Re: [Asterisk-Users] Dial command to connect two channels and bypass asterisk server

[Peter Fern]

You can enable this on a per-peer basis with:

sip peers:
canreinvite=yes

iax peers:
notransfer=no

Check the iax.conf.sample and sip.conf.sample files for usage.

Nitin Gupta wrote:

Hi I was wondering if its possible to make Dial command bridge two 
channels and after bridging bypass asterisk, so that the voice doesn't 
need to pass through my asterisk server.
 
For e.g., I have a user dialed in and he verifies himself and then 
dials an international extension, after the call connects I don't want 
the call to pass through asterisk server anymore. Is there any command 
already there for any particular channel type?
 
Thanks,

Nitin



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Re: [Asterisk-Users] Different Voice Prompts at Different Times

[trixter aka Bret McDanel]

On Mon, 2006-02-13 at 22:20 -0800, Faisal Inam wrote:
> Hello there,
>  
> I want to have different voice prompts(of the Digital Receptionist) at
> different Times. For example, 
> From 10:00AM to 11:00 AM   Voice Message 1
> From 1:00PM to 3:00 PM ..   Voice Message 2
> All other times .Voice Message 3
>  
> I will be very grateful to any help in this regard.
>  

gotoiftime doesnt work for this?  Or were you unaware of it?

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

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] TDM04B/TDM2401E Card

[bbench]

On Monday 13 February 2006 20:43, housi mueller wrote:
> Hi there,
>
>   I plan to use Aterisk in our small office. Until now we used a Panasonic
> D-1232 Super Hybrid System. The figure is representing the future
> configuration I where thinking about to have in the office.
>
>   Question 1:
> We need only 4 lines and I thought to buy a TDM04B or a TDM2401E card.
> There is quite a price difference. Which card would you recommend me to
> buy.
>
>   Question 2:
> Is such a configuration as shown on the figure with a TDM04B/TDM2401E card 
> at all realizable?
I am not familiar with that model in particular, but I've done some 
reasurch about D500. I think all of them have BRI interface so you may 
consider a BRI http://www.junghanns.net/asterisk/page17.html
or http://www.avm.de/en/Produkte/Server-Produkte/C4/index.js.html
or http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4bri.htm
to interconnect Panasonic and your Asterisk.
By my opinion Digium cards are more end user/provider oriented which is 
not your case. Like they say "card that supports FXS and FXO station 
interfaces for connecting analog telephones and analog POTS lines through a 
PC".
Let me know how did you do it.
benchev
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[Asterisk-Users] Different Voice Prompts at Different Times

[Faisal Inam]

Hello there,     I want to have different voice prompts(of the Digital Receptionist) at different Times. For example,   From 10:00AM to 11:00 AM       Voice Message 1  From 1:00PM to 3:00 PM ..   Voice Message 2  All other times .    Voice Message 3     I will be very grateful to any help in this regard.     Waiting for Reply at [EMAIL PROTECTED]
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[Asterisk-Users] Dial command to connect two channels and bypass asterisk server

[Nitin Gupta]

Hi I was wondering if its possible to make Dial command bridge two channels and after bridging bypass asterisk, so that the voice doesn't need to pass through my asterisk server.
 
For e.g., I have a user dialed in and he verifies himself and then dials an international extension, after the call connects I don't want the call to pass through asterisk server anymore. Is there any command already there for any particular channel type?

 
Thanks,
Nitin
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Re: [Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent.

[Scott Hooten]

Thank you john!!
Thanks to you, i found my problem.the variable wrapuptime is not
the amount of time in milliseconds, its seconds!!  The text i had
read previously was wrong.  So now it no longer waits 5000 seconds
to send queue calls to the agent.lol
ScottOn 2/13/06, John Bittner <[EMAIL PROTECTED]> wrote:





Hi Scott,
Do you see asterisk trying to send him calls in the 
CLI?
What does your config file look like? What do you have 
wrapuptime set for? Try setting wrapuptime=0.
 
I had an issue with Cisco phones. When asterisk sent a 
call with a invalid format for callerid it locked up the phone, and the 
phone would not accept anymore calls.
I would see busy back from the phone in the 
CLI.
 
John Bittner
Simlab.net
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Scott 
HootenSent: Monday, February 13, 2006 8:23 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk: 
Agent logs into queue,and there are calls in the queue, but calls don't go to 
agent.
When an agent logs into a queue using AgentCallBackLogin, he should 
be ready to take calls until he logs out right?  For some reason the first 
time a customer calls the queue, it rings the agent just fine but after the 
agent hangs up the phone and the next caller calls the queue, no more calls will 
be transferred to the agent.  He shows as logged in, but the calls wait in 
the queue forever and never get sent to the agent.  Ive googled for this 
but couldnt find anyone else with this problem.Ive tried many different 
ways and nothing seems to work.  Ive been baffled by this for days. Any 
Ideas?I am using Asterisk 1.2.4 on slackware 
10.2Scott

-- Scott Hooten[EMAIL PROTECTED]
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RE: [Asterisk-Users] attended call transfer

[Michael Collins]

JCC,

The issue boils down to this: how much work does the human have to do to
get the calls routed to the right place?  In a traditional PBX
environment, a receptionist does not have to choose beforehand whether
he/she is going to do a blind or attended transfer.  Like I said before,
how many different types of transfer keys do you have on a "normal" PBX
telephone?  Just one.  A transfer is a transfer, regardless of whether
it is blind or attended.  If the phone system forces the human to choose
which type of transfer to do before he executes the operation then it is
limiting his ability to do his work effectively.  

Ira's example about needing the flexibility to do a blind or attended
transfer is not at all far fetched.  Our receptionists answer over 1000
calls per day.  If they had to choose between blind and attended
transfers on every single call then that would increase their workload
significantly.  Thankfully, they don't have to because our PBX allows
them to do blind or attended transfers by allowing them to release the
transferred call simply by hanging up, regardless of whether or not the
destination station has answered yet.  That is the "proper" way of
executing transfers in 10's of millions of offices around the world
using legacy PBX's and key systems.

With Asterisk pushing the envelope forward in so many other areas, does
it really seem like a good idea not to have what many feel is a very
basic feature?  You've already seen 3 or 4 who would be willing to kick
in a few $ to get it into the stable release, and I'd wager that dozens
more would pay a modest sum to add this feature to their Asterisk
installations.  It's certainly less expensive than adding another
receptionist.

-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JCC
Sent: Monday, February 13, 2006 5:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] attended call transfer

Huh? I don't understand.. If the operator can't pick up the call you
need
more operators to compensate.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Sunday, February 12, 2006 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] attended call transfer

At 12:57 AM 02/12/2006, you wrote:
>Why don't you think it is correct behaviour? The purpose of attended 
>transfer is that you consult with the party before transferring with 
>hooking, otherwise it would be a blind transfer for which there is a 
>blind transfer option.

So let's consider an operator, takes a call and decides to attended 
transfer it to Bob because it's slow and she want's to ask something, 
but the instant she picks that option another call comes in. If 
hanging up converted it to blind transfer she could get on with her 
work and answer the next call, as it is she needs to wait till 
something happens and possibly lose the next call.  OK, it's a 
stretch but it does seem like hanging up the call is just wrong!

Ira 


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Re: [Asterisk-Users] Manager cmd: originate without picking up the fone?!

[Ravi Shankar]

Here is what you have to do with the call files. I think similar thing 
can be done with manager API "originate" command just translate/format 
into corresponding manager API option.


The .call file should appear something like this and it has to be placed 
in /var/spool/asterisk/outgoing of asterisk-1,


Channel: local/[EMAIL PROTECTED] ; Any extension can be called using 
local/@

MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: sip
Extension: 2001
Priority: 1

In asterisk we should have the following entries in extensions.conf file,

[sip]
exten => 3001,1,MyOriginateScript()
exten => 3001,2,Hangup

2001 is the called party and as soon as he answers MyOriginateScript() 
will pick up the call. Hope this helps.


regards,
Ravi

Arnd Vehling wrote:


Hi There,

we are developing a dialer application using the java lib
to interface with the asterisk manager protocol. It works
fine so far. The only problem we have is that if we use
the "originate" command the user is required to pick up
the fone _bevore_ asterisk will originate the call to
the desired destination.

What we would like to do is to place the call, check if
the other end is available (ringing event) and only then
let the user pickup the fone. Otherwise we would only display
a text message "Destination 'Busy' etc.

Does anyone know if/how this is possible?

cheers,

  Arnd

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RE: [Asterisk-Users] Best quad-port fxo solution with EC?

[gw]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, February 13, 2006 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Best quad-port fxo solution with EC?


> > I am trying to figure out which way to go for a quad port fxo 
> > solution with a good echo can on it.  My options are the sangoma 
> > remora, a mediatrix fxo, or something similar.
> > 
> > The issue is that I would need a good EC.  This would be on about a 
> > 9000 foot loop, and the lines don't function well on a spa-3000 or 
> > zaptel tdm
> > 4 port card.
> > 
> > Anyone have experience that drives them in a certain direction when 
> > considering a good ec on a quad port?
> > 
> > I tried this also with some fxo clones, but echo killed it.
> 
> The nearest CO my POTS line goes to is 11 miles away.  My POTS line 
> works when plugged into my TDM400P FXO port.  I DID have to fiddle 
> with the gains a bit and I still have to get rid of the last of the 
> echo, but overall it seems to work well.

Are you sure the telco is not using fiber-extended line modules?
- Not certain.

Have you measured the loss from their milliwatt generator?
- Yes, results were way off, unless I did it wrong, my gains should be
about -2 on a fttp line but the measurements suggest about +7

(The numbers would be very interesting to see since a large number of
spa3k and tdm04 users at that distance have significant EC issues.)

- I'll post when I get more details...

Greg
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RE: [Asterisk-Users] Terminating AGI Scripts

[Douglas Garstang]

I don't think that will work. I wanted to catch any number dialled and put 
_everything_ through the script. If there's bits in the script and bits in 
extensions.conf, it becomes a mess. It also increases load if you have to 
execute multiple AGI scripts for a single call, rather than just once.
 
It also isn't as simple as executing a single dial command. We may want to 
execute any Asterisk application for a given number. Queue, Meetme for example. 
 
Argh, why does this have to be so hard.

-Original Message- 
From: Andy Brezinsky [mailto:[EMAIL PROTECTED] 
Sent: Mon 2/13/2006 9:32 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Terminating AGI Scripts



Are you running dial from an AGI script?  We had this exact same 
problem with
one of our setups.  Our solution was to run the Agi script and have it 
set a
DIALCMD variable inside.  When the script terminates it check to see if
DIALCMD is set and then dials that.  If inside your script you know 
you're
going to dial and go away for a long time you do that, otherwise you 
can sit
there the whole time and handle dial yourself.

All except one of our apps now use DIALCMD and has cut our system load 
by 75%

--
~Andy Brezinsky

On Monday 13 February 2006 11:26 pm, Douglas Garstang wrote:
> I've noticed that Asterisk AGI scripts don't terminate when a call is
> answered. Does anyone know how to do this? I would think that this 
would be
> a very big problem, if the scripts stayed in memory, doing nothing, 
until
> the call terminates.
>
> Not only do you have to have a process for routing each call, but all 
the
> previous calls, that are still in progress, also have scripts 
running. It
> wouldn't take very long for even the best system to become overloaded 
with
> processes.
>
> Wouldn't matter if it was AGI or FastAGI ether. Threads or processes,
> either way it's a resource that is essentially unbound, especially if 
you
> service lots of long calls. Once a call is answered anyway, I don't 
see a
> need for the script to continue running. Asterisk has done it's thing.
>
> Doug.

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Re: [Asterisk-Users] Terminating AGI Scripts

[Andy Brezinsky]

Are you running dial from an AGI script?  We had this exact same problem with 
one of our setups.  Our solution was to run the Agi script and have it set a 
DIALCMD variable inside.  When the script terminates it check to see if 
DIALCMD is set and then dials that.  If inside your script you know you're 
going to dial and go away for a long time you do that, otherwise you can sit 
there the whole time and handle dial yourself.

All except one of our apps now use DIALCMD and has cut our system load by 75%

-- 
~Andy Brezinsky

On Monday 13 February 2006 11:26 pm, Douglas Garstang wrote:
> I've noticed that Asterisk AGI scripts don't terminate when a call is
> answered. Does anyone know how to do this? I would think that this would be
> a very big problem, if the scripts stayed in memory, doing nothing, until
> the call terminates.
>
> Not only do you have to have a process for routing each call, but all the
> previous calls, that are still in progress, also have scripts running. It
> wouldn't take very long for even the best system to become overloaded with
> processes.
>
> Wouldn't matter if it was AGI or FastAGI ether. Threads or processes,
> either way it's a resource that is essentially unbound, especially if you
> service lots of long calls. Once a call is answered anyway, I don't see a
> need for the script to continue running. Asterisk has done it's thing.
>
> Doug.

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[Asterisk-Users] Terminating AGI Scripts

[Douglas Garstang]

I've noticed that Asterisk AGI scripts don't terminate when a call is answered. 
Does anyone know how to do this? I would think that this would be a very big 
problem, if the scripts stayed in memory, doing nothing, until the call 
terminates.
 
Not only do you have to have a process for routing each call, but all the 
previous calls, that are still in progress, also have scripts running. It 
wouldn't take very long for even the best system to become overloaded with 
processes.
 
Wouldn't matter if it was AGI or FastAGI ether. Threads or processes, either 
way it's a resource that is essentially unbound, especially if you service lots 
of long calls. Once a call is answered anyway, I don't see a need for the 
script to continue running. Asterisk has done it's thing.
 
Doug.
 
