[Asterisk-Users] May be OT , but comparing

[ram]

Hi all
This might be OT question
 
But still i want to ask , if any one have idea about.
 
Does any one point me to URL SER Vs Asterisk
 
advantage and disadvantage
 
where to use SER, and where to use Asterisk
 
thanks
 
ram
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Re: [Asterisk-Users] CallerID

[Jay Milk]

Michelle,

1. Courtesy would suggest that you would have contacted the author of 
the script (me) to ask permission to modify this and host it elsewhere. 
2. What possessed you to remove ALL credits and original download 
location from the readme file?  Are you trying to pawn other people's 
work off as yours?
3. It's not exactly smart to continue someone else's versioning scheme 
if you're intending to make a "fork". 
4. Your spelling is atrocious.
5. The script is not orphaned, even though you seem to imply this in the 
readme file.


Since you are selling support for this script, that qualifies as 
commercial use and is expressly prohibited by the micro-license included 
in the original script.  Please remove it from your download page until 
you have made arrangements for further distribution with me.  I'm 
utterly amazed at the bad form I see here.


Downloads of the original script are available here:
http://www.muware.com/asterisk/

The script is alive and working well, and I've made various enhancements 
to user-requests in the recent past.


-- JM


Technical Support wrote:

Miles,

You can also download cid_rewrite from www.generationd.com  This PHP script
looks up the phone numbers in a local MySQL table, and/or uses reverse 411
on the web to lookup the name, and/or more options.

Michelle 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Friday, April 07, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
  
Could you give me an example code of how this would work, and how to 
setup the database, I'm pretty new and while what you have written 
makes sense, and sounds like a good plan I'm not sure I can implement it.



I'm using my own agi-bin for "patching" callerid and adding the name if the
number is found in a table (a csv that is mantained with a spreadsheet), it
adds the name taken from this table. Then you can see the name in the
display of the phones.

--
Alejandro Vargas
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[Asterisk-Users] Problems with registering iaxy

[Bartosz Wegrzyn - asterisk]

I used to have my iaxy registered to my old version of asterisk.
I switched to 1.2 ver and now registration fails.

my config for iax.conf for that client looks like this:

[user]
username=user
type=friend
context=sip
auth=plaintext
secret=password
host=dynamic
disallow=all
allow=ulaw
trunk=no

I provisioned my iax with this config:
[EMAIL PROTECTED] iaxyprov]# cat  iaxy
;
; IAXY Provisioning description
;
;dhcp
ip: 192.168.1.249
netmask: 255.255.255.0
gateway: 192.168.1.251
codec: ulaw
;codec: adpcm
server: 192.168.1.251
;altserver: 192.168.0.2
user: user
pass: password


When I do iax2 debug

I see this:

IAX2 Debugging Enabled
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
   Timestamp: 2ms  SCall: 12640  DCall: 0 [192.168.1.249:4569]
   USERNAME: user
   REFRESH : 60
   DEVICE TYPE : iaxy2
   SERVICE IDENT   : 0003640005a8
   PROVISIONG VER  : 3503263220
voip*CLI>
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REGAUTH
   Timestamp: 00012ms  SCall: 00011  DCall: 12640 [192.168.1.249:4569]
   AUTHMETHODS : 1
   USERNAME: user
voip*CLI>
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL
   Timestamp: 0ms  SCall: 12640  DCall: 00011 [192.168.1.249:4569]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
   Timestamp: 2ms  SCall: 08797  DCall: 0 [192.168.1.249:4569]
   USERNAME: user
   REFRESH : 60
   DEVICE TYPE : iaxy2
   SERVICE IDENT   : 0003640005a8
   PROVISIONG VER  : 3503263220


Any ideas what is wrong?
Does new asterisk differs in the iax2 registration?

Thanks

Bart


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RE: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

[Alexander Lopez]

Without modifications to Dial, I don't think so.

However,

Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED])

[dialstatus]
_X.,1,Set(TECH=${CUT(${EXTEN},-,1)}) 
_X.,2,Set(DEVICE=${CUT(${EXTEN},-,2)})
_X.,3,Dial(${TECH}/${DEVICE}||)


Or something like this...

I would also create Variable name to track each one.


>>-Original Message-
>>From: [EMAIL PROTECTED] 
>>[mailto:[EMAIL PROTECTED] On Behalf Of 
>>Douglas Garstang
>>Sent: Friday, April 07, 2006 2:21 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers
>>
>>Folks,
>>
>>When I have a dial string like this:
>>
>>Dial(SIP/3254101&SIP/3254102,20,tr)
>>
>>and I want to check the ${DIALSTATUS} variable after the 
>>dial, how do I know which number I am getting the variable for?
>>
>>And, what about this?
>>
>>Dial(SIP/3254101&SIP/[EMAIL PROTECTED],20,tr)
>>
>>What happens in that case? How can I get the ${DIALSTATUS} 
>>variable for EACH NUMBER dialled?
>>
>>Thanks,
>>Doug.
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Re: [Asterisk-Users] [OT] Centrex Question

[Tom Rymes]

I would suggest that you let your fingers do the walking and call  
your local phone company's business sales office and talk to them  
about what your needs are and how centrex will fit. I have found that  
non-technical people (ie: your client) will occasionally mix up very  
important details in situations like this and you might find that  
they were actually talking about five centrex lines, not two. Talk to  
the phone company and let them explain it to you. They should know  
better than anyone


Tom

On Apr 8, 2006, at 12:05 AM, Brian Capouch wrote:


Alexander Lopez wrote:

With strange promos and tariffs, it is possible that Centrex 'lines'
offer a larger Caller area and may in fact be cheaper than  
standard POTS when other

services are added.
For example I need a bunch of POTS lines for our ISP a few, more than
10!, years ago.


What I'm trying to understand is whether their two proposed  
"Centrex" lines will allow for the five separate "extensions" that  
they have now with their PBX.


In other words, is there any way a 2x5 PBX (with the customer  
paying for 2x POTS lines) could be swapped out with two Centrex lines?


Thanks, all.  I'm learning a lot here.

B.
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Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

"Intelligent technology solutions for small businesses."


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Re: [Asterisk-Users] [OT] Centrex Question

[Brian Capouch]

Alexander Lopez wrote:

With strange promos and tariffs, it is possible that Centrex 'lines'
offer a larger 
Caller area and may in fact be cheaper than standard POTS when other

services are added.

For example I need a bunch of POTS lines for our ISP a few, more than
10!, years ago. 



What I'm trying to understand is whether their two proposed "Centrex" 
lines will allow for the five separate "extensions" that they have now 
with their PBX.


In other words, is there any way a 2x5 PBX (with the customer paying for 
2x POTS lines) could be swapped out with two Centrex lines?


Thanks, all.  I'm learning a lot here.

B.
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Re: [Asterisk-Users] [OT] Centrex Question

[Gabriel Afana]

> I haven't dealt with Centrex for a long time, and one of my customers is
> being courted heavily by a Sprint salesperson.
>
> Am I not correct in assuming that each "line" of Centrex corresponds to
> an "extension" in the PBX world?
>
> This site has 2 POTS lines and 5 extensions, and they told me that for
> the same thing they're paying right now (~$40/POTS line) they will be
> getting two Centrex "lines" that will do the same thing.
>
> The way I understood it, each of those two Centrex lines is an extension.
>
> In general, would they still be paying their POTS fees, too?


Hmm, I read a little more (checked the index) about Centrex and I found this
as well:

Centrex phone lines are POTS lines with special, business-releated calling
features like four-digit private endpoint dialing.  These are the lines on
which you have to dial a 9 to get out.  Centrex has the same economics of
POTS - you generally pay a monthly fee that covers a certain amount of
utilization on the line; after that, you pay by the minute.  Centrex is less
widely available than POTS.  It's often absent from residential and rural
areas.  Like POTS, each Centrex line can support one phone call at a time.

Seems the Centrex lines are exactly like regular POTS, you just get some
added features.  So I am not 100% sure how they have it worked out because I
would think each POTS line which have a number (555- and 555-1112 would
be extenion  and 1112 within the Centrex group).  However, you say they
have two lines but 5 extensions.

The only thing I can think is they have two POTS which give them only two
simultaneous calls at any given time, but have 5 telephone numbers assigned
to this Centrex and just have the two Centrex lines in hunts group...but
then again this would imply that if two extensions are talking directly to
eachother, this would be utilizing the only two POTS available (since its
Centrex and has to go all the way to the CO), leaving no available lines for
incoming calls.no?

- Gabe

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RE: [Asterisk-Users] [OT] Centrex Question

[Alexander Lopez]

With strange promos and tariffs, it is possible that Centrex 'lines'
offer a larger 
Caller area and may in fact be cheaper than standard POTS when other
services are added.

For example I need a bunch of POTS lines for our ISP a few, more than
10!, years ago. 

We ended up going with Centrex lines as they were cheaper than standard
a 1FB (1 Fixed-rate Business). We ended up saving about 20% when we went
with Centrex, Obviously, NOT the intended application, but that is what
I call 'creative tariff interpretation':-)


>>-Original Message-
>>From: [EMAIL PROTECTED] 
>>[mailto:[EMAIL PROTECTED] On Behalf Of 
>>Brian Capouch
>>Sent: Friday, April 07, 2006 11:04 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: [Asterisk-Users] [OT] Centrex Question
>>
>>I haven't dealt with Centrex for a long time, and one of my 
>>customers is being courted heavily by a Sprint salesperson.
>>
>>Am I not correct in assuming that each "line" of Centrex 
>>corresponds to an "extension" in the PBX world?
>>
>>This site has 2 POTS lines and 5 extensions, and they told me 
>>that for the same thing they're paying right now (~$40/POTS 
>>line) they will be getting two Centrex "lines" that will do 
>>the same thing.
>>
>>The way I understood it, each of those two Centrex lines is 
>>an extension.
>>
>>In general, would they still be paying their POTS fees, too?
>>
>>Sorry for the noise, but I can't discuss this intelligently 
>>with them, and that's hurting me.
>>
>>Thanks.
>>
>>B.
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Re: [Asterisk-Users] [OT] Centrex Question

[Gabriel Afana]

> I haven't dealt with Centrex for a long time, and one of my customers is
> being courted heavily by a Sprint salesperson.
>
> Am I not correct in assuming that each "line" of Centrex corresponds to
> an "extension" in the PBX world?

This is straight out of my "Switching to VoIP" book (sorry if there will be
mistakes in the writing...im gonna type & post...Im not an editor :-)

Centrex
Centrex is POTS enhanced with business-grade telephony features like call
conferencing, four-digit dialing, and per-call billing rather than
per-minute billing.  It was designed to curb the need for small businesses
to invest in PBX equipment in order to get modern telephony features.  A
single Centrex customer ca use many Centrex lines, collectively called a
Centrex group.  Within the group, each line can be called using four-digit
dialing instead of the usual 7-digit dialing (i.e., the caller can omit the
prefix when placing calls within her Centrex group).  Some other PBX-like
features include the ability to easily transfer calls between lines in the
same group, or enable and disable call forwarding for a given line by
dialing a special sequence of DTMF digits.  Normally, users of Centrex have
to dial an 8 or 9 at the beginning of each call that is destined for a
receiver outside their Centrext group.

Dont know if this exactly answers your question...but it seems you still
need to pay for each line/extension in a Centrext group.  Its just a way to
get PBX-like features without having to buy a PBX.  So if your customer is
only buying two Centrext lines, they will only have two lines therefore two
extensions.

Corrections anyone?

- Gabe

(ouch, my hands hurt)

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[Asterisk-Users] Announcing Astmanproxy 1.20

[dave]

Greetings everyone,

I'm pleased to announce the release of Astmanproxy 1.20, the fast, 
flexible proxy server for Asterisk's Manager Interface.  Astmanproxy 
allows you to communicate with multiple Asterisk boxes from a single point 
of contact using a variety of I/O formats, now including support for
XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. 
Astmanproxy is written in c/pthreads (just like Asterisk) for speed and 
robustness.


Many other features have been added, including a new authentication layer 
and support for the Action: Challenge MD5 authentication method.  SSL is 
now supported, so you can encrypt from client->proxy->asterisk, 
end-to-end.  Talking to Asterisk via SSL requires that you are running an 
SSL-capable version of Asterisk (see bugs.digium.com #6812), but if 
you're not ready to do that then you can talk client->proxy via SSL.


One really interesting side effect of having both SSL and HTTP support
natively is that we in fact now support HTTPS!

With the proxy configured on localhost:1234, you can do things
along these lines:

https://localhost:1234/?Action=ShowChannels&ActionID=Foo

This has been tested extensively with good results.  The HTTP handler 
supports both GET and POST and can properly deal with XML or Standard 
output formats.  With Autofilter=on, this paradigm is ideal for creating a 
simple REST-like interface into Asterisk (even multiple boxes!) with no 
web servers needed.


Digium has graciously offered the use of their SVN community server to 
host Astmanproxy development.


For the 1.20 release, 'svn checkout' from:
http://svncommunity.digium.com/svn/astmanproxy/tags/1.20

For the development trunk (cvs-head), checkout:
http://svncommunity.digium.com/svn/astmanproxy/trunk

Tarballs are also available here:
http://www.popvox.com/astmanproxy

And there is a yahoo-groups mailing list here for users and developers of 
Astmanproxy:

http://groups.yahoo.com/group/asterisk-astmanproxy

There are many new features, changes, and enhancements in 1.20.  Please 
check them out and get us your feedback!  We would love to hear what you 
think!


Cheers,
Dave

--
David C. Troy
President/CEO
popvox, LLC
[EMAIL PROTECTED]
Phone: +1-410-647-5812
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[Asterisk-Users] [OT] Centrex Question

[Brian Capouch]

I haven't dealt with Centrex for a long time, and one of my customers is 
being courted heavily by a Sprint salesperson.


Am I not correct in assuming that each "line" of Centrex corresponds to 
an "extension" in the PBX world?


This site has 2 POTS lines and 5 extensions, and they told me that for 
the same thing they're paying right now (~$40/POTS line) they will be 
getting two Centrex "lines" that will do the same thing.


The way I understood it, each of those two Centrex lines is an extension.

In general, would they still be paying their POTS fees, too?

Sorry for the noise, but I can't discuss this intelligently with them, 
and that's hurting me.


Thanks.

