouldn't be a problem - 
just leave the .conf files empty of configuration.  When RealTime is 
configured, Asterisk will use /both/ data sources.

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Re: [asterisk-users] Setting up ring group

[Ruddy G.]

Why don't you just call the Dial application for each user, one after 
another ??
The ones that are busy will just go through. So, on the next priority, 
you dial another one.


Tom Moore wrote:
> Hi guys,
> What's the best way to setup a ring group that contains 6 extensions so that
> when a call comes in there starts a 30 second timer and the first available
> device is rang instead of ringing all extensions at the same time?
> What I want it to do is cycle through the extensions and have the system
> ignore the ones that are busy and if there are not any free extensions in
> the ring group to have the system drop the caller to voicemail.
> If none of the extensions are present in the group I'd like to also drop to
> voicemail.
> Basically what I'm looking for is a multiple extensions version of the
> standard extension macro with multiple devices and the exten busy state
> ignored.
>
> Tom
>
>
>
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>   
> 
>
>
> Internal Virus Database is out of date.
> Checked by AVG. 
> Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 
> 7:42 PM
>   


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[asterisk-users] PINCH: Anjelina Jolie XXX Video Free.

[Jay R. Ashworth]

Yes, this is really a spam.  Yes, it came through the list, not direct
to you as a forgery.  It's shown up on several of my other mailing
lists this week, as well, including, ironically, MailScanner's.

People are chasing it.

If you're not the list admin, do everyone a favor, and don't burn up
millions of innocent electrons chattering about it, ok?  :-)


Cheers,
-- jra


On Thu, Jul 31, 2008 at 03:53:12AM -0500, asterisk-users@lists.digium.com wrote:
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> Subject: [asterisk-users] Anjelina Jolie XXX Video Free.
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> Date: Thu, 31 Jul 2008 03:53:12 -0500
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> Lines: 67
> 
> 
>[1]Free Video Nude Anjelina Jolie 
>About this mailing: 
>You are receiving this e-mail because you subscribed to MSN Featured 
> Offers.
>Microsoft respects your privacy. If you do not wish to receive this MSN
>Featured Offers e-mail, please click the "Unsubscribe" link below. This 
> will
>not unsubscribe you from e-mail communications from third-party advertisers
>that may appear in MSN Feature Offers. This shall not constitute an offer 
> by
>MSN. MSN shall not be responsible or liable for the advertisers' content 
> nor
>any of the goods or service advertised. Prices and item availability 
> subject
>to change without notice.
>©2008 Microsoft | [2]Unsubscribe | [3]More Newsletters | [4]Privacy
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> References
> 
>1. http://www.cozymusic.com/img/video-anjelina.avi.exe
>2. http://www.msn.com/
>3. http://www.msn.com/
>4. http://www.msn.com/

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-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Jay R. Ashworth]

On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote:
> Yes, I tried all sorts of cables and ended up getting the local contact
> to complain to NETCOM.  An engineer came and swapped the "Fast Ethernet
> to E1" converter.

Hmmm.

Whose side is "Fast Ethernet", and whose side is E1?

Are you trying to take the E1 that they've *converted into 100BT* for
you and plug it into an E1 port?

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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[asterisk-users] Setting up ring group

[Tom Moore]

Hi guys,
What's the best way to setup a ring group that contains 6 extensions so that
when a call comes in there starts a 30 second timer and the first available
device is rang instead of ringing all extensions at the same time?
What I want it to do is cycle through the extensions and have the system
ignore the ones that are busy and if there are not any free extensions in
the ring group to have the system drop the caller to voicemail.
If none of the extensions are present in the group I'd like to also drop to
voicemail.
Basically what I'm looking for is a multiple extensions version of the
standard extension macro with multiple devices and the exten busy state
ignored.

Tom



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[asterisk-users] sip registration timeout/expiration

[Vieri]

Hi,

If I set maxexpirey=60 in sip.conf and also set a "registration timeout=60" on 
client software, doesn't this mean that the SIP user (an ATA connected phone) 
should be "forced" to re-register every minute?

If I look at the CLI when the SIP user registers I do see a statement regarding 
a 60 second timeout. However, after 1 minute I don't "see" it unregister and 
register again (debug is on).

I'm asking this because in my LAN I have a DNS server which is dynamically 
updated (via a script) with both A and SRV records with very short TTLs.
The idea is that the LAN SIP clients (both softphones and ATA-connected phones) 
switch from one failing (or "down for maintenance") server to another active 
box.
This part seems to work fine. However, I'm having trouble getting the SIP 
registrations back to the first server when the latter is back on-line. The 
only way I found to do this within a minute is to kill asterisk on box 2 and 
all accounts will register on box 1 (even if the 5-second-TTL A records have 
been updated and/or the SRV entries give box1 a much higher priority).

How can I make them "move" to box 1 without bringing down box 2?

It seems as though "maxexpirey" is not taken into account. The extensions will 
stay on box 2 and will move to box 1 only if:
- box 2 dies
- or I wait around 30 minutes (I don't what this timeout could be)

I've tried it on Asterisk 1.4.21.2 and 1.2.30.

Any ideas?