 
 
 
 
 
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Re: [Asterisk-Users] ZAP Digit Timeout

[Jarrod O'Connell]

Does anyone have any info on this?On 1/14/06, Geoff Manning <[EMAIL PROTECTED]> wrote:
We useSetVar(TIMEOUT(digit)=8)In our dialplan to make sure that the user is done dialing before Asteriskexecutes the call. I just recently came across the piece I've copied below.It says for new incoming ZAP connections, the default digit timeout is 3
seconds and can only be configured in the source code.Is that trueHow long will Asterisk wait?(http://www.voip-info.org/wiki-Asterisk+Extension+Matching
)How long Asterisk will wait for more digits, before giving up and processingthe number as dialed so far, depends on two factors. First of all, itdepends on whether this is a new incoming connection or an established
connection.* New incoming connections are those where Asterisk has not startedprocessing any Dialplan commands for the call yet. For Zap channels, thismeans that someone has lifted the handset and started dialing a number, and
Asterisk is processing each digit as the user dials it, to determine whichextension in the Dialplan to go first.* For new incoming connections, the second factor affecting how longAsterisk waits for more digits before giving up is the type of channel that
the call is coming in on. Each channel type determines its own timeoutperiod. The Zap channel has a fixed timeout period of 3 seconds, and thiscan not be changed without modifying the source code and recompiling the Zap
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[Asterisk-Users] Problem Faxing

[Carlos Chavez]

 I have an Asterisk 1.2.4 64 bt server with 10 lines on an E1 (running on
Unicall), and 2 lines on a TDM02B card.  I am trying to use a Linksys PAP2 to
connect a couple of fax machines.  I can send faxes between the two fax
machines so I know that everything is configured properly.  The problem is
when I ty to send or receive an external fax.  I can dial and the remote
machine answers but after that only once has the fax gone through.  I have not
been able to receive any external faxes at all on both fax machines.

 I really do not know what the problem may be as I have used several ATA
devices in the past to connect analog faxes and they usually do not have any
problems.  Any sugestions?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP

[Rick Smith]

Phil;

What link ?



We're got a T1 from Sprint that we use for internet.  During VIOP calls,
if you download something, the VOIP calls break up.

I found some info at Sprint for adding 'class of service', and I also
have some information on configuring our Cisco routers.

I've read the relevent pages on the wiki, but it seems vauge what's
required and what's required by the NSP (Sprint).

How have you dealt with this problem?  Is this something which requires
the NSP to be involved, or can this all be done on the premises side in
IOS configuration(s)?


Phil


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Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP

[David Choo]

Hi 

Best Regards

==
David Choo
Sales Engineer
Citrix Certified Administrator
Polycom Qualified Tech & Sales Rep
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-6842 2725, Ext - 404
DID: 65-9006 2645
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=

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Philip Edelbrock <[EMAIL PROTECTED]>

Sent by: [EMAIL PROTECTED]
14/02/2006 07:23 AM



Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion      
 





To
asterisk-users@lists.digium.com


cc



Subject
[Asterisk-Users] Traffic prioritization
and 'class of service' for        SIP









We're got a T1 from Sprint that we use for internet.  During VIOP
calls, 
if you download something, the VOIP calls break up.

I found some info at Sprint for adding 'class of service', and I also 
have some information on configuring our Cisco routers.

I've read the relevent pages on the wiki, but it seems vauge what's 
required and what's required by the NSP (Sprint).

How have you dealt with this problem?  Is this something which requires

the NSP to be involved, or can this all be done on the premises side in

IOS configuration(s)?


Phil
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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 90

[Michaël Gaudette]

>That is what I have.  Unfortunately, the context=someprovider-in is being
>ignored.  I am running asterisk-1.2.4...
>
>The local-phone context is working properly though.  I can't see why one is
>behaving as I expect and the other isn't.
>  
>
>Its because your incoming call from the provider is not being matched to 
>[someprovider].  Do a sip debug with an incoming call and you will see 
>the call being matched to the [general] section that probably points to 
>the default context.

Thanks Joseph and Andres.  This is actually a 100% right. I assumed (my bad)
that a call that didn't match any provider would simply be dropped.  The
provider had given me the wrong info, and when I put it in it did go to the
default context.

One day I'll learn to read those debug logs properly.  Until then I have to
say a bit thank you for the help I got here.

Regards,



Mike

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[Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent

[Scott Hooten]

Here are some of the config files

here is queues.conf:

[CSR-queue]
music=default
strategy=ringall
timeout=15
retry=5
wrapuptime=5000
maxlen = 0
periodic-announce = Did-you-know-support
periodic-announce-frequency = 60
announce-frequency = 60
announce-holdtime = yes
member => Agent/204 ;Dangelo
member => Agent/207 ;Eugene
member => Agent/208 ;Alzina
member => Agent/209 ;Jerry
member => Agent/210 ;GeorgeU
member => Agent/211 ;Mauricio
member => Agent/212 ;Josh
member => Agent/213 ;Test
member => Agent/214 ;Tomas
member => Agent/218 ;Andy
member => Agent/219 ;AlexV
member => Agent/220 ;Max
member => Agent/221 ;MarcoB
member => Agent/224 ;Scott Test

here is extensions.conf:


[globals]
scott=Zap/1
outboundlocal=Zap/g1
outboundUS=Zap/g2

[pstn]
; incoming calls from the pstn FXO ports are directed
; to this context from zapata.conf
exten => s,1,Goto(mainmenu,s,1)

[dow]
; incoming calls from the dow FXO ports are directed
; to this context from zapata.conf
exten => s,1,Goto(mainmenu,s,1)


[mainmenu]
include => invalid
exten => s,1,Background(thank-you-for-calling-pok
er-support)        ;Welcome to DBPNexten => s,n,Background(dbpn-main-menu)        ;Announce menu of options for callersexten => s,n,WaitExten
exten => 1,1,Goto(submenu-ext,s,1)        ;Press one to dial an extensionexten => 2,1,Goto(submenu-support-main,s,1)        ;Press two for supportexten => 3,1,Playback(SEC-queue-msg)exten => 3,n,Queue(SEC-queue)
        ;Press three for securityexten => 4,1,Playback(ACC-queue-msg)exten => 4,n,Queue(ACC-queue)        ;Press four for accountingexten => 8,1,Directory(default,pstn,f)        ;Press eight for directory
exten => 0,1,Goto(submenu-support-main,s,1)        ;If zero is pressed, goto supportexten => t,1,Goto(submenu-support-main,s,1)        ;If timeout then goto support-main[submenu-ext]include => extensions
include => queuesexten => s,1,Wait,1exten => s,n,Background(please-enter-extension)        ;Please enter your partys extexten => t,1,Goto(mainmenu,s,1)[submenu-support-main]include => invalid
exten => s,1,Ringing        ;Make them comfortable with 2 seconds of ringbackexten => s,n,Wait,2exten => s,n,Background(support-main-menu)        ;Announce support main menu of options for callers
exten => s,n,WaitExtenexten => 1,1,Playback(ACC-queue-msg)exten => 1,n,Queue(ACC-queue)        ;Press one for status of a checkexten => 2,1,Goto(submenu-support-depositing,s,1)        ;Press two for help with depositing money
exten => 3,1,Goto(submenu-support-tech,s,1)        ;Press three for technical issuesexten => 0,1,Goto(submenu-support-main,s,1)        ;If zero is pressed, goto supportexten => t,1,Playback(CSR-queue-msg)
exten => t,n,queue(CSR-queue)        ;If timeout then goto CSR-queue[submenu-support-depositing]include => invalidexten => s,1,Ringing        ;Make them comfortable with 2 seconds of ringback
exten => s,n,Wait,2exten => s,n,Background(support-depositing-menu)        ;Announce support depositing menu of options for callersexten => s,n,WaitExtenexten => 1,1,Playback(SEC-queue-msg)
exten => 1,n,Queue(SEC-queue)        ;Press one for help with depositing money with credit cardexten => 2,1,Playback(ACC-queue-msg)exten => 2,n,Queue(ACC-queue)        ;Press two for help with depositing money with epassporte
exten => 3,1,Playback(CSR-queue-msg)exten => 3,n,queue(CSR-queue)        ;Press three for help with depositing money with otherexten => 0,1,Goto(submenu-support-depositing,s,1)        ;If zero is pressed, goto support depositing
exten => t,1,Goto(submenu-support-depositing,s,1)        ;If timeout then repeat menu[submenu-support-tech]include => invalidexten => s,1,Ringing        ;Make them comfortable with 2 seconds of ringback
exten => s,n,Wait,2exten => s,n,Background(support-tech-menu)        ;Announce support depositing menu of options for callersexten => s,n,WaitExtenexten => 1,1,Playback(CSR-queue-msg)exten => 1,n,queue(CSR-queue)
        ;Press one for difficulties dl poker softwareexten => 2,1,Playback(TECH-queue-msg)exten => 2,n,queue(TECH-queue)        ;Press two for difficulties connecting to poker serverexten => 3,1,Playback(TECH-queue-msg)
exten => 3,n,queue(TECH-queue)        ;Press three for all other questionsexten => 0,1,Playback(TECH-queue-msg)exten => 0,n,queue(TECH-queue)        ;If zero is pressed, goto tech queueexten => t,1,Playback(TECH-queue-msg)
exten => t,n,queue(TECH-queue)        ;If timeout then goto TECH-queue[invalid]exten => i,1,Playback(pbx-invalid)exten => i,n,Goto(s,1)[prompts]exten => _[0]X,1,Answer()
exten => _[0]X,2,Record(prompt${EXTEN:1}:gsm)exten => _[0]X,3,Playback(prompt${EXTEN:1})exten => _[0]X,4,Hangup()[queues]exten => 200,1,Playback(CSR-queue-msg)exten => 200,n,queue(CSR-queue)
exten => 300,1,Playback(TECH-queue-msg)exten => 300,n,queue(TECH-queue)exten => 400,1,Playback(ACC-queue-msg)exten => 400,n,queue(ACC-queue)exten => 600,1,Playback(SEC-queue-msg)exten => 600,n,queue(SEC-queue)
[confe

[Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent

[Scott Hooten]

Here is some dialog from the Console:



    -- Starting simple switch on 'Zap/13-1'

Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)...

    -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack

    -- Goto (mainmenu,s,1)

    -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker
-support") in new stack
    -- Playing 'thank-you-for-calling-poker-support' (language 'en')
Feb 10 07:22:36 WARNING[21105]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 13
    -- Executing BackGround("Zap/13-1", "dbpn-main-menu") in new stack
    -- Playing 'dbpn-main-menu' (language 'en')
  == Spawn extension (mainmenu, s, 2) exited non-zero on 'Zap/13-1'
    -- Hungup 'Zap/13-1'
    -- Starting simple switch on 'Zap/1-1'
    -- Executing AgentCallbackLogin("Zap/1-1", "||@extensions") in new stack
    -- Playing 'agent-user' (language 'en')
    -- Playing 'agent-pass' (language 'en')
    -- Playing 'agent-newlocation' (language 'en')
    -- Playing 'agent-loginok' (language 'en')
  == Callback Agent '224' logged in on [EMAIL PROTECTED]
    -- Playing 'vm-goodbye' (language 'en')
  == Spawn extension (internal, #1, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
asterisk*CLI> show agents
204          (Dangelo) not logged in (musiconhold is 'default')
207          (Eugene) not logged in (musiconhold is 'default')
208          (Alzina) not logged in (musiconhold is 'default')
209          (Jerry) not logged in (musiconhold is 'default')
210          (GeorgeU) not logged in (musiconhold is 'default')
211          (Mauricio) not logged in (musiconhold is 'default')
212          (Josh) not logged in (musiconhold is 'default')
213          (Test) not logged in (musiconhold is 'default')
214          (Tomas) not logged in (musiconhold is 'default')
218          (Andy) not logged in (musiconhold is 'default')
219          (AlexV) not logged in (musiconhold is 'default')
220          (Max) not logged in (musiconhold is 'default')
221          (MarcoB) not logged in (musiconhold is 'default')
224          (Scott Test) available at '[EMAIL PROTECTED]' (musiconhold is 'default')
601          (Grant) not logged in (musiconhold is 'default')
604          (Eliecer) not logged in (musiconhold is 'default')
605          (Marcus) not logged in (musiconhold is 'default')
607          (Alex C) not logged in (musiconhold is 'default')
608          (Monty) not logged in (musiconhold is 'default')
603          (Michael C) not logged in (musiconhold is 'default')
302          (Scott) not logged in (musiconhold is 'default')
303          (Alex) not logged in (musiconhold is 'default')
305          (Todd) not logged in (musiconhold is 'default')
306          (Post) not logged in (musiconhold is 'default')
24 agents configured [1 online , 23 offline]
asterisk*CLI>
asterisk*CLI> show queue CSR-queue
CSR-queue    has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members: >
      Agent/204 (Unavailable) has taken no calls yet
      Agent/207 (Unavailable) has taken no calls yet
      Agent/208 (Unavailable) has taken no calls yet
      Agent/209 (Unavailable) has taken no calls yet
      Agent/210 (Unavailable) has taken no calls yet
      Agent/211 (Unavailable) has taken no calls yet
      Agent/212 (Unavailable) has taken no calls yet
      Agent/213 (Unavailable) has taken no calls yet
      Agent/214 (Unavailable) has taken no calls yet
      Agent/218 (Unavailable) has taken no calls yet
      Agent/219 (Unavailable) has taken no calls yet
      Agent/220 (Unavailable) has taken no calls yet
      Agent/221 (Unavailable) has taken no calls yet
      Agent/224 (Not in use) has taken no calls yet
   No Callers
asterisk*CLI>
    -- Starting simple switch on 'Zap/13-1'
Feb 10 07:24:25 NOTICE[21115]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)...
    -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack
    -- Goto (mainmenu,s,1)
    -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker-support") in new stack
    -- Playing 'thank-you-for-calling-poker-support' (language 'en')
Feb 10 07:24:25 WARNING[21115]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 13
    -- Executing BackGround("Zap/13-1", "dbpn-main-menu") in new stack
    -- Playing 'dbpn-main-menu' (language 'en')
  == CDR updated on Zap/13-1
    -- Executing Goto("Zap/13-1", "submenu-ext|s|1") in new stack
    -- Goto (submenu-ext,s,1)
    -- Executing Wait("Zap/13-1", "1") in new stack
    -- Executing BackGround("Zap/13-1", "please-enter-extension") in new stack
    -- Playing 'please-enter-extension' (language 'en')
  == CDR updated on Zap/13-1
    -- Executing Playback("Zap/13-1", "CSR-queue-msg") in new stack
    -- Playing 'CSR-queue-msg' (language 'en')
    -- Executing Queue("Zap/13-1", "CSR-queue") in new stack
    -- Started music on hold, class 'default', on Zap/13-1
    -- outgoing agentca

Re: [Asterisk-Users] Asterisk Televantage integration

[asterisk]

On Mon, 13 Feb 2006, Anish Basu wrote:

I am trying to trunk calls from an old televantage system into asterisk.
The version of televantage being used does not support SIP, so H.323 trunks
must be used.  Has anyone had any experience with this?  Would I have to use
opengk and get both asterisk and televantage to register as endpoints, or
can direct peer to peer connections be possible?  Also, which h.323 asterisk
implementation is best for this situation?


you dont need a gk. asterisk can talk to televantage with h323 without a 
gk no problem. i have had most success with oh323.