B.
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Re: [Asterisk-Users] asterisk box as a voip gateway

[Mark Quitoriano]

ok i'll try that. tnx!On 4/8/06, Infobox Peru <[EMAIL PROTECTED]> wrote:
Your zaptel is wrong...

it must be:

zaptel.conf:
span=1,1,0,ccs,hdb3
dchan=16
bchan=1-15,17-31On 4/7/06, JP Carballo <
[EMAIL PROTECTED]> wrote:
Mark Quitoriano wrote:> Hi Guys,>> Im configuring my asterisk box as a voip gateway. I have TE110P which> is connected on my PBX and i will be using voip for my outgoing.>> Here's my config
>> zaptel.conf:>> span=1,1,0,ccs,hdb3> fxoks=1-32>>> zapata.conf:>> context=default> signalling=fxs_ks> group=1> channel =>1-32
>> --> Regards,> Mark Quitoriano, CCNAWhat seems to be the problem?--JP Carballo
http://www.netfone2x.comBringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.___--Bandwidth and Colocation provided by 

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 --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Mark Quitoriano, CCNAFan the flame...
http://www.spreadfirefox.com/?q=user/register&r=19441
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RE: [Asterisk-Users] Beeps and noises during calls

[mustardman29]

Try going through this PCI bus troubleshooting guide.
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting 

> -Original Message-
> From: Sean Garland [mailto:[EMAIL PROTECTED] 
> Sent: Friday, April 07, 2006 12:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Beeps and noises during calls
> 
> Sounds like it might be the pci bus..  I have a single tdm400 
> card and it isn't sharing an irq with other devices.  So that 
> leaves the pci bus.
> Weird that I would get it from 2 separate computers though 
> and different cards (had s100u's before).  The mobo is an 
> ASUS A7N8x-E deluxe, with Nforce 2, Althlon xp 3200+ and gig 
> of ram...  Guess I could replace the box with other hardware. 
>  I think I have another box here and I still have the s100u 
> cards, maybe I'll put together something else to see if there 
> is a difference...
> 
> Any other ideas would be great.
> Thanks
> Sean
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Andrew Kohlsmith
> Sent: Friday, April 07, 2006 12:27 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Beeps and noises during calls
> 
> On Friday 07 April 2006 15:03, Sean Garland wrote:
> > The beeps are not DTMF tones (at least they don't sound 
> like it).  It 
> > sounds more like the system is trying to compensate for 
> something or 
> > adjusting something.  There is a beep, sometimes several, 
> or maybe one
> 
> > or 2 in a row, and it can be faint, or loud, or whatever, but is 
> > always the same pitch and tone.  Sometimes it is 
> accompanied with loud
> talkback
> > to the earpiece.   I'm going nuts, and cannot in good conscience,
> > install or recommend this to anyone till I can resolve this.  It has
> 
> Sounds like the system is either sharing interrupts or the 
> system has a REALLY crappy PCI bus.  I ran across this on two 
> motherboards, one of which was really suprising because it 
> was a decent vendor (Asus) and wasn't doing anything other 
> than Asterisk.
> 
> You don't need shared interrupts to get this.  I had issues 
> with a Sangoma A101u and Sangoma S518 in the same box 
> (cheapass Dell P3) -- they were not sharing interrupts but 
> the T1 would have all kinds of glitches JUST like you 
> describe.  Put a Digium T100P in place of the A101u and it 
> worked great.  
> (Sounds counter to the typical threads here, but it's the truth, I
> swear.) Again, these two cards were NOT sharing interrupts 
> with each other or any other devices on the system.
> 
> -A.
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> 
> 
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[Asterisk-Users] RE: can we lend a hand?

[JR Richardson]

True Story,

I heard about these guys, a real crack team!

sjobeck, you need to post on the wiki, asterisk.org and the digium site as a
resource.  Good luck!

JR Richardson
Engineering for the Masses

> 
> Message: 12
> Date: Fri, 7 Apr 2006 12:57:31 -0700
> From: <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] can we lend a hand?
> To: 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
> 
> The Sjobeck Company provides Asterisk Integration, Configuration,
>  Support, and Training.
> 
> We are a crack team of Unix, Windows and Apple
> system integrators with 10+ years of experience working with
> clients both large and small.
> 
> The Sjobeck Company can provide turn-key solutions, or design,
> build, deliver, install, configure and deploy solutions any where in
> the world. We also do performance tuning and troubleshooting of
> existing systems.
> 
> Can we lend a hand?
> 
> www.sjobeck.com


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Re: [Asterisk-Users] Look What 911 Will Cost in Canada

[John Novack (port)]

Bob's Leaky News Service wrote:


Are you nuts or french canadian? The increase is from just under $4.00
to almost $1000.

 


I am neither. Given your rudeness, You MUST be French, though.
Certainly not Canadian.
I suppose you expect a free ride
WTF  does this have to do with Asterisk anyway?

John Novack



On 4/7/06, John Novack <[EMAIL PROTECTED]> wrote:
 


Where is the problem?
Does one expect this service to be provided for free?
1600 bucks to set up a VOIP provider, and 2K per month sounds reasonable

John Novack


Bob's Leaky News Service wrote:

   


Check out the proposed prices when this is approved.
 


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Re: [Asterisk-Users] Attended Transfer howto

[Melcon Moraes]

Well, 

Since its an Attended Transfer:

4. ) Wait for the other side to answer
5. ) Hangup and the call is transfered to the other party.

If the other party is busy you got the caller back to you.
If the other party is unavailable and you get the VM, you can dial the
hangup sequence(defined in features.conf) to get the caller back.

[]'s
MM

 -Original Message-
From:   Miles Scruggs <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: 
Sent:  Fri, 07 Apr 2006 09:37:21 -0700
Delivered:  Fri,  07 Apr 2006 13:36:15 
Subject:[Asterisk-Users] Attended Transfer howto

There is plenty of information on the wiki for setting asterisk up for 
transferring calls both from the Dail() command, and features.conf.

What really seems to be missing, is simply how do you actually perform 
the transfer?

Blind transfers are pretty simple as you only have two obvious steps.  
How though do you do attended transfers?

1.)  You have a call
2.)  You dial *2 or whatever you have setup in features.conf
3.)  You dial the ext of the person you want to bring into the call
4.) ?
5.) ?

Basically what do you have to dial, and bring the caller back into the 
call?  Also how do you get the call back if the ext your are attempting 
to transfer to is not avalible?

Thanks

Miles
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E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1144428278.286704.25565.ambrose.hst.terra.com.br,4110,Des15,Des15

 --Original Message Ends--

-- 
Melcon Moraes <[EMAIL PROTECTED]>

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[Asterisk-Users] Canada Nomadic 911 - From the Yes it will Screw Your Biz Dept

[Bob's Leaky News Service]

ESWG Consensus 12-month Report on Nomadic VoIP Technical and Operating
Impediments to 9-1-1/E9-1-1 Service Delivery in Canada


Executive Summary

Emergency Services Working Group (ESWG) recommends on a consensus
basis the Commission order the deployment of NENA Interim-2 (i2)
compliant emergency services components, systems and upgrades to
result in the operation within 18 months of enhanced 9-1-1 services
for nomadic and fixed/non-native VoIP callers in Canada.  ESWG also
recommends that the Commission establish for planning purposes a
milestone for the transition of all legacy analogue emergency services
networks to IP-based emergency networks (so called next generation
9-1-1 networks) in Canada no sooner than 36 months after the
deployment of i2.
ESWG further recommends that the Commission order eight specific tasks
with sequential milestones to assist with the orderly deployment of
i2:
1.  CISC should be ordered to deliver within 6 months a preferred PSAP
funding model for VoIP E9-1-1 addressing regional/provincial variances
and practices to produce a common national standard.
2.  CISC should be ordered to deliver a comprehensive architecture for
the implementation of VoIP E9-1-1 to deliver within 9 months
specifying roles and responsibilities of all emergency services
industry participants.
3.  All 9-1-1 Service Providers ordered to provide MSAG for the
purposes of LIS validity checking within 12 months subject to amended
agreements.
4.  All Broadband Internet Service Providers be ordered to provide LIS
capability within 12 months at their own expense.
5.  All 9-1-1 Service Providers be ordered to provide ALI/ANI
capability consistent with NENA i2 implementation within 15 months at
their own expense.
6.  All local VoIP service providers be ordered to provide Call Servers
and/or Proxy Gateway capability within 15 months at their own expense.
7.  All 9-1-1 Service Providers be ordered to provide ESGW capability
within 15 months at their own expense.
8.  All VoIP 9-1-1 calls to be E9-1-1 delivered to the correct PSAP
within 18 months (Full Production).
ESWG also recommends the establishment of at least one pilot program /
test region in Canada to evaluate and determine the best method and
practices for transition from legacy to IP emergency services.
Finally, ESWG requests Commission continue their practise of fostering
advancement in emergency services by providing deadlines for the
accomplishment of specific tasks through decisions and order the
commencement of this deployment as quickly as is prudent.

1 Background
1.1 Decision CRTC 2005-21 Mandate
This Emergency Services Working Group (ESWG) Consensus 12-month Report
on Nomadic VoIP Technical and Operating Impediments to 9-1-1/E9-1-1
Service Delivery in Canada (the 12-month Report or the Report) is in
response to the mandate given to CRTC Interconnection Steering
Committee (CISC) by the Commission in Telecom Decision CRTC 2005-21 as
follows:
72. The Commission remains of the view that, as these are technical
and operational issues, the most effective approach to resolving them
is through the CISC process, provided that CISC is guided by a fixed
timeline.
73.  Accordingly, the Commission requests CISC to submit to the
Commission, within six months from the date of this Decision, a report
identifying the technical and operational issues that impede
9-1-1/E9-1-1 service delivery when local VoIP service is offered on a
fixed/non-native basis, and, within one year from the date of this
Decision, a similar report with respect to local VoIP service offered
on a nomadic basis. Each report should identify all viable solutions
and recommend the preferred solution(s), with supporting rationale,
and a proposed timeframe for implementation. [Emphasis added]
74. The Commission notes that certain parties suggested that CISC may
benefit from participation in the NENA process in the United States.
The Commission recognizes that the progress made by other national
telecommunications regulators, with respect to the provisioning of
emergency services with local VoIP services, may be of value to the
Canadian industry and encourages CISC to monitor the reports and
progress being made in other jurisdictions on this important issue.
This 12-month Report follows up upon the issues identified in the ESWG
6-month Report on Fixed/Non-Native VoIP Technical and Operating
Impediments to 9-1-1/E9-1-1 Service Delivery (the 6-month Report) as
it was the conclusion of ESWG that the impediments in Canada were
common between the Fixed/Non-native and Nomadic VoIP 9-1-1/E9-1-1
service delivery.
In addition, this Report lays out the careful monitoring of the
US-based National Emergency Number Association (NENA) process done by
ESWG as well as the monitoring and contrast of the regulatory
environment in the United States provided by the Federal
Communications Commission (FCC) used to guide the development of the
Report.
1.2 ESWG 6-month Report on Fixed/Non-N

Re: [Asterisk-Users] Look What 911 Will Cost in Canada

[Bob's Leaky News Service]

Are you nuts or french canadian? The increase is from just under $4.00
to almost $1000.

On 4/7/06, John Novack <[EMAIL PROTECTED]> wrote:
> Where is the problem?
> Does one expect this service to be provided for free?
> 1600 bucks to set up a VOIP provider, and 2K per month sounds reasonable
>
> John Novack
>
>
> Bob's Leaky News Service wrote:
>
> >Check out the proposed prices when this is approved.
> >
> >
> >
> >BELL CANADA REPORT
> >
> >
> >ON THE
> >
> >
> >ECONOMIC EVALUATION
> >
> >
> >FOR
> >
> >
> >THE TARIFF REVISION
> >
> >
> >OF
> >
> >
> >Bell Canada's Access Services Tariff Item 315 – Zero-Dialed
> >
> >Emergency Call Routing Service (0-ECRS)
> >
> >
> >
> >*2 March 2006
> >
> >
> >
> >
> >TABLE OF CONTENTS
> >
> >   Page
> >
> >1.0GENERAL 3
> >1.1Purpose of the Study3
> >2.0SERVICE DESCRIPTION 3
> >2.1Service Characteristics 3
> >2.2Service Benefits3
> >2.3Marketing Considerations3
> >3.0TARIFF CONSIDERATIONS   4
> >3.1Tariff Components   4
> >3.2Rate Determination Principles   4
> >3.3Proposed Service Commencement Date  4
> >4.0IMPUTATION TEST 4
> >5.0DEMAND AND REVENUE INFORMATION  5
> >5.1Forecast Assumptions and Methodology5
> >5.2Number of Customers 5
> >5.3Number of 0-ECRS Calls  5
> >5.4Bell Canada Average 0-ECRS Call Duration5
> >5.5Estimates of Demand Quantities  5
> >6.0PHASE II COSTS  6
> >6.1Study Assumptions   6
> >6.2Study Period7
> >6.3Financial Parameters and Tax Rates  7
> >6.4Cost Inclusions 7
> >6.4.1  Expenses Causal to the Service  7
> >6.4.2  Capital Causal to the Service   8
> >6.4.3  Capital Causal to Demand8
> >6.4.4  Expenses Causal to Demand   8
> >6.4.5  Phase II Cost Summary   9
> >7.03RD PARTY ACQUISITION COSTS AND COSTS OF UNDERLYING CATEGORY I
> >COMPETITOR SERVICE COMPONENTS  9
> >
> >
> >
> >1.0GENERAL
> >
> >1.1Purpose of the Study
> >
> >1. The purpose of this study is to support the following revisions to
> >Bell Canada's (the Company's) Access Service Tariff 7516 (AST) Item
> >315 – 0-ECRS (Emergency Call Routing Service).
> >
> >Telecom Decision CRTC 2006-5: VoIP 9-1-1 call routing directs Bell Canada to:
> >
> >-  make 0-ECRS available to Voice over Internet Protocol Service
> >Providers (VoIPSPs) who register as resellers with the CRTC.
> >
> >-  offer 0-ECRS to VoIPSPs who are registered as resellers with the
> >CRTC at the same rate it is offered to other eligible parties -
> >Wireless Service Providers (WSPs), Canadian Pay Telephone Service
> >Providers (CPTSPs), Alternate Operator Service Providers (AOSPs),
> >Competitive Local Exchange Carriers (CLECs) and Interexchange Carriers
> >(IXCs).
> >
> >-  provide the Call Routing Lists and Traffic Operator Position
> >Records (TOPR) lists that are currently provided to traditional 0-ECRS
> >customers.
> >
> >
> >2.0SERVICE DESCRIPTION
> >
> >2. The revision to the 0-ECRS Service is to allow VoIPSPs who are
> >registered as resellers with the CRTC to access Bell Canada's 0-ECRS.
> >Using 0-ECRS, VoIPSPs will be able to route fixed non native and
> >nomadic 9-1-1 VoIP calls to Primary 9-1-1 Public Safety Answering
> >Points (PSAPs).
> >
> >2.1Service Characteristics
> >
> >3. Bell Canada will provide VoIPSPs with a Call Routing List, a TOPR
> >list and an authorization PIN number under the terms of 0-ECRS.
> >VoIPSPs will be responsible for providing a call answer centre to
> >perform location determination of a 9-1-1 VoIP caller.  The VoIPSP
> >call answer centre will then use the Call Routing List or TOPR list to
> >automatically route the call to a Primary Public Safety Answering
> >Point (PSAP) without Bell Canada Operator assistance.
> >
> >2.2Service Benefits
> >
> >4. The revision to the 0-ECRS will enable VoIPSPs to provide basic
> >9-1-1 service in Bell Canada territories.
> >
> >2.3Marketing Considerations
> >
> >5. Potential customers are currently WSPs, CPTSPs AOSPs, CLECs and
> >IXCs.  New target customers are VoIPSPs that are registered as local
> >resellers with the CRTC.
> >
> >
> >3.0TARIFF CONSIDERATIONS
> >
> >3.1Tariff Components
> >
> >6. The following rates and charges apply to 0-ECRS:
> >
> >Tariff Components  Monthly RateService Charge
> >
> >Set-up Charge, per customerN/A $1,658.09
> >Access Charge, per customer$2011.15N/A
> >
> >7. This service is provided initially to the customer under a two-year
> >contract under the terms and conditions of which are specified in the
> >0-ECRS agreement and is renewed on a successive one-year term basis.
> >
> >3.2Rate Determination Principles
> >
> >8. The proposed tariff rate(s) for 0-ECRS is based on Phase II costs
> >plus a 15% mark-up as per the Commission's determinations at paragraph
> >231 of Regulatory framework for second Price Cap, Telecom Decisio

Re: [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 9 11 /VOIP Providers

[Bob's Leaky News Service]

This is approx 247% increase. Anyone who thinks this is right is out
of their mind.