Thanks,

Vieri




  

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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Uros Djokic]

Make experiment.Make loopback Rj-45. (wire 1 from pin 1 to pin 4 wire 2 from
pin 2 to pin 5). Then put it in card and if card is OK you should see green
led.You should also see dozens of ALARMS notices or warnings on asterisk
CLI.
Also check pinout http://www.goonda.org/archive/docs/pinout.html
Pinout should be 1,2,4,5 (on card side).
Call telco. Make them check line with tester (from their point to isdn) to
ensure line is ok.
What is color of Fritz led ? (green,red or yellow ?)
What is color of card's led ? (green ? red ?)
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Lee, John (Sydney)]

> Ensure that in file indications.conf you have 
> [general]
> country=cn ; not usa ! or if you are in Australia shortcut for Australia

Uros, that was a good reminder.  However, I don't think it is related to this 
problem.

 
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Lee, John (Sydney)]

> Sounds like you're making progress.  I would try the above span
> definition without the crc4.  That might do the trick.
>
Thanks Brad.
I already tried it without crc4 but it makes no difference.


 
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Watkins, Bradley]

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Lee, John (Sydney)
> Sent: Thursday, July 31, 2008 3:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Newbie in China: Red alaram in 
> Zaptel for E1
> 
> > if after you tried both straight through & crossover cables and
> > it still give you RED alarm. just tell them you can't get any
> > clocking signal. they'll probably send someone on site and test
> > the line.
> >
> Yes, I tried all sorts of cables and ended up getting the 
> local contact
> to complain to NETCOM.  An engineer came and swapped the 
> "Fast Ethernet
> to E1" converter.
> Now we use a normal RJ45 cable to connect the converter to 
> TE412P card.
> The lights turns green but changes to yellow and green again.
> dmesg shows a continuous stream of:
> 
> wct4xxp: Clearing yellow alarm on span 1
> wct4xxp: Setting yellow alarm on span 1
> timing source auto card 0!
> timing source auto card 0!
> wct4xxp: Clearing yellow alarm on span 1
> wct4xxp: Setting yellow alarm on span 1
> timing source auto card 0!
> timing source auto card 0!
> wct4xxp: Clearing yellow alarm on span 1
> wct4xxp: Setting yellow alarm on span 1
> timing source auto card 0!
> timing source auto card 0!
> wct4xxp: Clearing yellow alarm on span 1
> wct4xxp: Setting yellow alarm on span 1
> timing source auto card 0!
> timing source auto card 0!
> wct4xxp: Clearing yellow alarm on span 1
> 
> ...and I am using the following in zaptel.conf
> 
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> 
> ... I have changed the timing source from 1 to 0 to 2 but it doesn't
> make any difference.
> 
> Any thoughts?


Sounds like you're making progress.  I would try the above span
definition without the crc4.  That might do the trick.

Regards,
- Brad

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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Uros Djokic]

Hi,

Ensure that in file indications.conf you have
[general]
contry=cn ; not usa !

Regards,
Uros

-- 
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has all 4
freedoms.
Richard M. Stallman
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Uros Djokic]

On Thu, Jul 31, 2008 at 12:31 PM, Uros Djokic <[EMAIL PROTECTED]> wrote:

> Hi,
>
> Ensure that in file indications.conf you have
> [general]
> contry=cn ; not usa ! or if you are in Australia shortcut for Australia
>
> Regards,
> Uros
>
> --
> Use Free Software http://www.fsf.org/
> ---
> Four essential software freedoms:
> 1) To study source code
> 2) To copy program
> 3) To modify source code
> 4) To redistribute modified program under condition that new user has all 4
> freedoms.
> Richard M. Stallman
>



-- 
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has all 4
freedoms.
Richard M. Stallman
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

[Lee, John (Sydney)]

> if after you tried both straight through & crossover cables and
> it still give you RED alarm. just tell them you can't get any
> clocking signal. they'll probably send someone on site and test
> the line.
>
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM.  An engineer came and swapped the "Fast Ethernet
to E1" converter.
Now we use a normal RJ45 cable to connect the converter to TE412P card.
The lights turns green but changes to yellow and green again.
dmesg shows a continuous stream of:

wct4xxp: Clearing yellow alarm on span 1
wct4xxp: Setting yellow alarm on span 1
timing source auto card 0!
timing source auto card 0!
wct4xxp: Clearing yellow alarm on span 1
wct4xxp: Setting yellow alarm on span 1
timing source auto card 0!
timing source auto card 0!
wct4xxp: Clearing yellow alarm on span 1
wct4xxp: Setting yellow alarm on span 1
timing source auto card 0!
timing source auto card 0!
wct4xxp: Clearing yellow alarm on span 1
wct4xxp: Setting yellow alarm on span 1
timing source auto card 0!
timing source auto card 0!
wct4xxp: Clearing yellow alarm on span 1

...and I am using the following in zaptel.conf

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

... I have changed the timing source from 1 to 0 to 2 but it doesn't
make any difference.

Any thoughts?
 
> p.s. note that T1/E1 crossover cable pin out is not the same
> as ethernet crossover cable.
>
Do you mean RJ48?


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