-Dan
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Re: [Asterisk-Users] bug in bristuff?

[Conrad Wood]

On Mon, 2006-02-06 at 22:58 +, Conrad Wood wrote:

> Unqiueid: asterisk-1713-1139266402.909
> ^
> 
> Please note the spelling of uniqueid. I find the spelling in
> res_features.c - but only once I patched it with bristuff patches.
> Does anyone know whether that is a known problem with bristuff? If so is
> it fixed in a later version?
> Where do I report a bug in bristuff? ;)

For what  it's worth: this seems to be fixed in newer version of
bristuff... thanks everybody... ;-)

Conrad


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[Asterisk-Users] Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent.

[Scott Hooten]

When an agent logs into a queue using AgentCallBackLogin, he should be
ready to take calls until he logs out right?  For some reason the first
time a customer calls the queue, it rings the agent just fine but after
the agent hangs up the phone and the next caller calls the queue, no
more calls will be transferred to the agent.  He shows as logged in,
but the calls wait in the queue forever and never get sent to the
agent.  Ive googled for this but couldnt find anyone else with this
problem.

Ive
tried many different ways and nothing seems to work.  Ive been baffled
by this for days. Any Ideas?

I am using Asterisk 1.2.4 on slackware 10.2

Scott
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RE: [Asterisk-Users] attended call transfer

[JCC]

Huh? I don't understand.. If the operator can't pick up the call you need
more operators to compensate.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Sunday, February 12, 2006 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] attended call transfer

At 12:57 AM 02/12/2006, you wrote:
>Why don't you think it is correct behaviour? The purpose of attended 
>transfer is that you consult with the party before transferring with 
>hooking, otherwise it would be a blind transfer for which there is a 
>blind transfer option.

So let's consider an operator, takes a call and decides to attended 
transfer it to Bob because it's slow and she want's to ask something, 
but the instant she picks that option another call comes in. If 
hanging up converted it to blind transfer she could get on with her 
work and answer the next call, as it is she needs to wait till 
something happens and possibly lose the next call.  OK, it's a 
stretch but it does seem like hanging up the call is just wrong!

Ira 


-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database: 267.15.6/257 - Release Date: 02/10/2006


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Re: [Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration

[Paul Liew]

Hi Dean,

We at ATP have a range of resellers/integrators on our system to provide 
solutions around Australia. Get them to contact us, and we'll put them 
in touch with the nearest integrator with the correct skillset.


Cheers,
Paul

www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL PROTECTED]
IAX) [EMAIL PROTECTED]
IAXtel) 1700-482-8273
ATP Centrex) 



Dean Collins wrote:


Hi all,

I was just on the phone with a B2C company in Australia who are 
looking to integrate an Asterisk solution with their Salesforce.com 
CRM platform.


They are looking for a consultant/team to provide the following 
functionality


* Complex IVR
  Eg can interface via API into Salesforce for customer service
  interaction and product initiation.

* Call centre
  Eg basic queues, fallovers etc, reports,

* Salesforce.com CTI integration, screen pop, outbound calling etc.

If anyone on this list has extensive API experience with ivr and 
preferably with some salesforce.com screen pop experience then please 
email me and I’ll pass along their details.


Regards,

Dean Collins

Cognation Pty Ltd

[EMAIL PROTECTED] 

+1-212-203-4357

+61-2-9016-5642 (Sydney in-dial).



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RE: [Asterisk-Users] problem with outgoing calls Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

[Michael Collins]

Nik,

I'm not sure that "NOP" is correct, but I'm in the states so I'll to
defer to someone who knows E1/PRI.  When I run zttool I have "OK" under
the alarms.  Is there a way you can call the telco and confirm the
settings?  Make sure that framing, coding and D channels are set up on
their end the same way you're set up.

As for asterisk, here's what I get when I stop asterisk, restart it from
the command line and then log back on to the command line:

[EMAIL PROTECTED] asterisk]# safe_asterisk
[EMAIL PROTECTED] asterisk]# asterisk -r
Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>

=
Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 4524)
NIX connection
Verbosity is at least 3
-- B-channel 0/1 successfully restarted on span 1
-- B-channel 0/2 successfully restarted on span 1
-- B-channel 0/3 successfully restarted on span 1
-- B-channel 0/4 successfully restarted on span 1
-- B-channel 0/5 successfully restarted on span 1
-- B-channel 0/6 successfully restarted on span 1
-- B-channel 0/7 successfully restarted on span 1
-- B-channel 0/8 successfully restarted on span 1
-- B-channel 0/9 successfully restarted on span 1
-- B-channel 0/10 successfully restarted on span 1
-- B-channel 0/11 successfully restarted on span 1
-- B-channel 0/12 successfully restarted on span 1
-- B-channel 0/13 successfully restarted on span 1
-- B-channel 0/14 successfully restarted on span 1
-- B-channel 0/15 successfully restarted on span 1
-- B-channel 0/16 successfully restarted on span 1
-- B-channel 0/17 successfully restarted on span 1
-- B-channel 0/18 successfully restarted on span 1
-- B-channel 0/19 successfully restarted on span 1
-- B-channel 0/20 successfully restarted on span 1
-- B-channel 0/21 successfully restarted on span 1
-- B-channel 0/22 successfully restarted on span 1
-- B-channel 0/23 successfully restarted on span 1


Here are the relevant portions of my zaptel.conf and Zapata.conf files:
Zapata.conf:
[channels]
language=en
context=from-pstn
signalling=pri_cpe
switchtype=5ess
rxwink=300  
callerid=asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
channel => 1-23


Zaptel.conf:

span=1,1,0,esf,b8zs # My PRI line
bchan=1-23
dchan=24
# Global data
loadzone= us
defaultzone = us


HtH!
-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Monday, February 13, 2006 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problem with outgoing calls
Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel
congestion)

On 2/13/06, nik600 <[EMAIL PROTECTED]> wrote:
> On 2/13/06, Michael Collins <[EMAIL PROTECTED]> wrote:
> > When Asterisk first starts up, it will attempt to "bring up" the B
> > channels on any PRI circuits.  If you are using [EMAIL PROTECTED] then you 
> > can log
on
> > to the Asterisk CLI (asterisk -r) and then do "stop now" to stop
> > asterisk.  Start up Asterisk again by typing asterisk -cvvv at the
Linux
> > command line.  You should see a bunch of messages on the terminal
and
> > then you'll get the Asterisk CLI prompt.  A few moments later you
should
> > see some more messages saying something like "B channel 1/1
successfully
> > started on span 0" and then "B channel 1/2 successfully started..."
I
> > don't have my system active at the moment but when I do I'll email
you a
> > screen capture of the messages I see on my PRI.
> >
>
> i get:
> *CLI>   == Primary D-Channel on span 1 up
>
> at the moment i've got only one span connected...il the message
correct?
>
Description  Alarms IRQbpviol 
   CRC4
T2XXP (PCI) Card 0 Span 1NOP0  0  
   0
T2XXP (PCI) Card 0 Span 2RED/NOP0  0  
   0
ZTDUMMY/1 1  UNCONFIGUR 0  0
0
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[Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration

[Dean Collins]

Hi all,

 

I was just on the phone with a B2C company in Australia who
are looking to integrate an Asterisk solution with their Salesforce.com CRM
platform.

 

They are looking for a consultant/team to provide the
following functionality

 


 Complex IVR
 Eg can interface via API into Salesforce for customer service interaction
 and product initiation.


 


 Call centre
 Eg basic queues, fallovers etc, reports,


 


 Salesforce.com CTI
 integration, screen pop, outbound calling etc.


 

 

If anyone on this list has extensive API experience
with ivr and preferably with some salesforce.com screen pop experience then
please email me and I’ll pass along their details.

 

 

 

 

Regards,

 

 

Dean Collins

Cognation Pty Ltd

[EMAIL PROTECTED]

+1-212-203-4357

+61-2-9016-5642 (Sydney in-dial).

 






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RE: [Asterisk-Users] Send HookFlash after answering a ZAP(analog) channel

[Michael Collins]

Curious: Why did you need the wait times to be so long - was it because of your 
PBX or is that simply what you wanted?

-MC

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Juergen 
Brand
Sent: Monday, February 13, 2006 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Send HookFlash after answering a ZAP(analog) 
channel

Found solution:

exten => s,1,Wait(10)
exten => s,2,Answer()
exten => s,3,Wait(5)
exten => s,4,Flash()
exten => s,5,Wait(2)
exten => s,6,SendDTMF(200)
exten => s,7,Wait(5)
exten => s,8,Hangup() 



> --- Ursprüngliche Nachricht ---
> Von: "Hans-Juergen Brand" <[EMAIL PROTECTED]>
> An: asterisk-users@lists.digium.com
> Betreff: [Asterisk-Users] Send HookFlash after answering a ZAP
> (analog)  channel
> Datum: Mon, 13 Feb 2006 19:16:23 +0100 (MET)
> 
> My hardware configuration looks like this:
> 
> public network --->>other PBX ---analog line (ext 100) ---> FXO,Asterisk
>   |
>   |--- extension
>  (200)
> 
> by calling extension 100 from public network I can call Asterisk. Now I
> would like to answer the call, SayNumber and the transer back to the
> extension 200 in the other pbx. When I put a telephone instead of Asterisk
> on the line 100 I can do the transfer by pressing Hookflash and the dial
> 200
> and then hookon. The call is no at 200.
> 
>  extension.conf ---
> 
> exten => s,1,Answer
> exten => s,2,Wait(1)
> exten => s,3,SayNumber(101)
> exten => s,4,Wait(2)
> exten => s,5,Transfer(**51)
> exten => s,6,Wait(40)   
> 
> --
> 
> I trie something, but it is not working.
> 
> kinds regards.
> 
> HJB
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[Asterisk-Users] AGI Scripts Staying in Memory

[Douglas Garstang]

So, I'm noticing that when Asterisk executes an AGI script, that the AGI script 
keeps running until the call is complete. 

Is there any way to  have the script terminate when the call is answered?

Also noticed that when user makes a call to user B, if user B hangs up the 
call, then Asterisk returns a result code. If user A hangs up the call, a 
result code is not returned. Why?

Doug


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[Asterisk-Users] Asterisk Televantage integration

[Anish Basu]

I am trying to trunk calls from an old televantage system
into asterisk.  The version of televantage being used does not support SIP, so
H.323 trunks must be used.  Has anyone had any experience with this?  Would I
have to use opengk and get both asterisk and televantage to register as
endpoints, or can direct peer to peer connections be possible?  Also, which
h.323 asterisk implementation is best for this situation?

 

Anish Basu






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[Asterisk-Users] HELP, SPA-2002 - SPA-2002 singleside sound

[Bernard van de Koppel]

Hi,

I have the folowing setup

SPA-2002 (exten 22) with a Motorola ME4052 dect
 |
Asterisk 1.2
 |
SPA-2002 (exten 58) with a Motorola ME4052 dect

All connected to a simple switched lan.

When making calls, sometimes the incoming (58) side does not seem to be 
capable to send voice data back to the outbound (22) side.

Setting reinvite to no, does not seem to help much (on the asterisk side 
traffic seems to flow in both directions according to show channel xx).

Detail can be found on http://nat1.sipman.net/help
Questions:
1. Is this a known bug (I am using version 3.1.5)
2. Is there a workaround to these kind of problems?

Hope anyone can help.

Bernard
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[Asterisk-Users] Traffic prioritization and 'class of service' for SIP

[Philip Edelbrock]

We're got a T1 from Sprint that we use for internet.  During VIOP calls, 
if you download something, the VOIP calls break up.


I found some info at Sprint for adding 'class of service', and I also 
have some information on configuring our Cisco routers.


I've read the relevent pages on the wiki, but it seems vauge what's 
required and what's required by the NSP (Sprint).


How have you dealt with this problem?  Is this something which requires 
the NSP to be involved, or can this all be done on the premises side in 
IOS configuration(s)?



Phil
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RE: [Asterisk-Users] sip expire 60

[Mike Pollitt]

Hi Jerry --

Have you tried adjusting the settings in the SIP device itself? That's where
you can adjust how frequently the device will try to register.

Regards,
Mike.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, 14 February 2006 9:50 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sip expire 60 

I am getting messages on the console about
Registered SIP ... expires 60

How do I increase that 60 to 3 minutes???

I have tried in [general] of sip.conf

to set
expirey=300
defaultexpirey=300

nothing  seems to affect it.

Thanks,

Jerry
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[Asterisk-Users] sip expire 60

[Jerry Geis]

I am getting messages on the console about
Registered SIP ... expires 60

How do I increase that 60 to 3 minutes???

I have tried in [general] of sip.conf

to set
expirey=300
defaultexpirey=300

nothing  seems to affect it.