On 4/7/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
> If you are a Canadian VoIP provider or CLEC, help make a difference by
> joining the CAVP:
>
> www.cavp.ca
>
> >OMFG,
>
> >I thought April Fools day was over.  This is hard to believe.  If true it
> >tells me that Bell has not changed at all.  They are still trying to
> >manipulate and take advantage of the parts of the market they have absolute
> >control over.  The CRTC was right to continue to regulate them and leave
> the
> >VoIP providers alone.
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Re: [Asterisk-Users] Re: gotoif

[Jeremy Wadhams]

I had the same problem on a script, I suspect this is the first time you're using the "holdopt" variable?  Try setting it to zero before the read.It looks like if holdopt is "NULL" (the user doesn't input anything and you haven't got something in the variable to start with) asterisk interprets that line as:s,202,GotoIf($[ = 1 ]?4)and that generates the error.--JW- Original Message From: Shaun <[EMAIL PROTECTED]>To: asterisk-users@lists.digium.comSent: Friday, April 7, 2006 1:38:00 AMSubject: [Asterisk-Users] Re: gotoifAlso forgot to say that the error is triggered by the gotoif (reason the subject is labeled that) and not
 read...-- ~Shaun"Shaun" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]> Here is a section of my dialplan (macro)>> exten => s,200,Wait(1)> exten => s,201,read(holdopt|screen-onhold|1)> exten => s,202,GotoIf($[${holdopt} = 1 ]?4)> exten => s,203,GoTo(200)>>> it's simple really it loops telling you the caller is on hold until you > press 1 and then it sends you off to another area.  The problem right now > is that if the read() times out i get these warnings...>>> Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:183 ast_yyerror: > ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting > TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:> = 1> ^> Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:187 ast_yyerror: If you have
 > questions, please refer to doc/README.variables in the asterisk source.>>> The dial plan works and all, it's just i want those warnings to go away!>>> -- >> ~Shaun>>> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>   http://lists.digium.com/mailman/listinfo/asterisk-users> ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___
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Re: [Asterisk-Users] Steps to make trunked iax2

[Rich Adamson]

jonny hashem wrote:

Hi:
Is the difference of Asterisk verisons on two servers
effect on the iax2 trunking between them ?


Yes, without a doubt. However, I've not tried to keep track of the 
changes and would only be speculating on compatibility issues, etc.


That happens to be one of the reasons why some itsp's provide less then 
quality audio on iax links (eg, older versions).


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[Asterisk-Users] Steps to make trunked iax2

[jonny hashem]

Hi:
Is the difference of Asterisk verisons on two servers
effect on the iax2 trunking between them ?

Thanks;
jonny 

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[Asterisk-Users] Re: Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4

[Axel Thimm]

On Fri, Apr 07, 2006 at 11:20:15AM -0700, William M Conlon wrote:
> Thanks.  Useful 3rd party repository.  I've added it to my yum.repos.d
> 
> Question:  I see you spandsp, but I don't imagine that app_rxfax/ 
> app_txfax are included in your asterisk rpm,

Just try it out and you'll have a pleasant surprise. :)

> so I'll still have to build * from source.  Google shows that
> someone has an asterisk- plugins rpm for Suse that includes the
> app_rxfax.so/app_txfax.so
> 
> I imagine others would find that of interest.
> 
> bill
> On Apr 6, 2006, at 5:36 PM, Axel Thimm wrote:
> 
> >On Wed, Apr 05, 2006 at 04:30:19PM -0700, William M Conlon wrote:
> >>I was just getting to work on fax for my * system, so I thought I
> >>would bring everything up to date since there would be some new
> >>compilations involved.
> >>
> >>yum update gave me kernel-2.6.16-1.2069_FC4
> >>
> >>but after recompiling zaptel, I kept getting "FATAL module zaptel not
> >>found"
> >>
> >>Chased this for an hour with multiple recompiles and reboots.
> >>Finally dropped back to 2.6.15-1.1833_FC4, which worked before, and
> >>still works now.
> >
> >You can try the packages at atrpms.net. They are built for
> >2.6.16-1.2069_FC4.

-- 
Axel.Thimm at ATrpms.net


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Re: [Asterisk-Users] can we lend a hand?

[TC]

> > Can we lend a hand?
> 
> Yeah, dig into the bug tracker  pick up a task and start working on  
> bugs and doc
> for Asterisk ?
> 
> Or contribute to the discussion on this non-commercial list ?
> 
> >
> > www.sjobeck.com
> 
> Tim Panton
well said I suggest you can with start 
1) fixing all asterisk mutex deadlocks


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Re: [Asterisk-Users] can we lend a hand?

[Tim Panton]

On 7 Apr 2006, at 20:57, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>  
wrote:



The Sjobeck Company provides Asterisk Integration, Configuration,
 Support, and Training.

We are a crack team of Unix, Windows and Apple
system integrators with 10+ years of experience working with
clients both large and small.


That's nice, but I worry about anyone who puts IE5 as top of their
'tools we use' list.



The Sjobeck Company can provide turn-key solutions, or design,
build, deliver, install, configure and deploy solutions any where in
the world. We also do performance tuning and troubleshooting of
existing systems.

Can we lend a hand?


Yeah, dig into the bug tracker  pick up a task and start working on  
bugs and doc

for Asterisk ?

Or contribute to the discussion on this non-commercial list ?



www.sjobeck.com


Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] Fedora 'service asterisk start' problems

[Matt]

And when all this happens, your full debug log file says.?

On 4/7/06, Bob McDowell <[EMAIL PROTECTED]> wrote:
>
> I ran into a weird one last night.  If I use 'service asterisk start' I
> have problems (see below).  If I exclusively use 'asterisk
> -c' everything works normally.
>
> It happens like this:
>
> 1) 'service asterisk start'
> 2) Use asterisk normally, etc, etc - eventually change something that
> requires a restart
> 3) Issue either a CLI 'stop now' or a 'service asterisk stop', and
> asterisk stops
> 4) 'service asterisk start'
> 5) Asterisk enters a death loop reporting 'Asterisk exited with code 1'
> over and over again
> 6) Switch to another session and issue 'service asterisk stop' about a
> dozen times, and it stops.
> 7) Death-loop resumes when starting asterisk with either method
> 8) After a reboot, things are normal again
>
> Weird, eh?  It's not critical, but if you've seen it before I'd love to
> know what you found to be causing it.
>
>
> Bob McDowell
>
>
>
>
>
>
>  *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION ***
>
>
> This e-mail message and all attachments, if any,may contain confidential and 
> privileged material and are intended only for the intended recipient. Any 
> unauthorized review, use, disclosure or distribution is prohibited. If you 
> are not the intended recipient, please contact the sender by reply e-mail or 
> by calling (417) 869-9192 and destroythe original and any copies of this 
> e-mail.
>
>
>
>
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[Asterisk-Users] simple wav ringtones?

[Dr. Michael J. Chudobiak]

Can anyone suggest a good source of simple-but-distinctive wav ringtones 
for a business environment, to use on Snom phones? The built-in Bellcore 
tones are hard to distinguish, to my ear.


I want variations of "ring, ring", not Madonna or Eminem :-)


- Mike

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[Asterisk-Users] can we lend a hand?

[support]

The Sjobeck Company provides Asterisk Integration, Configuration,
 Support, and Training. 

We are a crack team of Unix, Windows and Apple 
system integrators with 10+ years of experience working with 
clients both large and small. 

The Sjobeck Company can provide turn-key solutions, or design,
build, deliver, install, configure and deploy solutions any where in
the world. We also do performance tuning and troubleshooting of
existing systems. 

Can we lend a hand?

www.sjobeck.com
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RE: [Asterisk-Users] Beeps and noises during calls

[Sean Garland]

Sounds like it might be the pci bus..  I have a single tdm400 card and
it isn't sharing an irq with other devices.  So that leaves the pci bus.
Weird that I would get it from 2 separate computers though and different
cards (had s100u's before).  The mobo is an ASUS A7N8x-E deluxe, with
Nforce 2, Althlon xp 3200+ and gig of ram...  Guess I could replace the
box with other hardware.  I think I have another box here and I still
have the s100u cards, maybe I'll put together something else to see if
there is a difference...

Any other ideas would be great.
Thanks
Sean

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Friday, April 07, 2006 12:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Beeps and noises during calls

On Friday 07 April 2006 15:03, Sean Garland wrote:
> The beeps are not DTMF tones (at least they don't sound like it).  It 
> sounds more like the system is trying to compensate for something or 
> adjusting something.  There is a beep, sometimes several, or maybe one

> or 2 in a row, and it can be faint, or loud, or whatever, but is 
> always the same pitch and tone.  Sometimes it is accompanied with loud
talkback
> to the earpiece.   I'm going nuts, and cannot in good conscience,
> install or recommend this to anyone till I can resolve this.  It has

Sounds like the system is either sharing interrupts or the system has a
REALLY crappy PCI bus.  I ran across this on two motherboards, one of
which was really suprising because it was a decent vendor (Asus) and
wasn't doing anything other than Asterisk.

You don't need shared interrupts to get this.  I had issues with a
Sangoma A101u and Sangoma S518 in the same box (cheapass Dell P3) --
they were not sharing interrupts but the T1 would have all kinds of
glitches JUST like you describe.  Put a Digium T100P in place of the
A101u and it worked great.  
(Sounds counter to the typical threads here, but it's the truth, I
swear.) Again, these two cards were NOT sharing interrupts with each
other or any other devices on the system.

-A.
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RE: [Asterisk-Users] Beeps and noises during calls

[Guido Hecken]

First, you could use a softphone like sjphone or X-Lite.
If the problem is still there, pull any card from your server and try from
internal (sip) extension to another internal extension. Also have a closer
look on your nic.

only some ideas to isolate the problem...
hope it helps a little

Regards

Guido 
 
> The beeps are not DTMF tones (at least they don't sound like it).  It
> sounds more like the system is trying to compensate for something or
> adjusting something.  There is a beep, sometimes several, or maybe one
> or 2 in a row, and it can be faint, or loud, or whatever, but is always
> the same pitch and tone.  Sometimes it is accompanied with loud talkback
> to the earpiece.   I'm going nuts, and cannot in good conscience,
> install or recommend this to anyone till I can resolve this.  It has
> happened with 2 separate installs of *, with different hardware,
> different packages installed (one is * 1.2.4 with freepbx, the other was
> * 1.0 with nothing), and different digium hardware.  The only thing that
> was the same is the Polycom phones, and SBC as a provider for the POTS
> lines...
> 
> HELP
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[Asterisk-Users] Call tracking through chan_agent using the Manager API

[Paul Robins]

Hey,
  We've been working on tracking all inbound calls to certain
call-centre members and have hit a snag, it seems when a queue delivers
a call to an Agent, chan_agent will call Local/whatever but provide no
means of associating the call in the queue with the local call. To that
end i added in an event called AgentAssociate which looks like this:

  'DestChan' => 'Local/[EMAIL PROTECTED],1',
  'Event' => 'AgentAssociate',
  'Privilege' => 'agent,all',
  'UniqueID' => '1144425133.1',
  'Agent' => '1001'

Is this an appropriate solution to my problems or am i missing out on
something vital? The key thing that this event gives us is the agent +
call IDs plus the originating Local channel, this way it's trivial to
link the two sets of calls together.

Cheers

-- 
Paul Robins
paul AT wza.us
paul AT gamingmp.com
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[Asterisk-Users] Re: queue/agent and macros?

[Shaun Reitan]

I'm doing a caller announce/screen type setup where when the agent/member 
picks up the call it announces to them that they have a call from "blah" 
press 1 for blah 2 for blah 3 for blah  I have this setup using a normal 
dialplan with out queues but i want to use queues.

--

~Shaun

"Gareth Blades" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> Cant you set the calleridname before putting the call into the queue?
>
> On Thu, 2006-04-06 at 22:57, Shaun wrote:
>> I was wondering if it was possible to run a macro once the agent/member
>> picks up, I know I can do this with dial in the extensions.conf but 
>> wasn't
>> sure about the queue.  Basically I have a macro that identifies the 
>> caller
>> and need that run.
>
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[Asterisk-Users] Re: queue/agent and macros?

[Shaun Reitan]

It sounds like you may be doing what i want to do, are you using macros?  if 
so how?


-- 

~Shaun


"Johann" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
>I set the callerid name to show the employee that will get the call what 
>kind of call it is.  We have multiple options, but actually only use 2 
>queues(with most people answering both of them).  Eventually it may be 
>expanded so there are more queues with different people on it...but this 
>way when that comes there will be little if any chance for people calling 
>in.
>
>
> --johann
>
> Gareth Blades wrote:
>> Cant you set the calleridname before putting the call into the queue?
>>
>> On Thu, 2006-04-06 at 22:57, Shaun wrote:
>>
>>>I was wondering if it was possible to run a macro once the agent/member 
>>>picks up, I know I can do this with dial in the extensions.conf but 
>>>wasn't sure about the queue.  Basically I have a macro that identifies 
>>>the caller and need that run.
>>
>>
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Re: [Asterisk-Users] Sending Access codes to a 5EE switch.

[Andrew Kohlsmith]

On Thursday 06 April 2006 13:52, Gary Ritter wrote:
> Can you show me how to write the macro to do this?

Absolutely, but as Eric suggests, there is the 'D' option to the Dial() 
command which might be easier yet.

-A.
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Re: [Asterisk-Users] Beeps and noises during calls

[Andrew Kohlsmith]

On Friday 07 April 2006 15:03, Sean Garland wrote:
> The beeps are not DTMF tones (at least they don't sound like it).  It
> sounds more like the system is trying to compensate for something or
> adjusting something.  There is a beep, sometimes several, or maybe one
> or 2 in a row, and it can be faint, or loud, or whatever, but is always
> the same pitch and tone.  Sometimes it is accompanied with loud talkback
> to the earpiece.   I'm going nuts, and cannot in good conscience,
> install or recommend this to anyone till I can resolve this.  It has

Sounds like the system is either sharing interrupts or the system has a REALLY 
crappy PCI bus.  I ran across this on two motherboards, one of which was 
really suprising because it was a decent vendor (Asus) and wasn't doing 
anything other than Asterisk.

You don't need shared interrupts to get this.  I had issues with a Sangoma 
A101u and Sangoma S518 in the same box (cheapass Dell P3) -- they were not 
sharing interrupts but the T1 would have all kinds of glitches JUST like you 
describe.  Put a Digium T100P in place of the A101u and it worked great.  
(Sounds counter to the typical threads here, but it's the truth, I swear.)  
Again, these two cards were NOT sharing interrupts with each other or any 
other devices on the system.

-A.
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[Asterisk-Users] Re: gotoif

[Shaun Reitan]

I'm just about positive i tried that, i'll give it another go.

~Shaun

"Doug Lytle" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> Shaun wrote:
>>
>>
>> Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:183 ast_yyerror: 
>> ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting 
>> TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
>>
>>
>>
>> The dial plan works and all, it's just i want those warnings to go away!
>>
>>
>>
> This has been covered a few time in the last 2 months.  You need to 
> initialize the variable:
>
>Set(holdopt=0)
>
> Before doing any testing with it.
>
> 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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[Asterisk-Users] Re: Attended Transfer howto

[Shaun Reitan]

Theirs a example of screening a call on

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial

I'm building a more advanced dialplan of that but you get the jist of it.


-- 

Shaun




"Miles Scruggs" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> There is plenty of information on the wiki for setting asterisk up for 
> transferring calls both from the Dail() command, and features.conf.
>
> What really seems to be missing, is simply how do you actually perform the 
> transfer?
>
> Blind transfers are pretty simple as you only have two obvious steps.  How 
> though do you do attended transfers?
>
> 1.)  You have a call
> 2.)  You dial *2 or whatever you have setup in features.conf
> 3.)  You dial the ext of the person you want to bring into the call
> 4.) ?
> 5.) ?
>
> Basically what do you have to dial, and bring the caller back into the 
> call?  Also how do you get the call back if the ext your are attempting to 
> transfer to is not avalible?
>
> Thanks
>
> Miles
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Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

[Erick Perez]

How much RAM disk is needed or are you using for your current needs?
We're planning to do something like this. But I can't figure proper
dimensioning.

On 4/6/06, Matt Florell <[EMAIL PROTECTED]> wrote:
> > >That is what we do actually. One drive for Linux/Asterisk and a SCSI
> > >RAID for /var/spool/asterisk/monitor
> > >
> > >
> > I'm really surprised (and impressed) that this is working for you,
> > Matt.  What are the specs of the RAID array (filesystem, drive speeds,
> > RAID level, etc.)?
>
> We use LSILogic MegaRAID 320-1 with four 320U 15kRPM 78GB SCSI drives
> RAID 1 across two logical partitions
>
> > Taking that into consideration, I didn't think that a dedicated drive
> > would make much of a difference.  Do you agree that the problem is that
> > the frames are written to disk in the same thread that is responsible
> > for bridging them between the end-points?  If so, why do you feel that
> > Linux's file buffering is inadequate?  Have you considered using
> > MixMonitor instead, as it is supposed to address this problem?
>
> I don't think it's Linux's file buffer, it's Asterisk. There is not
> much of a buffer there and once you get 100 streams running through
> it, it begins to have problems.
>
> As for MixMonitor, we haven't really messed around with tweaking
> recording since what we have now works wonderfully and 9 months ago
> MixMonitor was not production-ready 9 months ago.
>
> You need to keep in mind that we have VERY stringent requirements for
> our audio recordings. A audio recording drop rate of 0.5% is too
> high for us and is not even noticable by other Asterisk users so we
> had to figure out how to totally eliminate any skips or audio gaps in
> recordings and that's the solution that works for our needs. Others
> might get by with a fwe skips and have a much higher concurrent
> recording capacity.
>
> > >Yes, you will have to wait until off-hours to mix the recordings, but
> > >while switching to GSM does help reduce the amount of data written to
> > >the drives, the gains are mostly cancelled out by the compression that
> > >is needed to convert each stream to GSM. In the case of 60 calls you
> > >would be converting 120 streams to GSM.
> > >
> > I agree with you.  We mix our PCM recordings down to GSM, but we do it
> > on a dedicated server.  We've hijacked soxmix so that the recording is
> > available almost immediately after the call has completed.  Check out
> > this post
> > 
> > for details.
>
> We started doing it off-server as well recently, it does help to have
> a machine to do just the mixing now for all of our Asterisk servers'
> audio.
>
> > >The problem that happens with a large amount of concurrent recordings
> > >is audio skips, from a quarter second to two seconds. There isn't much
> > >of a recording buffer built-in to Asterisk so if anything causes a
> > >delay in packet delivery there will be an audio skip. No matter what
> > >kind of setup we used we have not been able to guarantee skip-free
> > >recording files in any case above 60 concurrent recordings over the
> > >course of a week. It is very hard to test for, but if you have all of
> > >your recordings listened to you will be able to know if there is a
> > >problem. For us in one case(70 concurrent recordings) the skips only
> > >occured very infrequently, maybe 0.5% of the time(10 skips in 55
> > >hours of recordings) which doesn't sound bad, but if it happens during
> > >the wrong part of a critical sales confirmation that half second skip
> > >could cost my company hundreds of dollars.
> > >
> > >
> > >
> > Do these skips coincide with drops in the actual call audio?  What kind
> > of overall call quality are you experiencing (dropped audio, dropped
> > calls, etc)?  What does the load average look like on your system at peak?
>
> We are on only Zap channels on the recording servers, so no issues
> with audio dropping. The load is low, peaking at 50%.
>
> > >The best reliable solution that I have been able to depend on is to
> > >limit the number to 60 concurrent recordings on a single P4 3.4GHz
> > >server with a LSILogic MegaRAID 320-1 and four 15k-RPM SCSI drives in
> > >a RAID 10. With this we experience usually no audio skips across the
> > >course of a week(and thousands of recordings), and it has worked this
> > >way reliably for the last nine months.
> > >
> > >If you have less strict quality standards you could probably push that
> > >number up, but you will get more and more audio skips the higher you
> > >go.
> > >
> > Thanks for another very informative post, Matt.  It looks like we share
> > a lot of similar goals.  Feel free to contact me off-list if I can ever
> > be of any help to you.
>
> I wish I had the cash for a nice big RAM disk to play with :)
>
> MATT---
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RE: [Asterisk-Users] Beeps and noises during calls

[Sean Garland]

The beeps are not DTMF tones (at least they don't sound like it).  It
sounds more like the system is trying to compensate for something or
adjusting something.  There is a beep, sometimes several, or maybe one
or 2 in a row, and it can be faint, or loud, or whatever, but is always
the same pitch and tone.  Sometimes it is accompanied with loud talkback
to the earpiece.   I'm going nuts, and cannot in good conscience,
install or recommend this to anyone till I can resolve this.  It has
happened with 2 separate installs of *, with different hardware,
different packages installed (one is * 1.2.4 with freepbx, the other was
* 1.0 with nothing), and different digium hardware.  The only thing that
was the same is the Polycom phones, and SBC as a provider for the POTS
lines...

HELP!


Thanks
Sean Garland

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoa
Sent: Friday, April 07, 2006 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Beeps and noises during calls


Some more info:

TALKOFF is the wrong recognition of DTMF component in human voice as
true DTMF signal. This is an unavoidable factor since human voice always
contain valid DTMF combination. Fortunately, presence of these valid
DTMF components are unsteady. Unlike real DTMF generated from a
touch-tone keyboard, these 'human' DTMF cannot maintain on a constant
combination. So they can be isolated by DELAY discrimination. If a
decoded DTMF signal can stay on constantly for certain duration which
exceed those normal period experienced in human voice, then it can be
identified as a real DTMF command.

taken from:

http://www.qsl.net/ve3rgw/dtmfsql.html


Zoa wrote:

>
> Have a look at this :
>
> http://www.dslreports.com/forum/remark,9151528
>
> If anybody would have such a mitel or bellcore dtmf talkoff wav file, 
> i have a very big email box you can drop it in :p
>
> Zoa
>
>
> Sean Garland wrote:
>
>> I have a very annoying problem that we hear on our end, but the other

>> party doesn't hear.  There are random beeps and echo type noises that

>> occur.  They are present during voicemails, and present on my end 
>> during calls.  Is anyone experiencing the same deal?  I have asked 
>> this a number of ways on the list, and never get a response...
>> Thank you.
>>
>>
>> Sean Garland
>> Mount Shasta, CA
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>>  
>>
>
>
>
>
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Re: [Asterisk-Users] Uplink Skype2Sip

[Erick Perez]

I cant make the proggie link my sip to skype, but skype to sip work great.
Im running winxpsp2 with a cheapo onboard sound card.


On 4/7/06, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>
> Hi all,
> anyone get it worked ? Uplink route me the call incoming from skype but when
> i answer, my skype go in error on sound card ?
> I also set in my hosts this value:
>
> 127.0.0.1  pgp01.televolution.net
> 127.0.0.1  stun01.sipphone.com
>
> This is my sip.conf
>
> [skype]
> language = it
> username = skype
> secret = 
> host = dynamic
> defaultip = 
> port = 5060
> type = friend
> context = from_eth
> canreinvite = yes
> dtmfmode = info
> callgroup = 1
> pickupgroup = 1
> fromuser = 
> insecure = very
> qualify = yes
> callerid = Test <999>
> allow = all
> Thanks all
>
> Giordano
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>


--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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[Asterisk-Users] Audiconferencing System fon Asterisk

[Erick Perez]

Just came by this link

So I'm posting to keep the community informed. I don't use or endorse
this product. I'm just letting people know about it.

http://www.indosoft.ca/audioconferencesystem.htm

Audio Conferencing System & Teleconferencing Solution that connect
seamlessly over TDM and IP networks. This audio conference system
include a comprehensive set of features and easily customizable. It's
an audio conference bridge that cater to the high volume scalability
and integration required by Audio Conferencing/Teleconferencing
Service Provider, as well as ease of use and affordable cost required
by business and enterprises.The Audio Conference Bridge developed is
superior in voice quality among other Audio Conferencing Bridges in
the industry.


--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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RE: [Asterisk-Users] Sending Access codes to a 5EE switch.

[Michael Collins]

> "show application dial"  Pay special attention to the D() option.

Eric,

Question - does the D option know that on a PRI the DTMF stream goes out
the B channel and not the D channel?  I would assume that it knows but I
thought it best to ask the question outright.  If it does, then it
should be the answer to Gary's problem.

-MC
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RE: [Asterisk-Users] Soporte

[Bob McDowell]

Did my Spanish just get better?  I can actually read some of that...


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, April 07, 2006 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Soporte

Saludos:

No entiendo muy bien la pregunta, imagino que usted desea tener los
endpoints o gateways de manera NO dinamica, se hace sencillamente
declarando ese host en sip.conf sin username/password y seteando la
variable "host=ip.de.tu.gw"
y el el gatway se le dice que no se registre, so simple

You may check this site:
http://www.voip-info.org es un buen lugar para comenzar it is a good
place to start

PS:El idioma oficial de esta lista es ingles, so please post in english
your next help request, so you will have more probabilities to find
someone that will help you


> ALguien sabe como puedo:
> Si existe alguna manera para que el * trabaje con Sip a un gateway
> Cisco y que este ultimo no se tenga que registrar al * y ademas no
> tenga que configurar nada en el sip-ua del Cisco.
> Por las pruebas que hice no hay manera,siempre el ASterisk me termina
> pidiendo registracion, pero no saben de alguna direccion donde pueda
> respaldarme, es que mi jefe leyo algunos manuales donde con protocolo
> sip se puede trabajar sin necesidad de registrarse,  hasta donde he
> investigado en los foros me dicen lo mismo, es decir, que si o si
> tengo que registrarlo al *, pero no me indican alguna pagina donde
> pueda respaldarme o corregirme si es que estoy errado Gracias de
> antemano
>
>
>
> Atentamente
> Will Vélez
>
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   *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION ***


This e-mail message and all attachments, if any, may contain confidential and 
privileged material and are intended only for the intended recipient.  Any 
unauthorized review, use, disclosure or distribution is prohibited.  If you are 
not the intended recipient, please contact the sender by reply e-mail or by 
calling  (417) 869-9192 and destroy the original and any copies of this e-mail.
 
 


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RE: [Asterisk-Users] Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface

[Bob McDowell]

Well, you control both ends, so I can't see how it would
matter much.  So long as the Nortel card matches the Asterisk card, you
should be good to go.  Asterisk doesn't have artificial limits like Nortel
does, so check what will work in your Nortel first.  For example, my
Norstar system won't allow me to switch from E&M Wink to PRI without forking
over some cash.  Your E1/T1 issue could be in the same
boat.
 
There
was an Asterisk+Vonage discussion on this list a week or so
ago...
 
Bob McDowell 
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
CGSent: Friday, April 07, 2006 12:03 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Telephony
newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1
interface
I have gone through some archive about Nortel MICS (Meridian ?)+
Asterisk Integration but I'm not sure whether same as my case
.70 telephone sets   
|  
|  
Nortel MICS 4.1  - Asterisk  
|   PSTNI have read the David Gomillion's Guide
and got the idea . However, my plan is slightly different from what he did , I
need to use Nortel MICS to connect to PSTN (I have the 2 Vonage lines which I
think not allowed to be connected from Asterisk unless getting a FXO/FXS card )
However, the Nortel reseller told that I need E1 card instead of T1 card
to connect between Asterisk and Nortel MICS because Mexico is using
E1.Anyway advice and information is welcomed ,
thanks.



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Re: [Asterisk-Users] Soporte

[jat]

Saludos:

No entiendo muy bien la pregunta, imagino que usted desea
tener los endpoints o gateways de manera NO dinamica, se
hace sencillamente declarando ese host en sip.conf sin
username/password y seteando la variable "host=ip.de.tu.gw"
y el el gatway se le dice que no se registre, so simple

You may check this site:
http://www.voip-info.org es un buen lugar para comenzar
it is a good place to start

PS:El idioma oficial de esta lista es ingles, so please
post in english your next help request, so you will have
more probabilities to find someone that will help you


> ALguien sabe como puedo:
> Si existe alguna manera para que el * trabaje con Sip a un gateway Cisco y
> que este ultimo no se tenga que registrar al * y ademas no tenga que
> configurar nada en el sip-ua del Cisco.
> Por las pruebas que hice no hay manera,siempre el ASterisk me termina
> pidiendo registracion, pero no saben de alguna direccion donde pueda
> respaldarme, es que mi jefe leyo algunos manuales donde con protocolo sip
> se puede trabajar sin necesidad de registrarse,  hasta donde he
> investigado en los foros me dicen lo mismo, es decir, que si o si tengo
> que registrarlo al *, pero no me indican alguna pagina donde pueda
> respaldarme o corregirme si es que estoy errado
> Gracias de antemano
>
>
>
> Atentamente
> Will Vélez
>
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[Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

[Douglas Garstang]

Folks,

When I have a dial string like this:

Dial(SIP/3254101&SIP/3254102,20,tr)

and I want to check the ${DIALSTATUS} variable after the dial, how do I know 
which number I am getting the variable for?

And, what about this?

Dial(SIP/3254101&SIP/[EMAIL PROTECTED],20,tr)

What happens in that case? How can I get the ${DIALSTATUS} variable for EACH 
NUMBER dialled?

Thanks,
Doug.
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Re: [Asterisk-Users] Re: Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4

[William M Conlon]

Thanks.  Useful 3rd party repository.  I've added it to my yum.repos.d

Question:  I see you spandsp, but I don't imagine that app_rxfax/ 
app_txfax are included in your asterisk rpm, so I'll still have to  
build * from source.  Google shows that someone has an asterisk- 
plugins rpm for Suse that includes the app_rxfax.so/app_txfax.so


I imagine others would find that of interest.

bill
On Apr 6, 2006, at 5:36 PM, Axel Thimm wrote:


On Wed, Apr 05, 2006 at 04:30:19PM -0700, William M Conlon wrote:

I was just getting to work on fax for my * system, so I thought I
would bring everything up to date since there would be some new
compilations involved.

yum update gave me kernel-2.6.16-1.2069_FC4

but after recompiling zaptel, I kept getting "FATAL module zaptel not
found"

Chased this for an hour with multiple recompiles and reboots.
Finally dropped back to 2.6.15-1.1833_FC4, which worked before, and
still works now.