Thanks,

Jerry
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Re: [Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel

[Hans-Juergen Brand]

Found solution:

exten => s,1,Wait(10)
exten => s,2,Answer()
exten => s,3,Wait(5)
exten => s,4,Flash()
exten => s,5,Wait(2)
exten => s,6,SendDTMF(200)
exten => s,7,Wait(5)
exten => s,8,Hangup() 



> --- Ursprüngliche Nachricht ---
> Von: "Hans-Juergen Brand" <[EMAIL PROTECTED]>
> An: asterisk-users@lists.digium.com
> Betreff: [Asterisk-Users] Send HookFlash after answering a ZAP
> (analog)  channel
> Datum: Mon, 13 Feb 2006 19:16:23 +0100 (MET)
> 
> My hardware configuration looks like this:
> 
> public network --->>other PBX ---analog line (ext 100) ---> FXO,Asterisk
>   |
>   |--- extension
>  (200)
> 
> by calling extension 100 from public network I can call Asterisk. Now I
> would like to answer the call, SayNumber and the transer back to the
> extension 200 in the other pbx. When I put a telephone instead of Asterisk
> on the line 100 I can do the transfer by pressing Hookflash and the dial
> 200
> and then hookon. The call is no at 200.
> 
>  extension.conf ---
> 
> exten => s,1,Answer
> exten => s,2,Wait(1)
> exten => s,3,SayNumber(101)
> exten => s,4,Wait(2)
> exten => s,5,Transfer(**51)
> exten => s,6,Wait(40)   
> 
> --
> 
> I trie something, but it is not working.
> 
> kinds regards.
> 
> HJB
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[Asterisk-Users] iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)

[Chris Bagnall]

Hello all,

I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of them:
  codec_ilbc.c: Huh?  An ilbc frame that isn't a multiple of 50 bytes long
from RTP (38)?

The ATAs in question are various Grandstream models - the HT486 being the
predominant one. Looking at the list archives, it's been an issue for others
in the past, but there doesn't appear to have been a satisfactory solution I
could find. I understand it's an issue with payload types, so as recommended
in the posts I've read, DTMF is set to payload type 101 and iLBC to 97. I've
also tried 99 as per another post I read.

Is anyone reliably running Grandstream ATAs with iLBC? If so, what settings
did you use for the payload types? Any other suggestions for solving the
problem would be gratefully appreciated.

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[Asterisk-Users] Sagoma w/EC x TE411

[Bruno de Assumpção Loureiro]

Sorry I sent it to the wrong list!!!

Regards.

-- Forwarded message --
From: Bruno de Assumpção Loureiro <[EMAIL PROTECTED]>
Date: Feb 13, 2006 8:41 PM
Subject: Sagoma w/EC x TE411
To: Asterisk Users Mailing List - Nont -Commercial Discussion



Pessoal,
alguem gostou das TE411P, parece que elas nao fazem o deveriam fazer :-) ?!?!
alguem que usa sangoma com EC poderia nos dar um feedback?

Obrigado!
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED]


--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED]
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RE: [Asterisk-Users] Manager cmd: originate without picking up thefone?!

[Wai Wu]

Reverse your call order.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Arnd
Vehling
Sent: Monday, February 13, 2006 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Manager cmd: originate without picking up
thefone?!


Hi There,

we are developing a dialer application using the java lib
to interface with the asterisk manager protocol. It works
fine so far. The only problem we have is that if we use
the "originate" command the user is required to pick up
the fone _bevore_ asterisk will originate the call to
the desired destination.

What we would like to do is to place the call, check if
the other end is available (ringing event) and only then
let the user pickup the fone. Otherwise we would only display
a text message "Destination 'Busy' etc.

Does anyone know if/how this is possible?

cheers,

   Arnd

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[Asterisk-Users] Sagoma w/EC x TE411

[Bruno de Assumpção Loureiro]

Pessoal,
alguem gostou das TE411P, parece que elas nao fazem o deveriam fazer :-) ?!?!
alguem que usa sangoma com EC poderia nos dar um feedback?

Obrigado!
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED]
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Re: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 89

[Andres]

[someprovider-in]
exten => s,1,Dial(SIP/sipphone)
   



That is what I have.  Unfortunately, the context=someprovider-in is being
ignored.  I am running asterisk-1.2.4...

The local-phone context is working properly though.  I can't see why one is
behaving as I expect and the other isn't.
 

Its because your incoming call from the provider is not being matched to 
[someprovider].  Do a sip debug with an incoming call and you will see 
the call being matched to the [general] section that probably points to 
the default context.



--
Andres



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Re: RE : [Asterisk-Users] lists problem, Gmail????????

[Imran Ahmed]

I have experienced similar problems using gmail.
Gmail certainly had some problems with emails from asterisk lists.
I donot know if it was only restricted to asterisk lists.
As not all emails were being delayed (or dropped), some of you might be under
the impression that theres no problem.
Please compare your emails with the list archives to be sure you didnt miss
something important.
Also, the problems seems to have gone away this week.

Regards
Imran

On 2/13/06, Joseph Tanner <[EMAIL PROTECTED]> wrote:
> May be some truth to it though :(
>
> Personally I use gmail, but use a different email address that is
> forwarded to my gmail account.  With this setup, I haven't had any
> issues.  I use gmail because it's easily accessible from any PC, and I
> like how it groups conversations (probably why you see a lot of gmail
> addresses signed up on mailing lists).
>
> Joseph Tanner
>
> On 2/13/06, Olivier.taylor <[EMAIL PROTECTED]> wrote:
> > Pfff,
> >
> > What for an answer :(
> >
> > I use gmail and have no problems.
> >
> > Olivier
> >
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] De la part de Martin
> > Joseph
> > Envoyé : lundi 13 février 2006 20:36
> > À : Asterisk Users Mailing List - Non-Commercial Discussion
> > Objet : Re: [Asterisk-Users] lists problem, Gmail
> >
> >
> >
> > On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:
> >
> > > C F ha scritto:
> > >
> > >> Am I the only one having trouble with this list?
> > >> Since the begining of the week I have not been receiving mail from
> > >> the list like I used to, is this a gmail problem? or is it
> > >> subscription problem? or is something wrong with the list? anybody
> > >> else using gmail having any problems?
> > >>
> > > Yes, I'm also getting some "lag" sometimes, one or two days without
> > > receiving mails
> >
> > get a real mail server and it works great!
> >
> >
> > ___
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> >
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[Asterisk-Users] Detecting Agents and Chanspy

[Johann]

Is it possible to detect if a specific agent is on a call?  I can check 
${AGENTBYCALLERID_${AGENTID}} to see if they are logged in(ie if set they are), 
but I want to be able to detect if they are actually on a call.  The 
ChanIsAvail() doesn't seem to work for Agent channels.  I want to do this from 
within the dialplan.


Also it is possible to alter the format that ChanSpy() records in?  It seems to 
be hard coded to .raw (and lame/sox don't seem to like it for conversion).



--johann
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RE: [Asterisk-Users] automatically start application from the commandprompt

[Michael Collins]

This can also be done with the use of “call”
files.  Check this out:

http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

 

-MC

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan Kroon
Sent: Monday, February 13, 2006
7:10 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
automatically start application from the commandprompt



 

Hello,

 

Is it possible to start an asterisk
application from the command prompt? 

This application has to dial to a number.

When the calling party picks up the phone,
the asterisk application had to play certain voicefiles.

 

Kind Regards,

 

Arjan Kroon

Mobillion B.V. 
Copernicuslaan 30 
Postbus 554 / PO Box
 554 
6710 BN Ede 
tel: +31 (0)318-648920 
fax: +31 (0)318-648839 
mobile: +31 (0)6-55871460 
email: [EMAIL PROTECTED] 
internet: www.mobillion.nl 

 






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[Asterisk-Users] Manager cmd: originate without picking up the fone?!

[Arnd Vehling]

Hi There,

we are developing a dialer application using the java lib
to interface with the asterisk manager protocol. It works
fine so far. The only problem we have is that if we use
the "originate" command the user is required to pick up
the fone _bevore_ asterisk will originate the call to
the desired destination.

What we would like to do is to place the call, check if
the other end is available (ringing event) and only then
let the user pickup the fone. Otherwise we would only display
a text message "Destination 'Busy' etc.

Does anyone know if/how this is possible?

cheers,

  Arnd

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Re: RE : [Asterisk-Users] lists problem, Gmail????????

[Joseph Tanner]

May be some truth to it though :(

Personally I use gmail, but use a different email address that is
forwarded to my gmail account.  With this setup, I haven't had any
issues.  I use gmail because it's easily accessible from any PC, and I
like how it groups conversations (probably why you see a lot of gmail
addresses signed up on mailing lists).

Joseph Tanner

On 2/13/06, Olivier.taylor <[EMAIL PROTECTED]> wrote:
> Pfff,
>
> What for an answer :(
>
> I use gmail and have no problems.
>
> Olivier
>
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Martin
> Joseph
> Envoyé : lundi 13 février 2006 20:36
> À : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [Asterisk-Users] lists problem, Gmail
>
>
>
> On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:
>
> > C F ha scritto:
> >
> >> Am I the only one having trouble with this list?
> >> Since the begining of the week I have not been receiving mail from
> >> the list like I used to, is this a gmail problem? or is it
> >> subscription problem? or is something wrong with the list? anybody
> >> else using gmail having any problems?
> >>
> > Yes, I'm also getting some "lag" sometimes, one or two days without
> > receiving mails
>
> get a real mail server and it works great!
>
>
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Re: [Asterisk-Users] international calling via POTS in Russia

[John Novack]

Sorry for the late response, but the "w" for wait ONLY works with DTMF. 
Not well documented, but asterisk doesn't detect dialtone, therefore it  
can start to dial numbers before the CO is ready, and I don't know how 
you can wait for a second dialtone if it doesn't even wait for the first 
one!.


Seems the designers thought everyone in the world now used DTMF.

Perhaps someone else has a method that will work. w will not.

John Novack


Balint Kovacs wrote:


Hi Grigoriy,

Thanks for the reply. I have tried to implement this dial pattern by 
dialing from 8w10 to 8ww10, 8p10 (which should be the same as 
8ww10) and even just dialing 8 and sending the rest as DTMF, but it 
doesn't seem to work, all I hear in the line is dead air with 
occasional clicks. Do I have to set something specific in zapata.conf 
(besides pulsedial=yes)? Does the line that I get after dialing 8 
expect pulse or DTMF?


--
Regards,

Balint Kovacs
System Administrator
AES Cargo - MoveOne Relocations


[EMAIL PROTECTED] wrote:


Hello,

There're few POTS supporting touchtone, others - just pulse. In
Russia you need to dial 8, wait for tone and only then continue
dialing 10 (for intl. plan), country code, area code and number.

bkmc> Hi,

bkmc> I'm having a problem calling international numbers with debian's
bkmc> Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem 
to have
bkmc> touchtone dialing, so pulsedial is enabled on my TDM400P 
interface.

bkmc> Local numbers work fine, but when it comes to long distance or
bkmc> international, I'm lost.

bkmc> The prefix for these should be 8 (wait for dialtone) 10 
(country code)

bkmc> (city code) (phone number). I've tried with 8w10, 8p10 and even
bkmc> Dial(Zap/g1/8||D(10${PHONENUM})), but nothing seems to work, I get
bkmc> either dead air (first 2 methods) or a plain dialtone (for the 
last).
bkmc> The Asterisk console shows that exactly the desired number has 
been bkmc> dialed.


bkmc> Any help would be much appreciated. Thanks for reading this mail.

bkmc> P.S. Sorry if this turns out to be a double post, my provider's 
smtp server has

bkmc> sometimes serveral days' delays.

bkmc> --
bkmc> Regards,

bkmc> Balint Kovacs
bkmc> System Administrator
bkmc> AES Cargo - MoveOne Relocations


bkmc> -
bkmc> This mail was sent through IMP: http://horde.org/imp/
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--
Grigoriy Puzankin

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RE : [Asterisk-Users] lists problem, Gmail????????

[Olivier.taylor]

Pfff,

What for an answer :(

I use gmail and have no problems.

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Martin
Joseph
Envoyé : lundi 13 février 2006 20:36
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] lists problem, Gmail



On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:

> C F ha scritto:
>
>> Am I the only one having trouble with this list?
>> Since the begining of the week I have not been receiving mail from 
>> the list like I used to, is this a gmail problem? or is it 
>> subscription problem? or is something wrong with the list? anybody 
>> else using gmail having any problems?
>>
> Yes, I'm also getting some "lag" sometimes, one or two days without
> receiving mails

get a real mail server and it works great!


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Re: [Asterisk-Users] lists problem, Gmail????????

[Martin Joseph]

On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:


C F ha scritto:


Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to, is this a gmail problem? or is it subscription
problem? or is something wrong with the list?
anybody else using gmail having any problems?

Yes, I'm also getting some "lag" sometimes, one or two days without 
receiving mails


get a real mail server and it works great!


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Re: [Asterisk-Users] Best quad-port fxo solution with EC?

[Martin Joseph]

On Feb 13, 2006, at 10:20 AM, Eric "ManxPower" Wieling wrote:


The nearest CO my POTS line goes to is 11 miles away.



i take it you aren't a DSL customer?


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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 89

[Michaël Gaudette]

> Do you also have a SIP phone you are dialing from? 
> This is what I would have setup:
> 
> sip.conf:
> 
> 
> [sipphone]
> Bla
> Bla
> Bla
> context=local-phones
> 
> [someprovider]
> Bla
> bla
> bla
> context=someprovider-in
> 
> extensions.conf
> 
> [local-phones]
> exten => 55,1,Noop(test)
> 
> [someprovider-in]
> exten => s,1,Dial(SIP/sipphone)

That is what I have.  Unfortunately, the context=someprovider-in is being
ignored.  I am running asterisk-1.2.4...

The local-phone context is working properly though.  I can't see why one is
behaving as I expect and the other isn't.


Mike

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Re: [Asterisk-Users] Best quad-port fxo solution with EC?

[Rich Adamson]

> > I am trying to figure out which way to go for a quad port fxo solution
> > with a good echo can on it.  My options are the sangoma remora, a
> > mediatrix fxo, or something similar.
> > 
> > The issue is that I would need a good EC.  This would be on about a 9000
> > foot loop, and the lines don't function well on a spa-3000 or zaptel tdm
> > 4 port card.
> > 
> > Anyone have experience that drives them in a certain direction when
> > considering a good ec on a quad port?
> > 
> > I tried this also with some fxo clones, but echo killed it.
> 
> The nearest CO my POTS line goes to is 11 miles away.  My POTS line 
> works when plugged into my TDM400P FXO port.  I DID have to fiddle with 
> the gains a bit and I still have to get rid of the last of the echo, but 
> overall it seems to work well.

Are you sure the telco is not using fiber-extended line modules?

Have you measured the loss from their milliwatt generator?

(The numbers would be very interesting to see since a large number of
spa3k and tdm04 users at that distance have significant EC issues.)


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RE: [Asterisk-Users] Cisco 2620 as PRI gateway

[Tim Reimers]

Here's a debug---

Longish, but I'm not sure what info in this might be useful to anyone--

I have a zyxel SIP phone configured as ext '6351' on Asterisk--
I can successfully call the SIP phone from an Sjphone client on my PC
and talk between the two-
so the SIP phone is in fact registered with * correctly...

The Cisco router has matching for 63[5-9]x configured- 

Here's a debug from the router:

ACS-GW#
*May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid
type/plan 0x0
0x1 may be overriden; sw-type 13
*May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid
type/plan 0x0
0x0 may be overriden; sw-type 13
*May 26 13:02:42: ISDN Se1/1:23 Q931: Applying typeplan for sw-type 0xD
is 0x4 0
x1, Called num 3506351
*May 26 13:02:42: ISDN Se1/1:23 Q931: TX -> SETUP pd = 8  callref =
0x7A54
Bearer Capability i = 0x8090A2
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18387
Preferred, Channel 7
Calling Party Number i = 0x2181, '8283506180'
Plan:ISDN, Type:National
Called Party Number i = 0xC1, '3506351'
Plan:ISDN, Type:Subscriber(local)
*May 26 13:02:42: ISDN Se1/1:23 Q931: RX <- CALL_PROC pd = 8  callref =
0xFA54
Channel ID i = 0xA98387
Exclusive, Channel 7
*May 26 13:02:43: ISDN Se1/0:23 Q931: RX <- SETUP pd = 8  callref =
0x0014
Bearer Capability i = 0x8090A2
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98383
Exclusive, Channel 3
Calling Party Number i = 0x2181, '8283506180'
Plan:ISDN, Type:National
Called Party Number i = 0x80, '6351'
Plan:Unknown, Type:Unknown
*May 26 13:02:43: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  10.12.1.252:5060;branch=z9hG4bK12A7
From: ;tag=3FB70415-7B4
To: 
Date: Sun, 26 May 2002 18:02:43 gmt
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 2619306202-1879642582-3189047311-607696096
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY
, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID:
;party=calling;screen=yes;privacy=o
ff
Timestamp: 1022436163
Contact: 
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 211

v=0
o=CiscoSystemsSIP-GW-UserAgent 7318 2409 IN IP4 10.12.1.252
s=SIP Call
c=IN IP4 10.12.1.252
t=0 0
m=audio 16396 RTP/AVP 18
c=IN IP4 10.12.1.252
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

*May 26 13:02:43: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP  10.12.1.252:5060;branch=z9hG4bK12A7
From: ;tag=3FB70415-7B4
To: ;tag=as21b1fb71
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Content-Length: 0



*May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> CALL_PROC pd = 8  callref =
0x8014
Channel ID i = 0xA18383
Preferred, Channel 3
ACS-GW#
*May 26 13:02:43: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  10.12.1.252:5060;branch=z9hG4bK12A7
From: ;tag=3FB70415-7B4
To: ;tag=as21b1fb71
Date: Sun, 26 May 2002 18:02:43 gmt
Call-ID: [EMAIL PROTECTED]
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0



*May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> DISCONNECT pd = 8  callref =
0x8014
Cause i = 0x8081 - Unallocated/unassigned number
ACS-GW#
*May 26 13:02:43: ISDN Se1/1:23 Q931: RX <- PROGRESS pd = 8  callref =
0xFA54
Cause i = 0x8281 - Unallocated/unassigned number
Progress Ind i = 0x8288 - In-band info or appropriate now
available
*May 26 13:02:43: ISDN Se1/0:23 Q931: RX <- RELEASE pd = 8  callref =
0x0014
*May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref
= 0x801
4
ACS-GW#
*May 26 13:02:49: ISDN Se1/1:23 Q931: TX -> DISCONNECT pd = 8  callref =
0x7A54
Cause i = 0x8090 - Normal call clearing
*May 26 13:02:49: ISDN Se1/1:23 Q931: RX <- RELEASE pd = 8  callref =
0xFA54
*May 26 13:02:49: ISDN Se1/1:23 Q931: TX -> RELEASE_COMP pd = 8  callref
= 0x7A5
4
ACS-GW#
*May 26 13:03:07: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.12.1.252 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.28:5060;branch=z9hG4bK3741dad8
From: "Unknown" ;tag=as6479f479
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Mon, 13 Feb 2006 18:58:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0



*May 26 13:03:07: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 

Re: [Asterisk-Users] Re: Why is asterisk ignoring my context?

[Joseph Tanner]

If you send it to a different context, you still have to have the
appropriate extension, i.e.:

[test]
Exten => 555-555-,1,NoOp(test)

I've also noticed that with providers that I don't register with, who
just blindly send the call to the same address (i.e., IPKall), context
seems to be ignored.  If the default context is [default], and you
want it to be sent to the [test] context, just use a goto line, i.e.:

[default]
Exten => 555-555-,1,Goto(test,s,1)

And then it'll be sent to:

[test]
Exten => s,1,NoOp(test)

You could send it to any context/extension.  I use this trick to send
calls from multiple providers coming in different ways (iax, sip, zap)
to the same context/extension, so I only have one context to edit
instead of many.

Hope that helps some.

Joseph Tanner

On 2/13/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> Do you also have a SIP phone you are dialing from?
> This is what I would have setup:
>
> sip.conf:
>
>
> [sipphone]
> Bla
> Bla
> Bla
> context=local-phones
>
> [someprovider]
> Bla
> bla
> bla
> context=someprovider-in
>
> extensions.conf
>
> [local-phones]
> exten => 55,1,Noop(test)
>
> [someprovider-in]
> exten => s,1,Dial(SIP/sipphone)
>
>
>
>
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RE: [Asterisk-Users] attended call transfer

[Alex Barnes]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
> Sent: 13 February 2006 18:18
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] attended call transfer
> 
> As I understand it, these issues only apply to DTMF based transfers.
> Why not use the transfer feature of your IP Phone or the Zap FXS port?
> ___


Few reasons really:

1) Only one training method required for all users (we have multiple
types of phone within any single deployment what if a Snom user picks up
a call at a grandstream user's desk).  

2) If we decide to start using a different phone for new deployments we
wouldn't need to reprint training materials.

3) Some phones don't support attended transfer at all.

4) Causes problems when you want to limit the number of calls any device
can make or receive.  For example I have had to alter a script to allow
attended transfers to work but this means that internal -> internal
calls don't count towards the maximum limit.

5) Better CDR maybe, rather than having two calls and then joining them
together Asterisk itself knows it's a transfer so can log the action
much better.

6) The Snom 360 phones we use as reception phones do not behave well
either going from attended to blind (I think) although they don't hang
up the caller but rather than bring this up with Snom I believe this is
a PBX issue.

7) Asterisk shouldn't claim to have attended transfer capability if it
has fundamental flaws.


I cant think of anything else right now but am sure there's more  :-)

But you are entirely correct, there is no option at the moment but to
use the IP phones attended transfer.


HTH

Alex


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RE: [Asterisk-Users] Cisco 2620 as PRI gateway

[Tim Reimers]

Thanks! 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Friday, February 10, 2006 10:51 AM
To: Asterisk
Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway

debug ccsip message

Kurt
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[Asterisk-Users] TDM04B/TDM2401E Card

[housi mueller]

Hi there,     I plan to use Aterisk in our small office. Until now we used a Panasonic D-1232 SuperHybrid System. The figure is representing the future configuration I where thinking about to have in the office.     Question 1: We need only 4 lines and I thought to buy a TDM04B or a TDM2401E card.There is quite a price difference. Which card would you recommend me to buy.     Question 2:Is such a configuration as shown on the figure with a TDM04B/TDM2401E card  at all realizable?     Thanks in adwance     Housi Mueller
		Brings words and photos together (easily) with 
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Re: [Asterisk-Users] segmentation fault

[Sascha]

I had the same problem when I did an SVN of the latest version of the
1.2 branch this past Friday (2/10) and re-made zaptel, libpri and
asterisk.  I don't if someone goofed when updating the supposed stable
Asterisk 1.2 branch or what. But here's what I did to fix it...

I'm using Asterisk at Home. To get back to a working system I first
made a backup of everything through the AMP interface. I made backups
of my zaptel.conf and zapata.conf files to a separate location. Then, I
downloaded the tar version of AAH 2.5 (instead of the ISO). I followed
the instructions to untar and install AAH.  Because AAH doesn't really
have a 100% working restore feature, what I had to do was:
1) go ahead and use the restore function in the AMP interface -
restoring the backup I had just made.
2) Go through every single configuration window within the AMP
interface and click 'submit' to re-apply the settings one by one. This
really only takes like 10 minutes.
3) Then I clicked the red bar along the top to apply the change in
settings.
4) In the process of installing itself, AAH will blow away your
zaptel.conf and zapata.conf files that you spent so long just getting
right. That's why you made backup copies of those to another directory.
;)   Restore those files manually.
5) In order to get my T1 actually functioning I had to not only
shutdown the server but pull the plug on it for a few minutes and then
plug it back in/hit the on button.

Not sure if you're using AAH as well - but hope this helps someone else
who ended up in the same position as I did late Friday night.

Best of Luck,
Sascha



On Mon, 13 Feb 2006 10:02:28 -0500, Patrick Fortin wrote:


Hi
  
  
Asterisk died this morning with this message
  
  
safe_asterisk: line 83:  6828 Segmentation fault  (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY}

Re: [Asterisk-Users] Best quad-port fxo solution with EC?

[Eric \"ManxPower\" Wieling]

[EMAIL PROTECTED] wrote:

Hello All,

I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it.  My options are the sangoma remora, a
mediatrix fxo, or something similar.

The issue is that I would need a good EC.  This would be on about a 9000
foot loop, and the lines don't function well on a spa-3000 or zaptel tdm
4 port card.

Anyone have experience that drives them in a certain direction when
considering a good ec on a quad port?

I tried this also with some fxo clones, but echo killed it.


The nearest CO my POTS line goes to is 11 miles away.  My POTS line 
works when plugged into my TDM400P FXO port.  I DID have to fiddle with 
the gains a bit and I still have to get rid of the last of the echo, but 
overall it seems to work well.

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Re: [Asterisk-Users] attended call transfer

[Eric \"ManxPower\" Wieling]

Richard Perini wrote:

On Sun, Feb 12, 2006 at 03:35:56PM -0500, John Novack wrote:
That certainly is the way it SHOULD work. Blind and attended transfer 
should be able to be initiated the same way.  It certainly is the most 
efficient logical way. Attended transfer should revert to blind simply 
by the initiating party hanging up.

Most "legacy" hybrid key/pbx systems work that way, and have for many a year
Most users expect transfer to work that way.

I would consider that a defect or bug, not a new feature request.


Absolutely.  I have never used a conventional PBX system that didn't 
behave that way (or wasn't able to be programmed to do so).  I'm in

my early days of trialling Asterix, and our first application will be
an internal intercompany/interoffice tie-line system where the transfer
feature will be rarely required.  Certainly this issue would be a 
show-stopper for full PBX replacement for us in the future.


As I understand it, these issues only apply to DTMF based transfers. 
Why not use the transfer feature of your IP Phone or the Zap FXS port?

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[Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel

[Hans-Juergen Brand]

My hardware configuration looks like this:

public network --->>other PBX ---analog line (ext 100) ---> FXO,Asterisk
  |
  |--- extension
 (200)

by calling extension 100 from public network I can call Asterisk. Now I
would like to answer the call, SayNumber and the transer back to the
extension 200 in the other pbx. When I put a telephone instead of Asterisk
on the line 100 I can do the transfer by pressing Hookflash and the dial 200
and then hookon. The call is no at 200.

 extension.conf ---

exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,SayNumber(101)
exten => s,4,Wait(2)
exten => s,5,Transfer(**51)
exten => s,6,Wait(40)   

--

I trie something, but it is not working.

kinds regards.

HJB
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RE: [Asterisk-Users] TAPI Recommendations

[Bob McDowell]

I have installed AstManProxy, and it seems a bit better from within
Outlook, but things are the same from my application.  The error the
application gives me is 'LINE ERR NOT OWNER' or something similar.  I
would have copy-pasted, but I went into a hard lock after closing and
re-opening Outlook...

Also, pressing the 'Redial' button from within the app crashes the app.

I see no errors in "/var/log/asterisk/astmanproxy.log".

I will do the packet capture and see what it tells me.

Would anyone happen to know how I might isolate TAPI issues from the
application?  I can submit a change request to the application's
developers IF I can convince them it is their issue, not Asterisk's.


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Davies
Sent: Monday, February 13, 2006 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TAPI Recommendations

On 2/13/06, Bob McDowell <[EMAIL PROTECTED]> wrote:
>
> The issue appears to be something on the XP desktop side. I can
> end-task and restore TAPI functionality about 75% of the time.
> Otherwise, a reboot always clears it up.
>
> I'm unfamiliar with astmanproxy.  I'll look it up.
>
> I removed siptapi just after I determined that I couldn't get it
> working...