You can try the packages at atrpms.net. They are built for
2.6.16-1.2069_FC4.
--
Axel.Thimm at ATrpms.net
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Re: [Asterisk-Users] asterisk box as a voip gateway

[Infobox Peru]

Your zaptel is wrong...

it must be:

zaptel.conf:
span=1,1,0,ccs,hdb3
dchan=16
bchan=1-15,17-31On 4/7/06, JP Carballo <[EMAIL PROTECTED]> wrote:
Mark Quitoriano wrote:> Hi Guys,>> Im configuring my asterisk box as a voip gateway. I have TE110P which> is connected on my PBX and i will be using voip for my outgoing.>> Here's my config
>> zaptel.conf:>> span=1,1,0,ccs,hdb3> fxoks=1-32>>> zapata.conf:>> context=default> signalling=fxs_ks> group=1> channel =>1-32
>> --> Regards,> Mark Quitoriano, CCNAWhat seems to be the problem?--JP Carballohttp://www.netfone2x.comBringing the world closer.
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Re: [Asterisk-Users] Asterisk compiling problems...

[Paul Hewlett]

On Monday 03 April 2006 22:14, George A. Michalopoulos wrote:
> Hello all,
>
> I just got the latest cvs snapshot and I'm trying to compile asterisk..
> When I run nefeli:/usr/src/asterisk/zaptel# make clean; make linux26
> Compile stops with errors...
>
> /usr/src/asterisk/zaptel/zaptel.c:6509: warning: passing argument 2 of
> 'class_device_create' makes pointer from integer without a cast
> /usr/src/asterisk/zaptel/zaptel.c:6509: warning: passing argument 3 of
> 'class_device_create' makes integer from pointer without a cast
> /usr/src/asterisk/zaptel/zaptel.c:6509: warning: passing argument 4 of
> 'class_device_create' from incompatible pointer type
> /usr/src/asterisk/zaptel/zaptel.c:6509: error: too few arguments to
> function 'class_device_create'
> /usr/src/asterisk/zaptel/zaptel.c:6510: warning: passing argument 2 of
> 'class_device_create' makes pointer from integer without a cast
> /usr/src/asterisk/zaptel/zaptel.c:6510: warning: passing argument 3 of
> 'class_device_create' makes integer from pointer without a cast
> /usr/src/asterisk/zaptel/zaptel.c:6510: warning: passing argument 4 of
> 'class_device_create' from incompatible pointer type
> /usr/src/asterisk/zaptel/zaptel.c:6510: error: too few arguments to
> function 'class_device_create'
> make[2]: *** [/usr/src/asterisk/zaptel/zaptel.o] Error 1
> make[1]: *** [_module_/usr/src/asterisk/zaptel] Error 2
> make[1]: Leaving directory `/usr/src/linux-2.6.15.6'
> make: *** [linux26] Error 2
>


I had a similar problem with the sirrix drivers.

The new gcc compiler treats as an error if one specifies a normal string when 
a format string (in the style of printf) is expected. This warning is 
triggered by the __attribute printf (I forget the exact syntax)  for the 
prototype in the appropriate header file.

This error is fixed by inserting an extra argument "%s" where the format 
string is expected e.g.

function proto is 

int myprintf( int fd, char *fmt, ...)

and it is being called by

myprintf( fd,"Some string")

which generates the warning. Fix by changing to

myprintf (fd, "%s", "Some string")

Paul
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RE: [Asterisk-Users] CallerID

[Technical Support]

Miles,

You can also download cid_rewrite from www.generationd.com  This PHP script
looks up the phone numbers in a local MySQL table, and/or uses reverse 411
on the web to lookup the name, and/or more options.

Michelle 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Friday, April 07, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
> Could you give me an example code of how this would work, and how to 
> setup the database, I'm pretty new and while what you have written 
> makes sense, and sounds like a good plan I'm not sure I can implement it.

I'm using my own agi-bin for "patching" callerid and adding the name if the
number is found in a table (a csv that is mantained with a spreadsheet), it
adds the name taken from this table. Then you can see the name in the
display of the phones.

--
Alejandro Vargas
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Re: [Asterisk-Users] asterisk box as a voip gateway

[JP Carballo]

Mark Quitoriano wrote:


Hi Guys,

Im configuring my asterisk box as a voip gateway. I have TE110P which 
is connected on my PBX and i will be using voip for my outgoing.


Here's my config

zaptel.conf:

span=1,1,0,ccs,hdb3
fxoks=1-32


zapata.conf:

context=default
signalling=fxs_ks
group=1
channel =>1-32

--
Regards,
Mark Quitoriano, CCNA


What seems to be the problem?

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?

[JP Carballo]

Ronald Wiplinger wrote:


#
# At this point we have a valid card and pin number.
#

if ($phoneno eq "RESET_INUSE") {
  &setinuse($carddata->{number}, 0);
  exit(0);
}

&checkexpired($carddata->{number});
&checkinuse($carddata->{number});
&setinuse($carddata->{number}, 1);


I put this into 682 in the extensions.conf
exten => 681,1,DeadAGI(astcc.agi,${CALLERID(num)},BALANCE,1)
exten => 681,2,Hangup
exten => 682,1,DeadAGI(astcc.agi,${CALLERID(num)},RESET_INUSE,2)
exten => 682,2,Hangup

As soon the flag is set, 682 will also tell you: The card number is in 
use, try later !


What do I miss?


There could be a call to checkinuse() before the RESET_INUSE routine.
If the RESET_INUSE flag is set, the routine should exit and not proceed 
to the following calls to

&checkexpired()
&checkinuse() and
&setinuse()

I suggest you check that the callerid you are using matches a card in 
the astcc db.


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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[Asterisk-Users] Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface

[CG]

I have gone through some archive about Nortel MICS (Meridian ?)+ Asterisk Integration but I'm not sure whether same as my case .70 telephone sets    |   |   Nortel MICS 
4.1  - Asterisk   |   PSTNI have read the David Gomillion's Guide and got the idea . However, my plan is slightly different from what he did , I need to use Nortel MICS to connect to PSTN (I have the 2 Vonage lines which I think not allowed to be connected from Asterisk unless getting a FXO/FXS card )
However, the Nortel reseller told that I need E1 card instead of T1 card to connect between Asterisk and Nortel MICS because Mexico is using E1.Anyway advice and information is welcomed , thanks.

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RE: [Asterisk-Users] suggestions on an IP T1 to TDM T1 gateway solution

[Damon Estep]

IAX or SIP trunks have similar
requirements, correct? You still need an IP link as transport. Granted IAX
might be more efficient.

 

Another thought is to simply use a Telco
provided point to point T1 to extend an open port on an existing PRI card in
the asterisk server to the remote PBX PRI interface. Anyone every done it this
way?  The need to pass other data over the T1 is not really an issue, it
would only be a backup data route anyways, primary use is voice.

 











From: Ryan Amos
[mailto:[EMAIL PROTECTED] 
Sent: Friday, April 07, 2006 10:00
AM
To: Damon Estep
Subject: RE: [Asterisk-Users]
suggestions on an IP T1 to TDM T1 gateway solution



 



You still need a CSU/DSU card. There are
plenty of data CSU/DSUs out there, I would suggest eBay. You may be able to
find a hardware CSU/DSU card somewhere. The Cisco router is undoubtedly cheaper
(you can pick up a used 2524 with a T1 card for $50; PCI cards are $500+) and
that is how many places still do their data T1s.

 

I much prefer this setup, actually, as it
is easier to break the T1 off and use it for other things (just plug it into a
VLANned switch and go to town.) You’re not really reducing your points of
failure either way, you’ve just moved it from hardware to software
(kernel drivers; plus zaptel does not behave well when the machine is under
load.)

 

Just curious, if you have a PRI and
you’re just talking to asterisk on both sides, why not use IAX trunks?

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Friday, April 07, 2006 8:15
AM
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: [Asterisk-Users]
suggestions on an IP T1 to TDM T1 gateway solution



 

Can anyone offer up a suggestion on a reliable and cost
effective customer premise hardware setup to be able to take an inbound IP T1
and deliver a PRI interface to a remote office?

 

Trying to reduce the amount of hardware required to
implement this, right now we use a Cisco router to take the IP T1 in on a
serial port and then we go Ethernet to a slimmed asterisk box with a single
port T1 card, and from there to the PBX PRI port.

 

Seems like we should be able to skip the router and build an
asterisk solution with 2 T1 ports, one for the data T1 and one to the PBX
(PRI), and then an Ethernet connection to the LAN/Firewall.

 

Maybe there is a non asterisk solution that works well with
asterisk?

 

The other end is asterisk as well.

 

Anyone done this successfully on a single device?

 

 








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RE: [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 9 11 /VOIP Providers

[Colin Anderson]

If you are a Canadian VoIP provider or CLEC, help make a difference by
joining the CAVP:

www.cavp.ca

>OMFG,

>I thought April Fools day was over.  This is hard to believe.  If true it
>tells me that Bell has not changed at all.  They are still trying to
>manipulate and take advantage of the parts of the market they have absolute
>control over.  The CRTC was right to continue to regulate them and leave
the
>VoIP providers alone. 
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[Asterisk-Users] Attended Transfer howto

[Miles Scruggs]

There is plenty of information on the wiki for setting asterisk up for 
transferring calls both from the Dail() command, and features.conf.


What really seems to be missing, is simply how do you actually perform 
the transfer?


Blind transfers are pretty simple as you only have two obvious steps.  
How though do you do attended transfers?


1.)  You have a call
2.)  You dial *2 or whatever you have setup in features.conf
3.)  You dial the ext of the person you want to bring into the call
4.) ?
5.) ?

Basically what do you have to dial, and bring the caller back into the 
call?  Also how do you get the call back if the ext your are attempting 
to transfer to is not avalible?


Thanks

Miles
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RE: [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 911 /VOIP Providers

[mustardman29]

OMFG,

I thought April Fools day was over.  This is hard to believe.  If true it
tells me that Bell has not changed at all.  They are still trying to
manipulate and take advantage of the parts of the market they have absolute
control over.  The CRTC was right to continue to regulate them and leave the
VoIP providers alone. 

> -Original Message-
> From: Bob's Leaky News Service [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, April 06, 2006 11:49 PM
> To: asterisk-users
> Subject: [Asterisk-Users] Bell Canada Requests $987.14 Rate 
> increase 911 /VOIP Providers
> 
> From the bend me over news department.
> 
> 
> 2 March 2006
> 
> 
> Mr. Leonard Katz
> Executive Director
> Broadcasting and Telecommunications
> Canadian Radio-television and
>   Telecommunications Commission
> Ottawa, Ontario
> K1A 0N2
> 
> 
> Dear Mr. Katz:
> 
> Associated with Bell Canada Tariff Notice No. 6929
> 
> 1.Attached for the Commission's approval are proposed revisions to
> Bell Canada's Access Services Tariff Item 315 – Zero-Dialed 
> Emergency Call Routing Service (0-ECRS) and General Tariff 
> Item 24 – Resale and Sharing as directed by the Commission in 
> Telecom Decision CRTC 2006-5.
>  The proposed modifications to the 0-ECRS tariff and the 
> associated Report of the Economic Evaluation related to 
> 0-ECRS further reflect a rate increase that will enable the 
> Company to recover its costs associated with the changes to 0-ECRS.
> 
> 2.In paragraph 59 of Telecom Decision 2006-5, VoIP 9-1-1 
> call routing
> (Decision 2006-5) the Commission directed Aliant Telecom 
> Inc., Bell Canada, MTS Allstream Inc. and Saskatchewan 
> Telecommunications to file tariff revisions to allow Voice 
> over Internet Protocol (VoIP) service providers access to 
> 0-ECRS at the same rates that are applicable to any other 
> service provider.
> 
> 3.In this application, Bell Canada (or the Company) is 
> proposing to
> revise Access Services Tariff Item 315 to make 0-ECRS 
> available to allow VoIP service providers access to 0-ECRS, 
> which enables eligible customers to route zero dialed 
> emergency calls from their end-customers to the designated 
> Public Safety Answering Point (PSAP) or other emergency 
> response agencies within Bell Canada's operating territory.
> 
> 4.In paragraph 59 of Decision 2006-5, the Commission 
> further directed
> Aliant Telecom Inc., Bell Canada, MTS Allstream Inc. and 
> Saskatchewan Telecommunications to maintain their respective 
> VoIP 9-1-1 call routing services, until their proposed tariff 
> revisions to 0-ECRS are approved by the Commission and their 
> customers of VoIP 9-1-1 call routing service have been 
> migrated to 0 ECRS.
> 
> 5.Bell Canada will maintain its VoIP 9-1-1 call routing services
> until its proposed tariff revisions to 0-ECRS under this 
> application are approved by the Commission, at which time the 
> Company will begin to migrate their customers of VoIP 9-1-1 
> call routing service to 0 ECRS.
> 
> 
> 6.Bell Canada will require 90 days to implement changes 
> to its 0-ECRS
> service that will enable the Company to provide the service 
> to VoIP service providers in a responsible manner.
> 
> 7.In paragraph 59 of Decision 2006-5, the Commission also directed
> Aliant Telecom Inc., Bell Canada, MTS Allstream Inc. and 
> Saskatchewan Telecommunications to include provisions within 
> their respective Resale and Sharing tariffs to explicitly 
> include in these tariffs the condition that local VoIP 
> service providers are to abide by the directives set out by 
> the Commission in paragraphs 52, 68, 93, 94 and
> 98 of Telecom Decision 2005-21, Emergency service obligations 
> for local VoIP service providers (Decision 2005-21).
> 
> 8.The Company is proposing to revise General Tariff Item 24 by
> explicitly including the condition stated in the above 
> paragraph in its Resale and Sharing tariff.
> 
> 9.In addition to the proposed revisions to the Company's 0-ECRS
> tariff as directed by the Commission in Decision 2006-5, the 
> Company is proposing to increase the 0-ECRS Set up Charge 
> service charge by $3.93.  The Company further proposes to 
> increase the 0 ECRS Access Charge monthly rate by $987.14.  
> These increases will offset the Company's costs associated 
> with the network upgrades required to facilitate additional 
> volume of VoIP 9-1-1 calls being routed from the customers of 
> VoIP service providers.
> 
> 10.   The Company notes that the modifications reflected in 
> the proposed
> tariff pages will be incorporated into the Company's 0-ECRS contracts.
> 
> 11.   The Bell Canada Report of the Economic Evaluation of 0-ECRS in
> support of the proposed revisions to the Company's 0-ECRS 
> tariff is provided as an Attachment.  As an Appendix to the 
> Bell Canada Report of the Economic Evaluation of 0-ECRS, the 
> Company is providing the imputation test associated with the 
> proposed revisions to the 0-ECRS tariff.
> 
> 

Re: [Asterisk-Users] transfer call after advise

[Christian B]

what you ask for is called "attended transfer".
asterisk can do it, but i don't use the manager API so i have no idea
how you can realize it.

regards
christian


On Fri, 7 Apr 2006 16:44:41 +0200
nik600 <[EMAIL PROTECTED]> wrote:

> i am developing a web application to manage callcenter, i will shortly
> release it on sf.net
> 
> this is my problem:
> 
> i will grant to users the possibility to transfer calls to other users
> using a web interface,
> 
> for example if SIP/200 is talking with SIP/400 who wants to transfer
> the call to SIP/500 i use this commands with manager API:
> 
> Action: Redirect\r\n
> Channel: SIP/200-sads\r\n
> ExtraChannel: 500\r\n
> Exten: 500\r\n
> Context: from-internal\r\n
> Priority: 1\r\n\r\n
> 
> this works fine (maybe the sintax now isn't correct... but it works),
> but my problem is that the call is immediately transferred to 500.
> 
> I'd like if:
> 
> 1 - 200 calls 400
> 2 - 400 want to transfer the call to 500
> 3 - 400 asks 500 if 500 wants to talk with 200
> 
> if 500 hangsup 200 still talk with 400
> if 400 hangsup 200 talks now with 500
> 
> is it possible?
> thanks nik
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Re: [Asterisk-Users] Beeps and noises during calls

[Zoa]

Some more info:

TALKOFF is the wrong recognition of DTMF component in human voice as 
true DTMF signal. This is an unavoidable factor since human voice always 
contain valid DTMF combination. Fortunately, presence of these valid 
DTMF components are unsteady. Unlike real DTMF generated from a 
touch-tone keyboard, these 'human' DTMF cannot maintain on a constant 
combination. So they can be isolated by DELAY discrimination. If a 
decoded DTMF signal can stay on constantly for certain duration which 
exceed those normal period experienced in human voice, then it can be 
identified as a real DTMF command.


taken from:

http://www.qsl.net/ve3rgw/dtmfsql.html


Zoa wrote:



Have a look at this :

http://www.dslreports.com/forum/remark,9151528

If anybody would have such a mitel or bellcore dtmf talkoff wav file, 
i have a very big email box you can drop it in :p


Zoa


Sean Garland wrote:


I have a very annoying problem that we hear on our end, but the other
party doesn't hear.  There are random beeps and echo type noises that
occur.  They are present during voicemails, and present on my end during
calls.  Is anyone experiencing the same deal?  I have asked this a
number of ways on the list, and never get a response... 
Thank you.