It may be the two ends becoming "out of sync" in some fashion - Perhaps
you should try capturing a trace of the TCP communication, and see what
happens just before the hang.

Best of luck.

Regards,
Steve
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[Asterisk-Users] Re: Why is asterisk ignoring my context?

[Bromont Quebec]

Do you also have a SIP phone you are dialing from? 
This is what I would have setup:

sip.conf:


[sipphone]
Bla
Bla
Bla
context=local-phones

[someprovider]
Bla
bla
bla
context=someprovider-in

extensions.conf

[local-phones]
exten => 55,1,Noop(test)

[someprovider-in]
exten => s,1,Dial(SIP/sipphone)




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[Asterisk-Users] Oh323, opengk and asterisk

[Olivier.taylor]

Hi all,

Well, i need h323 and asterisk working together.

I have asterisk with oh323 working
I have opengk installed on the same server (working too).
I have a h323 handset (swissvoice ip10s)

The swissvoice register with opengk (don't ask me how)..
I need opengk register with asterisk to have the opportunity to relay
the calls to a sip pstn gateway.

I googled a lot but didn't find any solution or samples.
Just to avoid wasted time, does any of you have an opengk config file
and an asterifk config file making possible to have a working solution?

Thanks for all,

Olivier


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Re: [Asterisk-Users] TAPI Recommendations

[Steve Davies]

On 2/13/06, Bob McDowell <[EMAIL PROTECTED]> wrote:
>
> The issue appears to be something on the XP desktop side. I can end-task
> and restore TAPI functionality about 75% of the time.  Otherwise, a
> reboot always clears it up.
>
> I'm unfamiliar with astmanproxy.  I'll look it up.
>
> I removed siptapi just after I determined that I couldn't get it
> working...

It may be the two ends becoming "out of sync" in some fashion -
Perhaps you should try capturing a trace of the TCP communication, and
see what happens just before the hang.

Best of luck.

Regards,
Steve
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RE: [Asterisk-Users] TAPI Recommendations

[Bob McDowell]

The issue appears to be something on the XP desktop side. I can end-task
and restore TAPI functionality about 75% of the time.  Otherwise, a
reboot always clears it up.

I'm unfamiliar with astmanproxy.  I'll look it up.

I removed siptapi just after I determined that I couldn't get it
working...


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Davies
Sent: Monday, February 13, 2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TAPI Recommendations

On 2/13/06, Bob McDowell <[EMAIL PROTECTED]> wrote:
>
> Does anyone on the list have a recommendation for a TAPI interface to
> Asterisk?  I have tried all of the ones that Google produced, but have

> still not yet found a solution that I can move into production.  My
> favorite to date is AstTapi, but it dies after five-or-so calls.
>

Do you know which component is dead after 5 or so calls? Is it AstTAPI?
Or is it the manager interface? How many clients are you connecting so
far? Also, are you using a manager proxy such as astmanproxy?

I have not rolled this out on a production scale, but using AstTAPI and
astmanproxy, I have so far had no stability problems (Windows XP
Home/Pro clients) in my test environment.

If you have installed "siptapi.tsp" you might want to remove it - I
found that that reduced the stability of my TAPI setup, even when not in
use (I assume some basic initialisation is still done if the TSP is
present)

Regards,
Steve
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Re: [Asterisk-Users] TAPI Recommendations

[Steve Davies]

On 2/13/06, Bob McDowell <[EMAIL PROTECTED]> wrote:
>
> Does anyone on the list have a recommendation for a TAPI interface to
> Asterisk?  I have tried all of the ones that Google produced, but have
> still not yet found a solution that I can move into production.  My
> favorite to date is AstTapi, but it dies after five-or-so calls.
>

Do you know which component is dead after 5 or so calls? Is it
AstTAPI? Or is it the manager interface? How many clients are you
connecting so far? Also, are you using a manager proxy such as
astmanproxy?

I have not rolled this out on a production scale, but using AstTAPI
and astmanproxy, I have so far had no stability problems (Windows XP
Home/Pro clients) in my test environment.

If you have installed "siptapi.tsp" you might want to remove it - I
found that that reduced the stability of my TAPI setup, even when not
in use (I assume some basic initialisation is still done if the TSP is
present)

Regards,
Steve
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[Asterisk-Users] AAH 2.5 pone paging broken

[Kerry Garrison]

Using some scripts 
that have been posted, we have been able to get paging to phones working quite 
nicely. However, with a few [EMAIL PROTECTED] 2.5 installs, (Aserisk 1.2.4) the 
phones ring but never pick up. Any ideas on why or how to tweak the scripts to 
get the phone paging working again?
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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Re: [Asterisk-Users] Bug in AMP 1.10.010 in sip outbound callerid

[asterisk]

I tried to define the iax junction via AMP itself, instead that manually.

Doing this work, and the incorrect device <567> is replaced with what I
want to see <567>

The only problem is the need to redefine all the iax connection between the
various * boxes, moving them form iax_custom to
the amp interface.

So probably the bug is not a bug, but a feature avoiding you to manually
edit filesUnfortunately in version 1.10.008
the behaviour was different, and the callerid defined in sip was used
everywhere.

Anyway now it's clear to me, and I am moving anything to AMP.

Andrea



   
 [EMAIL PROTECTED] 
 .it   
 Sent by:   To 
 asterisk-users-bo asterisk-users@lists.digium.com 
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   [Asterisk-Users] Bug in AMP 
 13/02/2006 17.14  1.10.010 in sip outbound callerid   
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




If you define a sip peer, wheather or not you put an entry in the field
OUTBOUND CID, if you
dial an external extension (let's say an extension on another asterisk
server, connected via IAX2 connection) the callerid
received by the foreign asterisk is device : i.e device <567>

If you take a look at etc/asterisk/sip_additional.conf, you can see under
the SIP extension defined the line callerid=device <567>.
If you look at the mysql tables, the only place where the field outbound
CID you entered is recorded is in the users table.

If you manually edit  etc/asterisk/sip_additional.conf and adjust the line
callerid=what I would like to see <567>, then reload
the configuration with RELOAD from * console (NOT using web interface AMP,
otherwise you loose your modification)
you can see on the remote phone what I would like to see <567>

This bug is new to 1.10.010, or at least was not present in the last amp
version, where the fieldname was called callerid (and not outbound
callerid)

I would like to know if this bug can be solved/workarounded defining IAX2
trunks in amp;
actually I define IAX2 connection between * box manually, editing iax.conf.

Thanks in advance,

Andrea


Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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[Asterisk-Users] Asterisk 1.2.4 Quality Issues

[Adam Robins]

 
We have (had) two identical Asterisk servers for our outbound call
center.  Both were running Linux 2.4 kernel,  Asterisk 1.0.7, Libpri
1.0.7 and Zaptel 1.2.1.  Each server has a TE410P card with two PRIs.

Last week, we upgraded one of them to Asterisk 1.2.4, Zaptel 1.2.3,
Libpri 1.2.2.  

The agents on the new system suddenly started complaining that calls
were cutting in/out, and that customers were having problems hearing
them.  We then downgraded zaptel back to 1.2.1, but no improvements.  If
I move the agents over to the old 1.0.7 server, they have no issues.

Has anyone had similar issues?  Would downgrading Libpri help anything?

Thanks,
Adam

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[Asterisk-Users] TAPI Recommendations

[Bob McDowell]

Does anyone on the list have a recommendation for a TAPI interface to
Asterisk?  I have tried all of the ones that Google produced, but have
still not yet found a solution that I can move into production.  My
favorite to date is AstTapi, but it dies after five-or-so calls.

Thank you very much in advance for your insight...


Thanks,

Bob McDowell



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Re: [Asterisk-Users] attended call transfer

[Lenz]

I believe many users would benefit from such an improvement. That's the  
kind of annoyance that is usually not present in a decent PBX and that  
people have the worst time getting used to (at least in my experience).

l.

On Mon, 13 Feb 2006 14:46:35 +0100, Alex Barnes  
<[EMAIL PROTECTED]> wrote:





-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: 13 February 2006 00:58
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
Commercial Discussion
Subject: RE: [Asterisk-Users] attended call transfer

Questions for the community: is an "integrated" transfer feature
valuable to you?  If so, would you be willing to put out a bounty?

(In

other words, is it just a nice feature or is it so important that

you'd

be willing to pay a few bucks for it...)  Last question, but possibly
the most important: what have you done, if anything, to get around the
split between blind and attended transfers?

-MC



To get around this issue we have had to only use attended transfer with
Snom phones which are easy to train our users on.

People with DECT / ??? phones have been told to only ever use blind
(only about 10 out of 50 extensions so not the end of the world).


We would definitely chip in some money for this to happen but again it
would have be in the stable release soon as we cant deploy anything
except stable.


Cheers

Alex


--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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Re: [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

[nik600]

On 2/13/06, nik600 <[EMAIL PROTECTED]> wrote:
> On 2/13/06, Michael Collins <[EMAIL PROTECTED]> wrote:
> > When Asterisk first starts up, it will attempt to "bring up" the B
> > channels on any PRI circuits.  If you are using [EMAIL PROTECTED] then you 
> > can log on
> > to the Asterisk CLI (asterisk -r) and then do "stop now" to stop
> > asterisk.  Start up Asterisk again by typing asterisk -cvvv at the Linux
> > command line.  You should see a bunch of messages on the terminal and
> > then you'll get the Asterisk CLI prompt.  A few moments later you should
> > see some more messages saying something like "B channel 1/1 successfully
> > started on span 0" and then "B channel 1/2 successfully started..."  I
> > don't have my system active at the moment but when I do I'll email you a
> > screen capture of the messages I see on my PRI.
> >
>
> i get:
> *CLI>   == Primary D-Channel on span 1 up
>
> at the moment i've got only one span connected...il the message correct?
>
Description  Alarms IRQbpviol 
   CRC4
T2XXP (PCI) Card 0 Span 1NOP0  0  
   0
T2XXP (PCI) Card 0 Span 2RED/NOP0  0  
   0
ZTDUMMY/1 1  UNCONFIGUR 0  0  0
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Re: [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

[nik600]

On 2/13/06, Michael Collins <[EMAIL PROTECTED]> wrote:
> When Asterisk first starts up, it will attempt to "bring up" the B
> channels on any PRI circuits.  If you are using [EMAIL PROTECTED] then you 
> can log on
> to the Asterisk CLI (asterisk -r) and then do "stop now" to stop
> asterisk.  Start up Asterisk again by typing asterisk -cvvv at the Linux
> command line.  You should see a bunch of messages on the terminal and
> then you'll get the Asterisk CLI prompt.  A few moments later you should
> see some more messages saying something like "B channel 1/1 successfully
> started on span 0" and then "B channel 1/2 successfully started..."  I
> don't have my system active at the moment but when I do I'll email you a
> screen capture of the messages I see on my PRI.
>

i get:
*CLI>   == Primary D-Channel on span 1 up

at the moment i've got only one span connected...il the message correct?
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RE: [Asterisk-Users] Aastra phones and common directory?

[Bob McDowell]

I like the 9133i better than the Polycom 301 for
a similar price.  It is very similar to the Norstar equipment we have
now.  The e-mail support has been great for me.
 
I had some issues getting it set up, and am still a little
concerned about the echo and side tone.  I personally can tune them to be
acceptable, but a hyper-sensitive user may not be able to get past
them...
 
I will
know more if/when we proceed to roll-out.
 
Thanks,
Bob McDowell
 


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Sunday, February 12, 2006 10:37
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionCc: [EMAIL PROTECTED]Subject: Re:
[Asterisk-Users] Aastra phones and common directory?

Carlos,
 
I'm planning to use the Aastra 9133i for a new installaton. 
 
Can u please comment on your experiences with this equipment. Please let me
know if u have found any specific issues with it.
 
thanks in advance. 
On 13/02/06, Ira
<[EMAIL PROTECTED]>
wrote:
At
  01:49 PM 02/12/2006, you
  wrote:>  Does anyone know if it is
  possible to upload a common directory to all >Aastra phones (480i,
  9133)?  Is there someting equivalent to the way
  Polycom>phones do it?If there is, it's in the recently released
  XML documentation whichyou can find in the support section of the Aastra
  web site. Ira--No virus found in this outgoing
  message.Checked by AVG Anti-Virus.Version: 7.1.375 / Virus Database:
  267.15.6/257 - Release Date:
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Re: [Asterisk-Users] more cpu intensive echo cancellers ?

[Gerard Saraber]

On Fri, 2006-02-10 at 16:05 -0600, Matthew Fredrickson wrote:
> On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:
> >
> > Found it, going to go test it right now :) thanks!
> > So far reports have been positive on the echo, but its a slow day ;)
> > We're using cisco 7960 phones, they're pricy, but they work great and
> > sound good, if it wasn't for the echo issue, I would have been able to
> > roll the whole setup out already.
> > Actually that's not quite true, I still have to make the 7914 addon
> > module work with the 7960 phone, but that's not a show stopper.
> >
> > Either way, so far big thumbs up for the MG2 echo can, and if any
> > developers read this, feel free to add a compile flag to make it more
> > cpu intensive ;) and do more canceling.
> >
> 
> Does latest MG2 behave better than KB1 on your analog lines?  I heard 
> in the past that in some cases (primarily with analog lines) that KB1 
> worked better.  Also, have you tried the echotraining=800  (in 
> zapata.conf) tweak as well?
> 
> ---
> Matthew Fredrickson

In my case, MG2 blows KB1 away, the trunk version is a huge improvement,
in the past, echotraining= was always worse compared to
echotraining=yes so I didn't change it. I'll definately try that if I
get any echo complaints, so far, so good though.

-- 
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]

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RE: [Asterisk-Users] Aastra phones and common directory?