Sean Garland
Mount Shasta, CA
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[Asterisk-Users] wellgate registration 3802

[Jerry Geis]

I have a new wellgate 3802 unit. I have not gotten it to
register with asterisk 1.2.6.

My proxy setting is the correct IP in the 3802.
My security config is 1001/1001 and 1002/1002 on the wellgate (simple at 
this time).


My sip.conf has:

[wellgate3802L1]
type=friend
dtmfmode=inband
username=1001
secret=1001
host=dynamic
canreinvite=yes
nat=no
context=wellgate

[wellgate3802L2]
type=friend
dtmfmode=inband
username=1002
secret=1002
host=dynamic
canreinvite=yes
nat=no
context=wellgate

Apr  7 11:54:47 NOTICE[6288]: chan_sip.c:10879 handle_request_register: 
Registration from '' failed for '192.168.1.24' - 
Username/auth name mismatch
Apr  7 11:54:47 NOTICE[6288]: chan_sip.c:10879 handle_request_register: 
Registration from '' failed for '192.168.1.24' - 
Username/auth name mismatch


I am getting these two errors on the console. What have I missed that 
will let the

wellgate 3802 connect to asterisk?

Thanks,

jerry
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Re: [Asterisk-Users] Beeps and noises during calls

[Zoa]

Have a look at this :

http://www.dslreports.com/forum/remark,9151528

If anybody would have such a mitel or bellcore dtmf talkoff wav file, i 
have a very big email box you can drop it in :p


Zoa


Sean Garland wrote:


I have a very annoying problem that we hear on our end, but the other
party doesn't hear.  There are random beeps and echo type noises that
occur.  They are present during voicemails, and present on my end during
calls.  Is anyone experiencing the same deal?  I have asked this a
number of ways on the list, and never get a response...  


Thank you.


Sean Garland
Mount Shasta, CA
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RE: [Asterisk-Users] Cisco 7912 Phones & XML

[Ryan Amos]

This phone can be used
with the SCCP firmware and it does support XML services (you need a cisco
smartnet login to get the firmware.) However, the SCCP drivers available for
asterisk are not as mature (I have lots of random stability problems I can’t
track down,) and don’t have some key features like 3 way calling. 

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Mattina
Sent: Friday, April 07, 2006 10:17
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco
7912 Phones & XML



 

Friends,

http://www.voip-info.org/wiki/view/Cisco+7905%252F7912+IP+Phones
states under “SIP Software Limitations” that XML is not supported
on this phone.  However, my tech data summary of this product states: 

 

“In addition, XML applications deliver
impressive applications and network data to the Cisco IP Phone 7912G
display.”

 

Can someone please clear this up for me?

 

Thanks,

Adam Mattina
Networking & Systems Support
Layer 8 Group, Inc.
585.442.
[EMAIL PROTECTED]

 






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Re: [Asterisk-Users] Beeps and noises during calls

[Darrell Long]

We had a similar problem. Try turning relaxed DTMF off.

Darrell S. Long
BestWeb Corporation




Sean Garland wrote:

I have a very annoying problem that we hear on our end, but the other
party doesn't hear.  There are random beeps and echo type noises that
occur.  They are present during voicemails, and present on my end during
calls.  Is anyone experiencing the same deal?  I have asked this a
number of ways on the list, and never get a response...  


Thank you.


Sean Garland
Mount Shasta, CA
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Re: [Asterisk-Users] Look What 911 Will Cost in Canada

[John Novack]

Where is the problem?
Does one expect this service to be provided for free?
1600 bucks to set up a VOIP provider, and 2K per month sounds reasonable

John Novack


Bob's Leaky News Service wrote:


Check out the proposed prices when this is approved.



BELL CANADA REPORT


ON THE


ECONOMIC EVALUATION


FOR


THE TARIFF REVISION


OF


Bell Canada's Access Services Tariff Item 315 – Zero-Dialed

Emergency Call Routing Service (0-ECRS)



*2 March 2006




TABLE OF CONTENTS

Page

1.0 GENERAL 3
1.1 Purpose of the Study3
2.0 SERVICE DESCRIPTION 3
2.1 Service Characteristics 3
2.2 Service Benefits3
2.3 Marketing Considerations3
3.0 TARIFF CONSIDERATIONS   4
3.1 Tariff Components   4
3.2 Rate Determination Principles   4
3.3 Proposed Service Commencement Date  4
4.0 IMPUTATION TEST 4
5.0 DEMAND AND REVENUE INFORMATION  5
5.1 Forecast Assumptions and Methodology5
5.2 Number of Customers 5
5.3 Number of 0-ECRS Calls  5
5.4 Bell Canada Average 0-ECRS Call Duration5
5.5 Estimates of Demand Quantities  5
6.0 PHASE II COSTS  6
6.1 Study Assumptions   6
6.2 Study Period7
6.3 Financial Parameters and Tax Rates  7
6.4 Cost Inclusions 7
6.4.1   Expenses Causal to the Service  7
6.4.2   Capital Causal to the Service   8
6.4.3   Capital Causal to Demand8
6.4.4   Expenses Causal to Demand   8
6.4.5   Phase II Cost Summary   9
7.0 3RD PARTY ACQUISITION COSTS AND COSTS OF UNDERLYING CATEGORY I
COMPETITOR SERVICE COMPONENTS   9



1.0 GENERAL

1.1 Purpose of the Study

1.  The purpose of this study is to support the following revisions to
Bell Canada's (the Company's) Access Service Tariff 7516 (AST) Item
315 – 0-ECRS (Emergency Call Routing Service).

Telecom Decision CRTC 2006-5: VoIP 9-1-1 call routing directs Bell Canada to:

-  make 0-ECRS available to Voice over Internet Protocol Service
Providers (VoIPSPs) who register as resellers with the CRTC.

-  offer 0-ECRS to VoIPSPs who are registered as resellers with the
CRTC at the same rate it is offered to other eligible parties -
Wireless Service Providers (WSPs), Canadian Pay Telephone Service
Providers (CPTSPs), Alternate Operator Service Providers (AOSPs),
Competitive Local Exchange Carriers (CLECs) and Interexchange Carriers
(IXCs).

-  provide the Call Routing Lists and Traffic Operator Position
Records (TOPR) lists that are currently provided to traditional 0-ECRS
customers.


2.0 SERVICE DESCRIPTION

2.  The revision to the 0-ECRS Service is to allow VoIPSPs who are
registered as resellers with the CRTC to access Bell Canada's 0-ECRS. 
Using 0-ECRS, VoIPSPs will be able to route fixed non native and

nomadic 9-1-1 VoIP calls to Primary 9-1-1 Public Safety Answering
Points (PSAPs).

2.1 Service Characteristics

3.  Bell Canada will provide VoIPSPs with a Call Routing List, a TOPR
list and an authorization PIN number under the terms of 0-ECRS. 
VoIPSPs will be responsible for providing a call answer centre to

perform location determination of a 9-1-1 VoIP caller.  The VoIPSP
call answer centre will then use the Call Routing List or TOPR list to
automatically route the call to a Primary Public Safety Answering
Point (PSAP) without Bell Canada Operator assistance.

2.2 Service Benefits

4.  The revision to the 0-ECRS will enable VoIPSPs to provide basic
9-1-1 service in Bell Canada territories.

2.3 Marketing Considerations

5.  Potential customers are currently WSPs, CPTSPs AOSPs, CLECs and
IXCs.  New target customers are VoIPSPs that are registered as local
resellers with the CRTC.


3.0 TARIFF CONSIDERATIONS

3.1 Tariff Components

6.  The following rates and charges apply to 0-ECRS:

Tariff Components   Monthly RateService Charge

Set-up Charge, per customer N/A $1,658.09
Access Charge, per customer $2011.15N/A

7.  This service is provided initially to the customer under a two-year
contract under the terms and conditions of which are specified in the
0-ECRS agreement and is renewed on a successive one-year term basis.

3.2 Rate Determination Principles

8.  The proposed tariff rate(s) for 0-ECRS is based on Phase II costs
plus a 15% mark-up as per the Commission's determinations at paragraph
231 of Regulatory framework for second Price Cap, Telecom Decision
CRTC 2002-34, 30 May 2002, for Category I competitor services.

3.3 Proposed Service Commencement Date

9.  The Company is proposing to introduce Access Service Tariff 7516
(AST) Item 315 –0 ECRS to VoIPSP's on 1 May 2006.


4.0 IMPUTATION TEST

10. The imputation test associated with 0-ECRS has been met in
accordance with the imputation test methodology as set out in the
November 1998 Commission letter which was subsequently amended by
Issues related to imputation test methodology 

[Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 04/08/2006

[Shane Young]

SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC
   "Keep in touch with the World"

Hello,

The next Asterisk Users Group meeting has been scheduled for this Saturday 
March 11th at 11:30am.

Meetings are held monthly on the second Saturday of each month, excluding July 
and December.  The
Agenda is posted online
http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda

Meetings are held at Sound Choice Communications LLC...
http://maps.google.com/maps?oi=map&q=7839%2012th%20Ave%20S%2055425

Sound Choice Communications is located in Bloomington Minnesota, just 1/2 mile 
west of the Mall of
America. The address is: 7839 12th Ave S, Bloomington Minnesota 55425.  We are 
just south of
Interstate 494 on 12th Ave.  12th Avenue is one exit west of Hwy 77 (Ceder Ave).

This is the Semi-Annual New Asterisk Users meeting .  If you want to learn how 
to install asterisk
on your system, this is the meeting to attend.

If you're having a problem with Asterisk, bring your questions to a meeting for 
free help. We love
helping new users!

Come to a meeting to meet other asterisk users, see asterisk solutions, win a 
door prize, eat food,
or for the good company, to look for work, if your looking for employees, to go 
out for a drive, to
get out of your house, whatever, JUST COME TO THE MEETING!

New visitors can help themselves to FREE FXO Interface cards (So you can 
connect your phone line,
and have a timing source for meetme and IAX protocols). Some members have been 
known to swap
hardware at the meetings. Have extra VoIP gear, looking for VoIP gear?  There's 
plenty of
hardware to see. Have you been to a meeting recently?

Please come and share your own ideas and learn from others. As always, free 
food.


We are always looking for help with meeting topics. If you feel like taking the 
lead, please do and
simply let me know if you need anything.

Meeting starts at 11:30am and parking is available in the rear of the building. 
Runs about 2 hours
or less, and we'll order Pizza to the meeting for lunch.

Look forward to seeing you there.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA


If you have a product or service you'd like to introduce to our members, send a 
private message to
ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next 
month's sponsor.



This message was sent using IMP, the Internet Messaging Program.
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[Asterisk-Users] Beeps and noises during calls

[Sean Garland]

I have a very annoying problem that we hear on our end, but the other
party doesn't hear.  There are random beeps and echo type noises that
occur.  They are present during voicemails, and present on my end during
calls.  Is anyone experiencing the same deal?  I have asked this a
number of ways on the list, and never get a response...  

Thank you.


Sean Garland
Mount Shasta, CA
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[Asterisk-Users] Fedora 'service asterisk start' problems

[Bob McDowell]

I ran into a weird one last night.  If I use 'service asterisk start' I
have problems (see below).  If I exclusively use 'asterisk
-c' everything works normally.

It happens like this:

1) 'service asterisk start'
2) Use asterisk normally, etc, etc - eventually change something that
requires a restart
3) Issue either a CLI 'stop now' or a 'service asterisk stop', and
asterisk stops
4) 'service asterisk start'
5) Asterisk enters a death loop reporting 'Asterisk exited with code 1'
over and over again
6) Switch to another session and issue 'service asterisk stop' about a
dozen times, and it stops.
7) Death-loop resumes when starting asterisk with either method
8) After a reboot, things are normal again

Weird, eh?  It's not critical, but if you've seen it before I'd love to
know what you found to be causing it.


Bob McDowell






   *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION ***


This e-mail message and all attachments, if any, may contain confidential and 
privileged material and are intended only for the intended recipient.  Any 
unauthorized review, use, disclosure or distribution is prohibited.  If you are 
not the intended recipient, please contact the sender by reply e-mail or by 
calling  (417) 869-9192 and destroy the original and any copies of this e-mail.
 
 


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[Asterisk-Users] Cisco 7912 Phones & XML

[Adam Mattina]

Friends,

http://www.voip-info.org/wiki/view/Cisco+7905%252F7912+IP+Phones
states under “SIP Software Limitations” that XML is not supported
on this phone.  However, my tech data summary of this product states: 

 

“In addition, XML applications deliver impressive
applications and network data to the Cisco IP Phone 7912G display.”

 

Can someone please clear this up for me?

 

Thanks,

Adam
Mattina
Networking & Systems Support
Layer 8 Group, Inc.
585.442.
[EMAIL PROTECTED]

 






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[Asterisk-Users] editing the asterisk -addons makefile

[Jordan Novak]

Can someone point me to some documentation on how to add
app_CBMysql.c to my makefile. I also am a little unsure of the directions on
how to compile it.

 

Here is what I am working with…

 

 . Download
and compile app_cbmysql in /usr/src/asterisk/apps or wherever you have the
Asterisk source. Run as root:
cd /usr/src/asterisk/apps
wget http://www.fitawi.com/Asterisk/app_cbmysql.c
(site is
currently down)
Edit the Makefile in that folder using the patch
www.fitawi.com/Asterisk/Makefile-cbmysql-patch.txt  (site is currently down) Compile Asterisk : run make install in
the Asterisk source directory (not the subdirectory apps). In this way you
will compile only app_cbmysql.c and not all the other parts of Asterisk.

 

 

Fitawi.com is
out of commission for now.

 

Jordan Novak

Communications Technician

 






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[Asterisk-Users] Uplink Skype2Sip

[Giordano Grandis]

Hi 
all,
anyone get it worked 
? Uplink route me the call incoming from skype but when i answer, my skype go in 
error on sound card ?
I also set in my 
hosts this value:
 
127.0.0.1 
 pgp01.televolution.net 127.0.0.1  stun01.sipphone.com 

 
This is my 
sip.conf
 
[skype]language 
= itusername = skypesecret = host = 
dynamicdefaultip = 
port = 5060type 
= friendcontext = from_ethcanreinvite = yesdtmfmode = 
infocallgroup = 1pickupgroup = 1fromuser = 
insecure = veryqualify = yescallerid 
= Test <999>allow = all
Thanks 
all
 
Giordano
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[Asterisk-Users] transfer call after advise

[nik600]

i am developing a web application to manage callcenter, i will shortly
release it on sf.net

this is my problem:

i will grant to users the possibility to transfer calls to other users
using a web interface,

for example if SIP/200 is talking with SIP/400 who wants to transfer
the call to SIP/500 i use this commands with manager API:

Action: Redirect\r\n
Channel: SIP/200-sads\r\n
ExtraChannel: 500\r\n
Exten: 500\r\n
Context: from-internal\r\n
Priority: 1\r\n\r\n

this works fine (maybe the sintax now isn't correct... but it works),
but my problem is that the call is immediately transferred to 500.

I'd like if:

1 - 200 calls 400
2 - 400 want to transfer the call to 500
3 - 400 asks 500 if 500 wants to talk with 200

if 500 hangsup 200 still talk with 400
if 400 hangsup 200 talks now with 500

is it possible?
thanks nik
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Re: [Asterisk-Users] VPB cannot call out

[Dovid Bender]

check the settings between the phone and asterisk.hensem boy <[EMAIL PROTECTED]> wrote:  Hi DovidActually I dont how to set up my DTMF. Anyway here is the setting :-/etc/vpb/vtcore.conf[general]name = vtcorechannels=12cards=2[card0]type=openpcichannels=4hwplaygain=12hwrecordgain=-12chan = 0/etc/asterisk/vpb.conf[general]type = v4pcicards = 1[interfaces]board = 1echocancel = oncontext = from-pstnUseLoopDrop = 0mode = fxochannel = 0Using this setting, I can get the call. But when I tried to call out, it looks like it didnt set the DTMF. Do I need to configure any bal or txgain and rxgain setting? If so, what should I do? Thanks.Dovid Bender <[EMAIL PROTECTED]> wrote: 
 Check your DTMF Settings.--- hensem boy <[EMAIL PROTECTED]>wrote:> Hi all> > I have a problem when I want to call out using VPB> trunk line, it cannot send the DTMF. Is there anyone> has the same problem? Please share with me the> solution.> > Thanks.> > > -> New Yahoo! Messenger with Voice. Call regular phones> from your PC and save big.>___> --Bandwidth and Colocation provided by Easynews.com> --> > Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> >http://lists.digium.com/mailman/listinfo/asterisk-users> __Do You Yahoo!?Tired of
 spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users  Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] OT: local calling guide

[Joseph Tanner]

It's still up for me.  They did get a new domain, which currently just
redirects to the old site, but it may be a good idea to update your
bookmarks anyways in case they have it redirect to a different site in
the future.

http://www.localcallingguide.com/

Joseph Tanner

On 4/7/06, Jonathan k. Creasy <[EMAIL PROTECTED]> wrote:
> Anyone know what has happened to the local calling guide?
>
> http://members.dandy.net/~czg/search.html
>
> -Jonathan
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[Asterisk-Users] Re: IVR: Cant hear my message

[Dovid Bender]

I cant figure it out but why dont you dont you make an extension that you can dial and record yoursef.     Exten => 100,1,Wait(2)  Exten => 100,2,Record(FileName:gsm)  Exten => 100,3,Wait(2)  Exten => 100,4,Playback(FileName)  Exten => 100,5,Hangup  Antoine LOUIS <[EMAIL PROTECTED]> wrote:  Hello,I've reccorded a voice message for the IVR. (.wav, 16 bits, 8kHz)The file is /var/lib/asterisk/sound/11ivrrecording.wav.When asterisk (1.2.5) starts this file i can't hear it on my phone.Here is the log : Apr  6 17:00:16 VERBOSE[845] logger.c: -- Executing SetCallerID("SIP/11-97b9", ""Patrice" <11>") in new stackApr  6 17:00:16 VERBOSE[845]
 logger.c: -- Executing NoOp("SIP/11-97b9", "Using CallerID "Patrice" <11>") in new stack Apr  6 17:00:16 VERBOSE[845] logger.c: -- Executing Playback("SIP/11-97b9", "11ivrrecording") in new stackApr  6 17:00:16 DEBUG[845] channel.c: Scheduling timer at 160 sample intervalsApr  6 17:00:16 VERBOSE[845] logger.c: -- Playing '11ivrrecording' (language 'en')Apr  6 17:00:17 DEBUG[26916] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED] ' of Response 2: Match FoundApr  6 17:00:49 DEBUG[26916] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Apr  6 17:00:50 DEBUG[845] channel.c: Scheduling timer at 0 sample intervalsApr  6 17:00:50 VERBOSE[845] logger.c:   == Spawn extension (from-internal, *99, 2) exited non-zero on 'SIP/11-97b9'Anyone has an idea ? Thanks a lot.Antoine___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] CallerID

[Colin Anderson]

I do the same as well. From my SQL server, I have my Asterisk box query my
customer and B2B contacts using ODBCSock and compose them as as CSV on the
Asterisk box; a script then parses the CSV and DBPut's them into Asterisk
itself. The nice thing about it is you can modify the CallerID with rich
data, for example, when a customer calls, I prepend the customer ID number
for our CRM into the CallerID so the staff member can type in the customer
ID in to the CRM before they pick up. 

I have an .awk that parses the CSV and DBPut's it into Asterisk, if you are
interested email me offlist. 

-Original Message-
From: Alejandro Vargas [mailto:[EMAIL PROTECTED]
Sent: Friday, April 07, 2006 2:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID


2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
> Could you give me an example code of how this would work, and how to
> setup the database, I'm pretty new and while what you have written makes
> sense, and sounds like a good plan I'm not sure I can implement it.

I'm using my own agi-bin for "patching" callerid and adding the name
if the number is found in a table (a csv that is mantained with a
spreadsheet), it adds the name taken from this table. Then you can see
the name in the display of the phones.

--
Alejandro Vargas
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[Asterisk-Users] regexp in gotoif

[Christian B]

Hello!

this is a short one: in a gotoif-statement i would like to match a
variable to a number, where the number could have digits from 2-6.
asterisk only seems to be capable to match such a digit-range when used
in the extension, but not in a regexp, at least the following query
doesn't work:


exten => _X.,1,GotoIf($[${EXTEN} : 234[2-6]]?jump:)


obviously asterisk has a problem with this since it parses the brackets
differently, i've also tried to do it like:


exten => _X.,1,GotoIf($[${EXTEN} : 234\[2-6\]]?jump:)


but of course this didn't work(was a desperate try anyways).

is there a way to match a range like it is possible with unix regexps
(a-zA-Z1-9)?

thanks for your replies!

christian
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Re: [Asterisk-Users] queue/agent and macros?

[Johann]

I set the callerid name to show the employee that will get the call what kind of 
call it is.  We have multiple options, but actually only use 2 queues(with most 
people answering both of them).  Eventually it may be expanded so there are more 
queues with different people on it...but this way when that comes there will be 
little if any chance for people calling in.



--johann

Gareth Blades wrote:

Cant you set the calleridname before putting the call into the queue?

On Thu, 2006-04-06 at 22:57, Shaun wrote:

I was wondering if it was possible to run a macro once the agent/member 
picks up, I know I can do this with dial in the extensions.conf but wasn't 
sure about the queue.  Basically I have a macro that identifies the caller 
and need that run.



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RE: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]

[Colin Anderson]

I considered that but the Linux boxes also give me the flexibility to
traffic-shape with AstShape as well as use IAX trunking; to use the remote
Linux box as a SIP demarcation point with IAX back hauled to the primary
LAN. Also, in the event of subnet congestion, I can subnet the remote LAN
and use the Linux box as a router, it's pretty trivial to do. Haven't got to
that point yet since just running SIP over it seems fine right now, but I
can't preclude it in the future, my users are very bandwidth-hungry and
their needs grow and grow daily, using Linux gives me finer-grained control
over how these services are delivered. 

Besides, hey, it's Linux, how cool is that?

-Original Message-
From: Ronan Mullally [mailto:[EMAIL PROTECTED]
Sent: Friday, April 07, 2006 2:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600
metre s away with Cat 3 or telco wire [long]



On Fri, 7 Apr 2006, Ronan Mullally wrote:

> Why not just install an ethernet switch on both ends that supports
> trunking / etherchannel?  Less configuration, less chance for operator
> error, and no hard disks.  You'll most likely also need a switch on
> each end *anyway*...

Before I get toasted through and through - you would still need the VDSL
extenders of course, but the Linux boxes seem like overkill...


-Ronan
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RE: [Asterisk-Users] Inbound PRI calls drop after 5 seconds using

[Wai Wu]

I am surprised that his was able to make outbound calls. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Friday, April 07, 2006 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Inbound PRI calls drop after 5 seconds
using

Joe Prosser wrote:
> signalling = pri_net
>   
Shouldn't this be signalling = pri_cpe?

> Switchtype: National ISDN
> Type: CPE
>   

Since you have this also as CPE.

Doug

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[Asterisk-Users] OT: local calling guide

[Jonathan k. Creasy]

Anyone know what has happened to the local calling guide?

http://members.dandy.net/~czg/search.html

-Jonathan
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Re: [Asterisk-Users] digium card for xseries 346

[Kevin P. Fleming]

Mark Quitoriano wrote:

> What model can i use for an xseries 346 server, i think the pci slot is
> 64-bit? Im just going to use it for asterisk timing so the cheapest will be
> the best.

The x346 has PCI-X slots that supply 3.3V signaling, so any Digium 3.3V
compatible card will work just fine.
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Re: [Asterisk-Users] Inbound PRI calls drop after 5 seconds using

[Doug Lytle]

Joe Prosser wrote:

signalling = pri_net
  

Shouldn't this be signalling = pri_cpe?


Switchtype: National ISDN
Type: CPE
  


Since you have this also as CPE.

Doug

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Re: [Asterisk-Users] transforming g729 files to wav files

[Tofik Suleymanov]

Noah Miller wrote:

Hi Tofik -



is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on:  http://www.asteriskguru.com/audio_conversion.php



The wiki also lists GX::Transcoder which looks like it can do g729 to
wav, though I've never tried it.  Here's a link:

http://www.germanixsoft.de/index.php

Otherwise, you could probably rig up asterisk to transcode from g729
to another codec then record it to a file.

There's probably not more tools to do this since most people aren't
interested in going from the very lossy g729 codec to the non-lossy
wav format.

I am aware of this tool, but is there any other tool (same features like 
 mentioned gx::transcoder) for unix-like operating systems ?



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Re: [Asterisk-Users] transforming g729 files to wav files

[Tofik Suleymanov]

Darrell Long wrote:
The resulting file is not going to sound any better and its going to 
take up more space. What is the reason you need a WAV file? Perhaps 
there is a more efficient way to do what you are trying to do.


Darrell S. Long
BestWeb Corporation

  


I understand issues about sound quality.Here is the situation:

i am using g729-native sound files and g729 codecs everywhere.My 
voicemail is coming in g729 format also.Some time ago one of our 
customers asked for the voicemail to go to his e-mail and i want him to 
recieve just a .wav file.


I've also tried to use:
format=g729|wav

in my voicemail.conf in order to have copies of voicemails in wav format 
but for unknown reason (after this change) i wasnt able to hear 
voicemail announcements when trying to access voicemail.

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[Asterisk-Users] Inbound PRI calls drop after 5 seconds using Sangoma A101

[Joe Prosser]

uot;) in new stacklogger.c: -- Goto (from-pstn-reghours,s,1)pbx.c: _expression_ result is '1'logger.c: -- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours-nofax|s|1:2") in new stack
logger.c: -- Goto (from-pstn-reghours-nofax,s,1)logger.c: -- Executing SetVar("Zap/1-1", "intype=EXT-211") in new stacklogger.c: -- Executing Cut("Zap/1-1", "intype=intype|-|1") in new stack
pbx.c: _expression_ result is '1'logger.c: -- Executing GotoIf("Zap/1-1", "1?4:5") in new stacklogger.c: -- Goto (from-pstn-reghours-nofax,s,4)logger.c: -- Executing Goto("Zap/1-1", "ext-local|211|1") in new stack
logger.c: -- Goto (ext-local,211,1)logger.c: -- Executing Macro("Zap/1-1", "exten-vm|211|211") in new stacklogger.c: -- Executing Macro("Zap/1-1", "user-callerid") in new stack
logger.c: -- Executing DBget("Zap/1-1", "AMPUSER=DEVICE/617733/user") in new stacklogger.c: -- DBget: varname=AMPUSER, family=DEVICE, key=617733/userdb.c: Unable to find key '6177332750/user' in family 'DEVICE'
logger.c: -- DBget: Value not found in database.logger.c: -- Executing DBget("Zap/1-1", "AMPUSERCIDNAME=AMPUSER//cidname") in new stacklogger.c: -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname
db.c: Unable to find key '/cidname' in family 'AMPUSER'logger.c: -- DBget: Value not found in database.pbx.c: _expression_ result is '1'logger.c: -- Executing GotoIf("Zap/1-1", "1?5") in new stack
logger.c: -- Goto (macro-user-callerid,s,5)logger.c: -- Executing NoOp("Zap/1-1", "Using CallerID 617733") in new stacklogger.c: -- Executing SetVar("Zap/1-1", "FROMCONTEXT=exten-vm") in new stack
logger.c: -- Executing Macro("Zap/1-1", "record-enable|211|IN") in new stackpbx.c: Function result is '0'logger.c: -- Executing GotoIf("Zap/1-1", "0 > 0?2:4") in new stack
logger.c: -- Goto (macro-record-enable,s,4)logger.c: -- Executing AGI("Zap/1-1", "recordingcheck|20060407-091432|1144415672.41") in new stacklogger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
logger.c:   recordingcheck|20060407-091432|1144415672.41: Inbound recording not enabledlogger.c: -- AGI Script recordingcheck completed, returning 0logger.c: -- Executing NoOp("Zap/1-1", "No recording needed") in new stack
logger.c: -- Executing Macro("Zap/1-1", "dial|15|tr|211") in new stackpbx.c: _expression_ result is '0'logger.c: -- Executing GotoIf("Zap/1-1", "0?4:2") in new stack
logger.c: -- Goto (macro-dial,s,2)pbx.c: Function result is '0'pbx.c: _expression_ result is '0'logger.c: -- Executing GotoIf("Zap/1-1", "0?5:4") in new stacklogger.c: -- Goto (macro-dial,s,4)
logger.c: -- Executing AGI("Zap/1-1", "dialparties.agi") in new stacklogger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agilogger.c: < Protocol Discriminator: 
Q.931 (8)  len=9logger.c: < Call Ref: len= 2 (reference 59/0x3B) (Originator)logger.c: < Message type: RELEASE (77)logger.c: < [Apr  7 09:14:32 VERBOSE[16191] logger.c: < [08Apr  7 09:14:32 VERBOSE[16191] 
logger.c: < [08 02Apr  7 09:14:32 VERBOSE[16191] logger.c: < [08 02 82Apr  7 09:14:32 VERBOSE[16191] logger.c: < [08 02 82 86Apr  7 09:14:32 VERBOSE[16191] logger.c: < [08 02 82 86]logger.c: < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Public network serving the local user (2)
logger.c: <  Ext: 1  Cause: Unknown (6), class = Normal Event (0) ]logger.c: -- Processing IE 8 (cs0, Cause)logger.c: -- Channel 1/1, span 1 got hangupres_agi.c: Zap/1-1 hunguplogger.c
:   == Spawn extension (macro-dial, s, 4) exited non-zero on 'Zap/1-1' in macro 'dial'logger.c:   == Spawn extension (macro-exten-vm, s, 4) exited non-zero on 'Zap/1-1' in macro 'exten-vm'logger.c:   == Spawn extension (ext-local, 211, 1) exited non-zero on 'Zap/1-1'
cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2006-04-07 09:14:32','6177332750','6177332750','211','ext-local', 'Zap/1-1','','AGI','
dialparties.agi',0,0,'NO ANSWER',3,'','1144415672.41')chan_zap.c: Set o

[Asterisk-Users] suggestions on an IP T1 to TDM T1 gateway solution

[Damon Estep]

Can anyone offer up a suggestion on a reliable and cost
effective customer premise hardware setup to be able to take an inbound IP T1
and deliver a PRI interface to a remote office?