[Bob McDowell]

In the server's (or the phone's) cfg file, put in this line:

directory 1: company_directory.cfg

Then add a file of that name to your server.  I'm currently using a
basic csv format file that consists of:

Name,1234


Thanks,

Bob McDowell


Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Sunday, February 12, 2006 8:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Aastra phones and common directory?

At 01:49 PM 02/12/2006, you wrote:
>  Does anyone know if it is possible to upload a common directory
>to all Aastra phones (480i, 9133)?  Is there someting equivalent to the

>way Polycom phones do it?

If there is, it's in the recently released XML documentation which you
can find in the support section of the Aastra web site.

Ira


--
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Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database: 267.15.6/257 - Release Date:
02/10/2006


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Re: [Asterisk-Users] PrivacyManager Broken?

[John Novack]

Jeremy G. Gault wrote:


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi all,

I am running into some problems here with PrivacyManager.  We used to use it 
without any issue, but now there seems to be several problems.

We are currently running Asterisk 1.2.4.

First, it seems that if the user does not press the pound (#) key after 
entering their number, PrivacyManager will fail.  I have the minlength set to 
10, and entering 10 digits doesn't work.  Entering 11 digits doesn't either.  
But entering 10 followed by the pound key will.

Second, once the user figures out how to operate PrivacyManager, there's 
another problem: My extension rings to a SIP extension (Polycom IP phone) and a 
Zap extension (we have a 4-port card with 4 different extensions on it -- one 
of which is my cordless phone.)

On the cordless phone (through the Zap channel) the caller ID shows
"Privacy Manager" along with the user-specified number.  However, on the IP phone, it 
simply shows "unknown"

Is this just us, or is PrivacyManager not working correctly anymore?
Any help would be appreciated.

Jeremy


Has it EVER worked correctly?
From the Wiki and my experience with 1.2b1, if the callerid number 
isn't a number, but a word ( such as asterisk ) it doesn't even come 
into play, acting as if there were a valid phone number.  The Wiki had a 
"fix" that again according to the Wiki, fixes that problem but breaks 
the callerid name
IMO, Privacy manager should come into play if the callerid number field 
doesn't contain a string of digits, or, even better, a string of digits 
of a length that can be specified, all without breaking the callerid 
name..Replacing any callerID name delivered with "privacymanager" seems 
unnecessary.



John Novack


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[Asterisk-Users] Call over SIP channel becomes a zombie

[Stephane Ricard]

Hi all,

 

I am running 1.0.10 and calls over a SIP channel more than
often become zombies (call is still there but no more sound.  I get the
following:

 

“Unable to parse INFO message from
[EMAIL PROTECTED] Content”

 

And then the call continues but no more sound (my zombie
analogy). 

 

I have this problem for a few months now and it started to
appear in version 1.0.7 and it is still there with 1.2.4 so it brings me to
believe it may be something with my local configuration but I can’t find
it. I am running Asterisk on Gentoo.

 

Thanks

Stephane

[EMAIL PROTECTED]

 

 






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[Asterisk-Users] Asterisk register ip phone

[al gav]

Hi all     I have a problem to register a cisco 7960 to an asterisk 1.2.2     I defined in sip.conf the next :  ["phonenumber"]type=friendusername="username"secret="password"host=dynamiccontext=work     I am trying to catch the register requests with   sip debug  with no success (empty screen).     I can only catch the register messages with ngrep on host it's comming from.     #U CISCO_IP:50339 -> ASTERISK_IP:5060  REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK75640688..From: sip:[EMAIL PROTECTED]:  sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]: 41929 REGISTER..User  -Agent: CSCO/6..Contact:
 ..Content-Length: 0..Expires: 1200    #I ASTERISK_IP -> CISCO_IP 3:10  E..}a...>..o.rC...;;.i..REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK75640688  ..From: sip:[EMAIL PROTECTED]: sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]:  41929 REGISTER..User-Agent: CSCO/6..Contact: ..Content-Length: 0..Expires: 1200 
    #U CISCO_IP:50341 -> ASTERISK_IP:5060  REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3e649d37..From: sip:[EMAIL PROTECTED]:  sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]: 42010 REGISTER..User  -Agent: CSCO/6..Contact: ..Content-Length: 0..Expires: 1200    #I ASTERISK_IP -> CISCO_IP 3:10  E..}a...>..n.rC...;;.i..REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3e649d37  ..From: sip:[EMAIL PROTECTED]: sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]:   42010 REGISTER..User-Agent: CSCO/6..Contact: ..Content-Length: 0..Expires: 1200          If
 there any way to find what's the reason why i can not register the phone ??     Thanks for the help.
		  
What are the most popular cars? Find out at Yahoo! Autos 
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[Asterisk-Users] meetme announcement

[Barry Porch]

Hello,

 

Could someone tell me how to play an announcement (gsm file) to all of the members of a meetme
conference?

 

Thank you!

 

Barry






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[Asterisk-Users] Why is asterisk ignoring my context?

[Michaël Gaudette]

Hi,

I've been fighting with a sip configuration for a few days, and I just
realized why it wasn't working.

In my sip.conf, I have the following

[someprovider]
Bla
Bla
Bla
Bla

And in my extensions.conf file, I have this

Exten => 555-555-,1,Noop(test)

Sure enough, when I dial 555-555-, it works.  What DOESN'T work is if I
use an extension in the sip.conf and extensions.conf.  If I change my
sip.conf file to :

[someprovider]
Bla
Bla
Bla
Context=test

And in my extensions.conf, I add
[test] 

It doesn't work.  Further investigation shows me that if I remove the [test]
context in extensions.conf, It works REGARDLESS of whether I have the
context defined in sip.conf

In other words, it seems like whatever I put in sip.conf as a context for
those incoming calls, Asterisk just tries to find the extension in the
[general] context.

Is this
1) a known bug?
2) a misunderstanding on my part of hos contexts work?
3) a bad dream?

Mike

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RE: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

[gw]

Hello There,
Yes I have a tellabs installed, in fact I may have been one of those who helped 
you out :) 

What I need though is only 4 ports, that's a bit overkill.  I also did the spa 
and tdm400 with little luck.

Greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, February 13, 2006 9:15 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

In my expereince a channel bank with a digium single span card and a Tellabs EC 
perfomed the best, but is too expensive (it gives you a minimum of 8 ports). 
Next to that I use a mediatrix 1204, and compared to all others I have tried 
works best. I have tried:
Sipura SPA3000
Digium TDM400 with 4 FXO mods


On 2/13/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have good results with the new TDM2400P serie (with the hardware 
> echocan, of course).
> May be you must check one TDM2401E to see if it's ok for you...
>
> Good luck.
>
> Best Regards,
> Francois BERGERET,
> France.
>
>
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de 
> [EMAIL PROTECTED] Envoyé : lundi 13 février 2006 07:36 À : 
> asterisk-users@lists.digium.com Objet : [Asterisk-Users] Best 
> quad-port fxo solution with EC?
>
>
> Hello All,
>
> I am trying to figure out which way to go for a quad port fxo solution 
> with a good echo can on it.  My options are the sangoma remora, a 
> mediatrix fxo, or something similar.
>
> The issue is that I would need a good EC.  This would be on about a 
> 9000 foot loop, and the lines don't function well on a spa-3000 or 
> zaptel tdm 4 port card.
>
> Anyone have experience that drives them in a certain direction when 
> considering a good ec on a quad port?
>
> I tried this also with some fxo clones, but echo killed it.
>
> Thanks,
> Greg
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Re: [Asterisk-Users] Re: asterisk logger - urgent!!!

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Mon, 13 Feb 2006, Simone Cittadini wrote:


Dov Bigio ha scritto:


 I found the problem.

 Master.csv reached 2.0GB and since the moment this happened Asterisk went
 crazy!

 Since I am using cdr-mysql, how do I disable the use of csvs?


In /etc/asterisk/modules.conf:
noload => cdr_csv.so

will turn off logging to the .csv file



 Thank you
 Dov


Why don't you simply rotate the logs with logrotate ?
(no, I don't know how to disable them)
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- -- 
Ron Wellsted

[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
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Re: [Asterisk-Users] segmentation fault

[Justin Tunney]

On Mon, 13 Feb 2006 10:02:28 -0500, Patrick Fortin <[EMAIL PROTECTED]>  
wrote:



Hi

Asterisk died this morning with this message

safe_asterisk: line 83:  6828 Segmentation fault  (core dumped)  
asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY} 

Any idea what is the problem ?


You should post this on the bugtracker (bugs.digium.com) along with a back  
trace.  See asterisk-sources/doc/README.backtrace for info on how to do a  
backtrace.


  Justin Tunney
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[Asterisk-Users] FXO port on TDM400P hangs

[Cosmin Prund]

Hello everyone, I just ran into one big ugly problem: one of the FXO ports
on my TDM400P does something that causes my telco to mark my line as
deffective! Before installing the * I only ran into this situation when I
managed to "short" the two wires from the telco. Only this time the lines
are not shorted (at least not outside the TDM400P) because the problem goes
away with a reboot!

At the time the problem manifests itself "show channels" shows no active
channels, so as far as * is concerned, the line is not in use.

Where do I look?
What do I do?

Thanks!

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[Asterisk-Users] trunk 2 IAX server :- getting error ' Unable to support trunking on user 'ho' without zaptel timing'

[John Joseph]

Hi All 
   I am using RHEL , kernel 2.6.9-5, asterisk
1.2.4 , zaptel  1.2.3 installed , when I give modprobe
for zaptel and ztdummy , I do not any error message 
 my   iax.conf contains the entry for trunking
as 
[hoportal]
type=friend
host=192.168.20.32
secret=mysecret
context=local
trunk=yes

 my extensions.conf contains the entry for 
trunking as 
exten =>
_3XXX,1,Dial(IAX2/hoportal:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _3XXX,2,Hangup
exten => _3XXX,102,Hangup

When I start the asterisk , I get this message in
/var/log/asterisk/message as 
 Feb 13 18:22:19 WARNING[3531] chan_iax2.c: Unable to
support trunking on user 'hoportal' without zaptel
timing
Feb 13 18:22:19 WARNING[3531] chan_iax2.c: Unable to
support trunking on peer 'hoportal' without zaptel
timing

I am stuck over here , I had tried google, but not
able to find the reason why I get this errot message
and why I cannot  get connected to the other machine [
the other asterisk server also gives the same error ]
  requesting  guidance 
Thanks 
 Joseph John 





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RE: [Asterisk-Users] automatically start application from the commandprompt

[Wai Wu]

You 
can do this via the Manager API

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Arjan 
  KroonSent: Monday, February 13, 2006 10:10 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] 
  automatically start application from the commandprompt
  
  Hello,
   
  Is it possible to 
  start an asterisk application from the command prompt? 
  
  This application has 
  to dial to a number.
  When the calling 
  party picks up the phone, the asterisk application had to play certain 
  voicefiles.
   
  Kind 
  Regards,
   
  Arjan 
  KroonMobillion B.V. Copernicuslaan 
  30 Postbus 554 / PO 
  Box 554 6710 BN Ede tel: +31 (0)318-648920 
  fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: 
  [EMAIL PROTECTED] internet: www.mobillion.nl 
   
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[Asterisk-Users] automatically start application from the command prompt

[Arjan Kroon]

Hello,

 

Is it possible to start an asterisk application
from the command prompt? 

This application has to dial to a number.

When the calling party picks up the phone,
the asterisk application had to play certain voicefiles.

 

Kind Regards,

 

Arjan Kroon

Mobillion B.V. 
Copernicuslaan 30 
Postbus 554 / PO Box
 554 
6710 BN Ede 
tel: +31 (0)318-648920 
fax: +31 (0)318-648839 
mobile: +31 (0)6-55871460 
email: [EMAIL PROTECTED] 
internet: www.mobillion.nl 

 






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[Asterisk-Users] segmentation fault

[Patrick Fortin]

Hi

Asterisk died this morning with this message

safe_asterisk: line 83:  6828 Segmentation fault  (core dumped) 
asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY} 

Any idea what is the problem ?

here is a show channels before the crash

SIP/131-f5ad (None)   Ringing AppDial((Outgoing 
Line))
SIP/123-8bc1 [EMAIL PROTECTED]:1 RingDial(SIP/131|16|tr) 

Zap/11-1 [EMAIL PROTECTED]:2Up  Queue(reception|t|||300) 

Zap/10-1 [EMAIL PROTECTED]:2Up  Queue(reception|t|||300) 

Zap/3-1  [EMAIL PROTECTED]:1  Up  Bridged 
Call(SIP/137-adb6)
SIP/137-adb6 [EMAIL PROTECTED]:2 
Up  Dial(Zap/G1/915143334233)
Zap/8-1  [EMAIL PROTECTED]:2 Up  Queue(support|t|||300) 

Zap/7-1  [EMAIL PROTECTED]:2Up  Queue(reception|t|||300) 

SIP/141-178a [EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(141|@zapout
SIP/141-f5f6 [EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(141|@zapout
SIP/141-5371 [EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(141|@zapout
Zap/4-1  [EMAIL PROTECTED]:2Up  Queue(reception|t|||300) 

Agent/111[EMAIL PROTECTED]:1  Down(None) 

Local/[EMAIL PROTECTED] 
[EMAIL PROTECTED]:1Ring(None)
Local/[EMAIL PROTECTED] 
[EMAIL PROTECTED]:1  Down(None)
Zap/6-1  [EMAIL PROTECTED]:2Up  Queue(reception|t|||300) 

Zap/2-1  [EMAIL PROTECTED]:2Up  Queue(reception|t|||300) 



The queue application seemed to be crashed because I got no output from a 
show agents or a show queues


Patrick

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[Asterisk-Users] asterisk still tries native bridging

[Igor Zamocky]

  Hello,

  I've problems with following -

  - --- ---
  PSTN |  --- isdn --- | A | - iax2 -- | B |
  - --- ---

   On [B], there is unconditional call forwarding set back via [A]
   (dialparties.agi is used) to PSTN.
   So, call from PSTN is routed via [A] to [B] and than back again into
   PSTN.
   