 

Trying to reduce the amount of hardware required to
implement this, right now we use a Cisco router to take the IP T1 in on a
serial port and then we go Ethernet to a slimmed asterisk box with a single
port T1 card, and from there to the PBX PRI port.

 

Seems like we should be able to skip the router and build an
asterisk solution with 2 T1 ports, one for the data T1 and one to the PBX (PRI),
and then an Ethernet connection to the LAN/Firewall.

 

Maybe there is a non asterisk solution that works well with
asterisk?

 

The other end is asterisk as well.

 

Anyone done this successfully on a single device?

 

 






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Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]

[Andrew Kohlsmith]

On Thursday 06 April 2006 17:44, Colin Anderson wrote:
> I was given the challenge recently of creating a LAN-LAN bridge between two
> buildings several
> hundred metres from each other, using only existing Cat 3 wiring and
> without having to resort
> to an expensive and finicky 5 Ghz wireless link. I was able to create a 90
> megabit link for
> about $3,000 Cdn with new PC's, CentOS 4.1, and the newly avaliable Black
> Box VDSL Ethernet
> Extender, which supports 30 megabits over a single twisted pair.

I have to ask, what was wrong with a pair of media converters ($200/pair new, 
$50/pair on ebay) and some cheap-as-dirt multimode fiber?  Isolated, 100mbit 
and easily, easily gangable. Was the goal simply to get as fast as possible 
with regular copper wire, or was there a bigger objective?

I do appreciate the effort put into this, though, and more than anything I 
appreciate your posting it here for others.  I sincerely thank you for that.

-A.
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Re: [Asterisk-Users] IVR : Can't hear my message

[Dovid Bender]

I cant figure it out but why dont you dont you make an extension that you can dial and record yoursef.     Exten => 100,1,Wait(2)  Exten => 100,2,Record(FileName:gsm)  Exten => 100,3,Wait(2)  Exten => 100,4,Playback(FileName)  Exten => 100,5,Hangup  Antoine LOUIS <[EMAIL PROTECTED]> wrote:  Hello,I've reccorded a voice message for the IVR. (.wav, 16 bits, 8kHz)The file is /var/lib/asterisk/sound/11ivrrecording.wav.When asterisk (1.2.5) starts this file i can't hear it on my phone.Here is the log : Apr  6 17:00:16 VERBOSE[845] logger.c: -- Executing SetCallerID("SIP/11-97b9", ""Patrice" <11>") in new stackApr  6 17:00:16 VERBOSE[845]
 logger.c: -- Executing NoOp("SIP/11-97b9", "Using CallerID "Patrice" <11>") in new stack Apr  6 17:00:16 VERBOSE[845] logger.c: -- Executing Playback("SIP/11-97b9", "11ivrrecording") in new stackApr  6 17:00:16 DEBUG[845] channel.c: Scheduling timer at 160 sample intervalsApr  6 17:00:16 VERBOSE[845] logger.c: -- Playing '11ivrrecording' (language 'en')Apr  6 17:00:17 DEBUG[26916] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED] ' of Response 2: Match FoundApr  6 17:00:49 DEBUG[26916] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Apr  6 17:00:50 DEBUG[845] channel.c: Scheduling timer at 0 sample
 intervalsApr  6 17:00:50 VERBOSE[845] logger.c:   == Spawn extension (from-internal, *99, 2) exited non-zero on 'SIP/11-97b9'Anyone has an idea ? Thanks a lot.Antoine___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]

[BJ Weschke]

On 4/6/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
> I was given the challenge recently of creating a LAN-LAN bridge between two
> buildings several
> hundred metres from each other, using only existing Cat 3 wiring and without
> having to resort
> to an expensive and finicky 5 Ghz wireless link. I was able to create a 90
> megabit link for
> about $3,000 Cdn with new PC's, CentOS 4.1, and the newly avaliable Black
> Box VDSL Ethernet
> Extender, which supports 30 megabits over a single twisted pair.
>
> This is relevant to the list because I have seen many posts with people
> facing the same
> kind of challenge deploying Asterisk in remote locations. In my case, I am
> running
> ~40 Snom 360's from the remote building to where my Asterisk server is, and
> it's working
> fine.
>
> Hope this helps someone. Yay Linux.
>
> 
> ---
> HOWTO: Create a bonded Ethernet link over common Cat 3 or telco-grade cable
> up to 1.9 km
> 
> ---
>
> Abstract: Using the newly-available Black Box LB300A VDSL Ethernet Extenders
> and Linux bonding,
> it is possible to create a protocol-independent, redundant, high speed
> Ethernet bridge using common
> Cat 3 or telco-grade cabling at ranges from 600 metres (1,970 feet) to 1.9
> km (6,233 feet),
> with bandwidth ranging from 90 megabits to 3 megabits. The characteristic of
> the link is that of
> a regular bridged LAN, and is suitable for high speed or latency sensitive
> applications such as Voice
> over IP.
>
> HARDWARE:
>
> 2 X 4-PCI slot PC's
> 6 X Black Box LB300A VDSL Ethernet Extenders
> 8 X PCI Ethernet cards
> http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946
> RJ-12 crimper
> RJ-12 6-conductor plugs (will work with RJ-11)
>
> SOFTWARE:
>
> CentOS 4.1 or any Linux distro that supports bonding and bridging
>
>
> 1. Install hardware on both machines
>
> In my case, I used an ASUS A8V with an Athlon XP 3000+ with 4) 3C905CX
> cards. The reason I selected
> this board was because there is 5 avaliable PCI slots. Install the cards 0-3
> in slots 0-3.
>
> 2. Disable onboard hardware
>
> Because the NIC's will require an IRQ each, disable the following resources:
>
>a) Onboard NIC
>b) Onboard sound
>c) Onboard USB
>d) Onboard legacy ports
>e) Set "F1 on error" or "halt on errors" to off in the BIOS so you
> can boot headless
>
> 3. Install your OS
>
> I chose CentOS 4.1 as my distro, I really like it. I installed CentOS as a
> "server" with everything
> disabled except "development". I also set the detected NIC's to
> "unconfigured" - no DHCP,
> no start at boot.
>
> 4. Boot. Do an ifconfig eth0, eth1, eth2, eth3 to make sure all of the NIC's
> are running OK
>
> 5. Determine which NIC is eth0, eth1, eth2, eth3. eth0 will be the interface
> to the LAN and
> eth1,2,3 will be the bonded link. Both the bonded link and the LAN interface
> will be bridged together.
>
> Easy enough - plug in a network cable into a NIC and run dhclient. Once you
> get an IP from your
> DHCP server you can use ifconfig to determine whihc one got the IP. Repeat
> for each NIC. Once you
> have detemined which physical device is which, I used a sharpie to label
> each port so I wouldn't
> get confused.
>
> 6. Install bridging support, if your distro does not support it OOB.
> Fortunately, CentOS does so
> no problem there.
>
> 7. Install bridge-utils to configure the bridge. Cake in CentOS: "yum
> install bridge-utils"
>
> 8. Configure the bonding config
>
> In /etc/sysconfig/network-scripts make valid entries for your bonded device
> and each slave.
>
> ifcfg-bond0:
>
> DEVICE=bond0
> ONBOOT=no
> BOOTPROTO=none
>
> ifcfg-eth1/2/3:
>
> DEVICE=eth1/2/3 (replace as nessisary)
> USERCTL=no
> ONBOOT=yes
> MASTER=bond0
> SLAVE=yes
> BOOTPROTO=none
>
> 9. Startup script /etc/rc.d/rc.local:
>
> modprobe bonding miimon=1*
> brctl addbr br0
> brctl addif br0 eth0
> sleep 10s**
> brctl addif br0 bond0
>
> *miimon=1 means disable the port after one second on link down. This is the
> redundant part. If
> one of your links fail, the other two will keep working.
>
> ** The 'sleep' is to allow the bridge to stabilize. In testing, the bridge
> did not work if each
> brctl statement immediately followed each other.
>
> 10. Test without VDSL extenders
>
> Obtain or make 3 X Ethernet crossover cables. Plug 1 PC's eth0 port into the
> source LAN. Plug the
> 3 X cross over cables into eth1,2,3 on both PC's. Plug a device into the 2nd
> PC's eth0 with a
> crossover cable, or plug the second PC into a switch then plug your devices
> into the same switch.
> Make sure both PC's have a keyboard and monitor so you can see what's going
> on. Finally, plug
> crossover cables into each PC's eth1, eth2 & eth3
>
> On both PC's run tcpdump -i br0 from the console . In my case, I saw traffic
> right away

Re: [Asterisk-Users] queue/agent and macros?

[Gareth Blades]

Cant you set the calleridname before putting the call into the queue?

On Thu, 2006-04-06 at 22:57, Shaun wrote:
> I was wondering if it was possible to run a macro once the agent/member 
> picks up, I know I can do this with dial in the extensions.conf but wasn't 
> sure about the queue.  Basically I have a macro that identifies the caller 
> and need that run.

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[Asterisk-Users] 407 proxy authentication

[hgaillac-sip]

Hello,

Asterisk sent back 407 proxy authentication .
How can avoid this ?
I set insecure=very without success in sip.conf and my
sql server .

Harry








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Re: [Asterisk-Users] gotoif

[Doug Lytle]

Shaun wrote:



Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): 
syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or 
TOK_COMPL or TOK_LP or TOKEN; Input:




The dial plan works and all, it's just i want those warnings to go away!


  
This has been covered a few time in the last 2 months.  You need to 
initialize the variable:


   Set(holdopt=0)

Before doing any testing with it.

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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[Asterisk-Users] match callerid against outgoing calls

[James Andrewartha]

Hi,

Does anyone have an agi that compares the callerid of an incoming call to
recently dialled numbers in the CDR, and routes the call the phone that last
dialled that number? Basically it'd be so when someone returns a missed call
it goes back to the person who made the call, rather than having to go
through the receptionist.

Thanks,

-- 
James Andrewartha
Systems Administrator
Data Analysis Australia Pty Ltd
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[Asterisk-Users] Re: Suggested MeetMe feature: 'i' without review.

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jon Webster <[EMAIL PROTECTED]> wrote:
> I recently setup app_meetme with the 'i' option. My boss wants users to
> say their name and go directly into the conference instead of reviewing
> the recording.

Funny, I did just the same thing yesterday. All it involved was changing
the line (in app_meetme.c):

res = ast_record_review(chan, "vm-rec-name", user->namerecloc, 10, "sln", 
&duration, NULL);

to be:

res = ast_play_and_record(chan, "vm-rec-name", user->namerecloc, 10, "sln", 
&duration, 128, 1000, NULL);

The 1000 is the number of milliseconds of silence to detect end of name,
so the use doesn't even have to press #. While doing this, I uncovered
a bug in the truncation of the trailing silence (it didn't). See bug
number 6903 on Mantis for the fix.

> If anyone else is interested in this behavior becoming an option, has a
> suggestion what letter to use as the option (I was thinking 'i' -- with
> review and 'I' -- without review), or anything else, I'd appreciate some
> feedback.

I guess it might be useful to have both options. Personally, I think having
the review and confirm step is just a waste of time, and I've never heard
it on a commercial conferencing system, so I would be happy with a permanent
change of behaviour.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]

[Ronan Mullally]

On Fri, 7 Apr 2006, Ronan Mullally wrote:

> Why not just install an ethernet switch on both ends that supports
> trunking / etherchannel?  Less configuration, less chance for operator
> error, and no hard disks.  You'll most likely also need a switch on
> each end *anyway*...

Before I get toasted through and through - you would still need the VDSL
extenders of course, but the Linux boxes seem like overkill...


-Ronan
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Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]

[Ronan Mullally]

Hi Colin,

Nice post - thanks, but:

> HARDWARE:
>
> 2 X 4-PCI slot PC's
> 6 X Black Box LB300A VDSL Ethernet Extenders
> 8 X PCI Ethernet cards

Why not just install an ethernet switch on both ends that supports
trunking / etherchannel?  Less configuration, less chance for operator
error, and no hard disks.  You'll most likely also need a switch on
each end *anyway*...


-Ronan
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[Asterisk-Users] Re: gotoif

[Shaun]

Also forgot to say that the error is triggered by the gotoif (reason the 
subject is labeled that) and not read...

-- 

~Shaun


"Shaun" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> Here is a section of my dialplan (macro)
>
> exten => s,200,Wait(1)
> exten => s,201,read(holdopt|screen-onhold|1)
> exten => s,202,GotoIf($[${holdopt} = 1 ]?4)
> exten => s,203,GoTo(200)
>
>
> it's simple really it loops telling you the caller is on hold until you 
> press 1 and then it sends you off to another area.  The problem right now 
> is that if the read() times out i get these warnings...
>
>
> Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:183 ast_yyerror: 
> ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting 
> TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
> = 1
> ^
> Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:187 ast_yyerror: If you have 
> questions, please refer to doc/README.variables in the asterisk source.
>
>
> The dial plan works and all, it's just i want those warnings to go away!
>
>
> -- 
>
> ~Shaun
>
>
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> 



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[Asterisk-Users] gotoif

[Shaun]

Here is a section of my dialplan (macro)

exten => s,200,Wait(1)
exten => s,201,read(holdopt|screen-onhold|1)
exten => s,202,GotoIf($[${holdopt} = 1 ]?4)
exten => s,203,GoTo(200)


it's simple really it loops telling you the caller is on hold until you 
press 1 and then it sends you off to another area.  The problem right now is 
that if the read() times out i get these warnings...


Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): 
syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or 
TOK_COMPL or TOK_LP or TOKEN; Input:
 = 1
 ^
Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:187 ast_yyerror: If you have 
questions, please refer to doc/README.variables in the asterisk source.


The dial plan works and all, it's just i want those warnings to go away!


-- 

~Shaun 



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Re: [Asterisk-Users] CallerID

[Alejandro Vargas]

2006/4/7, Miles Scruggs <[EMAIL PROTECTED]>:
> Could you give me an example code of how this would work, and how to
> setup the database, I'm pretty new and while what you have written makes
> sense, and sounds like a good plan I'm not sure I can implement it.

I'm using my own agi-bin for "patching" callerid and adding the name
if the number is found in a table (a csv that is mantained with a
spreadsheet), it adds the name taken from this table. Then you can see
the name in the display of the phones.

--
Alejandro Vargas
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RE: [Asterisk-Users] IAX: Auto-congesting call due to slow response

[Mimmus]

 
> maybe firewall tends to close iax connection, you can try to 
> decrease qualify check interval (maybe qualify=5000?) PJ
Peraphs. 'qualify = 1000' seems to alleviate the problem.

Thanks
Domenico

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