   Everything looks good, but, after call is answered, B performs native
   bridging attempt and tries to step out of voice path. And that's bad.
   Because of CDR's collected from [B].

   On [B] and also on [A] there is "notransfer=yes" in [general] section and
   also in [peer/friend] definition.
   It probably doesn't work. I tried to use different iax2 peer for [B]->[A]
   call, so native bridging cannot occur. Fine, native bridging will fail,
   but Asterisk still writes CDR.

   Below is part of [B]'s config, and part of log:

[general]
bindport = 4569
bindaddr = x.x.x.x
disallow=all
allow=alaw
notransfer=yes
jitterbuffer=yes

; ---
register=sip1:[EMAIL PROTECTED]  ; y.y.y.y is [A]'s ip address
; ---
[peerA]
username=sip1
type=friend
secret=Q
host=y.y.y.y
context=from-pstn
tos=0x84
notransfer=yes
jitterbuffer=yes

[peerAX]
username=sip1
type=peer ; I tried "friend" also
secret=QQ
host=y.y.y.y
context=from-pstn
tos=0x84
disallow=all
allow=ulaw
notransfer=yes
jitterbuffer=yes   

So, incoming call comes via "peerA" (alaw), outgoing is made via "peerAX"
(ulaw).

Feb 13 15:25:48 DEBUG[27671]: Setting NAT on RTP to 4
Feb 13 15:25:48 DEBUG[27671]: Stopping retransmission on '[EMAIL PROTECTED]' of 
Request 102: Found
Feb 13 15:25:51 VERBOSE[27671]: -- IAX2/peerAX/6 is ringing
Feb 13 15:25:51 VERBOSE[27671]: -- Local/[EMAIL PROTECTED],1 is ringing
Feb 13 15:25:53 VERBOSE[27671]: -- IAX2/peerAX/6 answered Local/[EMAIL 
PROTECTED],2
Feb 13 15:25:53 VERBOSE[27671]: -- Local/[EMAIL PROTECTED],1 answered 
IAX2/[EMAIL PROTECTED]/3
Feb 13 15:26:00 DEBUG[27671]: Planning to masquerade IAX2/peerAX/6 into the 
structure of Local/[EMAIL PROTECTED],1
Feb 13 15:26:00 DEBUG[27671]: Done planning to masquerade Local/[EMAIL 
PROTECTED],1 into the structure of IAX2/peerAX/6
Feb 13 15:26:00 DEBUG[27671]: Actually Masquerading IAX2/peerAX/6(6) into the 
structure of Local/[EMAIL PROTECTED],1(6)
Feb 13 15:26:00 DEBUG[27671]: Got clone lock on 'IAX2/peerAX/6' at 0x8ec0ce0
Feb 13 15:26:00 DEBUG[27671]: Putting channel IAX2/peerAX/6 in 8/8 formats
Feb 13 15:26:00 DEBUG[27671]: Released clone lock on 'Local/[EMAIL 
PROTECTED],1'
Feb 13 15:26:00 DEBUG[27671]: Done Masquerading IAX2/peerAX/6 (6)
Feb 13 15:26:00 DEBUG[27671]: Bridge stops because we're zombie or need a soft 
hangup: c0=Local/[EMAIL PROTECTED],2, c1=Local/[EMAIL PROTECTED],1, 
flags: No,No,Yes,Yes
Feb 13 15:26:00 VERBOSE[27671]: -- Attempting native bridge of IAX2/[EMAIL 
PROTECTED]/3 and IAX2/peerAX/6
Feb 13 15:26:00 VERBOSE[27671]: -- Operating with different codecs, can't 
native bridge...
Feb 13 15:26:00 DEBUG[27671]: Bridge stops bridging channels Local/[EMAIL 
PROTECTED],2 and Local/[EMAIL PROTECTED],1
Feb 13 15:26:00 DEBUG[27671]: Exiting with DIALSTATUS=ANSWER.
Feb 13 15:26:00 VERBOSE[27671]:   == Spawn extension (macro-outsideX, s, 6) 
exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'outsideX'
Feb 13 15:26:00 VERBOSE[27671]:   == Spawn extension (from-internalX, 
ZZ, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2'
Feb 13 15:26:00 DEBUG[27671]: cdr_mysql: inserting a CDR record.
Feb 13 15:26:00 DEBUG[27671]: cdr_mysql: SQL command as follows:  INSERT INTO 
cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
 VALUES ('2006-02-13 
15:25:46','421220656111','421220656111','ZZ','from-internalX', 
'Local/[EMAIL 
PROTECTED],2','IAX2/peerAX/6','Dial','IAX2/peerAX/||',14,7,'ANSWERED',3,'')


Asterisk stored CDR, but call continued :-(.

Do You have any suggestion what I'm doing wrong?

I'm using Asterisk v. 1.0.9 and it's almost impossible to upgrade to 1.2.x 
right now.

Thanks a lot

Igor

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[Asterisk-Users] PrivacyManager Broken?

[Jeremy G. Gault]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi all,

I am running into some problems here with PrivacyManager.  We used to
use it without any issue, but now there seems to be several problems.

We are currently running Asterisk 1.2.4.

First, it seems that if the user does not press the pound (#) key after
entering their number, PrivacyManager will fail.  I have the minlength
set to 10, and entering 10 digits doesn't work.  Entering 11 digits
doesn't either.  But entering 10 followed by the pound key will.

Second, once the user figures out how to operate PrivacyManager, there's
another problem: My extension rings to a SIP extension (Polycom IP
phone) and a Zap extension (we have a 4-port card with 4 different
extensions on it -- one of which is my cordless phone.)

On the cordless phone (through the Zap channel) the caller ID shows
"Privacy Manager" along with the user-specified number.  However, on the
IP phone, it simply shows "unknown"

Is this just us, or is PrivacyManager not working correctly anymore?
Any help would be appreciated.

Jeremy

- --
- -
Jeremy Gault, KD4NED | Network Administrator/Telecomms
+1 (423) 303-2562 voice  | WinWorld Corporation
+1 (423) 472-9265 fax| http://www.winworld.com/
- -
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RE: [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

[Michael Collins]

When Asterisk first starts up, it will attempt to "bring up" the B
channels on any PRI circuits.  If you are using [EMAIL PROTECTED] then you can 
log on
to the Asterisk CLI (asterisk -r) and then do "stop now" to stop
asterisk.  Start up Asterisk again by typing asterisk -cvvv at the Linux
command line.  You should see a bunch of messages on the terminal and
then you'll get the Asterisk CLI prompt.  A few moments later you should
see some more messages saying something like "B channel 1/1 successfully
started on span 0" and then "B channel 1/2 successfully started..."  I
don't have my system active at the moment but when I do I'll email you a
screen capture of the messages I see on my PRI.

-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Monday, February 13, 2006 6:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problem with outgoing calls Unable
tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

On 2/13/06, Michael Collins <[EMAIL PROTECTED]> wrote:
> Nik,
>
> Just curious - what is your telco setup?  Do you have PRI with the
> specified D channels?  You need to make sure that your telco is set up
> to have the D channels on 16 and 47.  When you first start Asterisk,
or
> when you log on to the CLI, do you ever see messages stating the B
> channels are successfully started?
>

sorry, i don't uderstand the question

do you ever see messages stating the B channels are successfully
started?

how can i check this?

thanks
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Re: [Asterisk-Users] Waiting for your help...

[yrving rivas]

Thanks Tzafrir:     I am apologize because of my language problems writing the subject.  I am not good at english.  So thanks for telling me I am doing it the wrong way, and I will be more carefully next time.     Help me if it is possible to you.  The Asterisk version is 2.1 wich I downloaded trhough http://asteriskathome.sourceforge.net/.     To install de fax to email support I followed the instructions in  http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+8     How do I trace and how do I post my configuration files?.  I have some about asterisk programming, I know some in
 programming, and a good learner    Plz.     YrvingTzafrir Cohen <[EMAIL PROTECTED]> escribió:  On Mon, Feb 13, 2006 at 05:25:37AM -0600, yrving rivas wrote:> Hello every one.> > This is a question done by me, not yet answered. Please, help.How about a decent subject for your message?> > I:> > 1. Run install-pdf from linux to support faxes on my asterisk.Of what software package, exactly?What version?What version of Asterisk?What OS/distro? What version of it?> 2. Made the configurations throuhg AMP in> a. Setup->Inbound Routing->(the only route I have)->fax extension->System> b. Setup->Inbound Routing->(the only route I have)->fax
 email->(my email)> c. Setup->Inbound Routing->(the only route I have)->Immediate Answer-> yes> d. Setup->Inbound Routing->(the only route I have)->pause after answer-> 2> e. Setup->General Settings->fax machine for receiving faxes->system> f. Setup->General Settings->Email address to have->(my email)> 3. as a good boy made a test call from a fax, and it reports that couldn´t send the fax ( what means the aste risk couldn´t receive it).> > I didn´t receive any fax. What can I do to receive them?> > Tips:> 1- In my configuration I have a TDM04B.> 2- I receive via email the voice mail messages left to any extension.Looks like a CLI trace would come in handy.> > > In other hand (and not related to this case, as you will see):No, I'm not sure. AMP's dialplan is a mess, and there's no telling whata naive change t
 o it
 will do.> > I made changes to the extensions.conf file through AMP to construct a call forward on no answer, but at the next day all programming was like at beggining. What should I do to make the changes for ever?> amp normally does not override extensions.conf (except, maybe on upgradetime).Anyway, posting your modified extentions.conf may help. Yourextensions_additional.conf may help as well. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | bestICQ# 16849755 | | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
		  
Do You Yahoo!? 
La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx 
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RE: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

[Hunt, Bill]

I would suggest the Pika Technologies Daytona cards. We have been beta-testing 
them with Asterisk for a while now and the results have been very, very good. 
The only drawback right now is that they only support 1.09. 1.2.x support 
should be ready later this quarter. Contact me off list if you would like more 
information. 

Bill Hunt
Stroudwater Contact Point
 
207 347 8080 x219
877 870 1234 Toll Free
 
www.stroudwater.com  
 
"Realize the Value of Customer Contact!"TM

This e-mail is intended solely for the person or entity to which it is 
addressed and may contain confidential or privileged material. Any duplication, 
dissemination, action taken in reliance upon, or other use of this information 
by persons or entities other than the intended recipient is prohibited and may 
violate applicable law. If this e-mail has been received in error, please 
notify the sender and delete the information from your system.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, February 13, 2006 9:15 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

In my expereince a channel bank with a digium single span card and a
Tellabs EC perfomed the best, but is too expensive (it gives you a
minimum of 8 ports). Next to that I use a mediatrix 1204, and compared
to all others I have tried works best. I have tried:
Sipura SPA3000
Digium TDM400 with 4 FXO mods


On 2/13/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have good results with the new TDM2400P serie (with the hardware echocan,
> of course).
> May be you must check one TDM2401E to see if it's ok for you...
>
> Good luck.
>
> Best Regards,
> Francois BERGERET,
> France.
>
>
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de
> [EMAIL PROTECTED]
> Envoyé : lundi 13 février 2006 07:36
> À : asterisk-users@lists.digium.com
> Objet : [Asterisk-Users] Best quad-port fxo solution with EC?
>
>
> Hello All,
>
> I am trying to figure out which way to go for a quad port fxo solution with
> a good echo can on it.  My options are the sangoma remora, a mediatrix fxo,
> or something similar.
>
> The issue is that I would need a good EC.  This would be on about a 9000
> foot loop, and the lines don't function well on a spa-3000 or zaptel tdm 4
> port card.
>
> Anyone have experience that drives them in a certain direction when
> considering a good ec on a quad port?
>
> I tried this also with some fxo clones, but echo killed it.
>
> Thanks,
> Greg
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Re: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

[C F]

In my expereince a channel bank with a digium single span card and a
Tellabs EC perfomed the best, but is too expensive (it gives you a
minimum of 8 ports). Next to that I use a mediatrix 1204, and compared
to all others I have tried works best. I have tried:
Sipura SPA3000
Digium TDM400 with 4 FXO mods


On 2/13/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have good results with the new TDM2400P serie (with the hardware echocan,
> of course).
> May be you must check one TDM2401E to see if it's ok for you...
>
> Good luck.
>
> Best Regards,
> Francois BERGERET,
> France.
>
>
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de
> [EMAIL PROTECTED]
> Envoyé : lundi 13 février 2006 07:36
> À : asterisk-users@lists.digium.com
> Objet : [Asterisk-Users] Best quad-port fxo solution with EC?
>
>
> Hello All,
>
> I am trying to figure out which way to go for a quad port fxo solution with
> a good echo can on it.  My options are the sangoma remora, a mediatrix fxo,
> or something similar.
>
> The issue is that I would need a good EC.  This would be on about a 9000
> foot loop, and the lines don't function well on a spa-3000 or zaptel tdm 4
> port card.
>
> Anyone have experience that drives them in a certain direction when
> considering a good ec on a quad port?
>
> I tried this also with some fxo clones, but echo killed it.
>
> Thanks,
> Greg
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Re: [Asterisk-Users] problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

[nik600]

On 2/13/06, Michael Collins <[EMAIL PROTECTED]> wrote:
> Nik,
>
> Just curious - what is your telco setup?  Do you have PRI with the
> specified D channels?  You need to make sure that your telco is set up
> to have the D channels on 16 and 47.  When you first start Asterisk, or
> when you log on to the CLI, do you ever see messages stating the B
> channels are successfully started?
>

sorry, i don't uderstand the question

do you ever see messages stating the B channels are successfully started?

how can i check this?

thanks
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RE: [Asterisk-Users] Re: asterisk logger - urgent!!!

[Alexander Lopez]

You may not be able to disable the logs but if you do not care about the
information you cal always link to /dev/null

Ie

 ln -s /path/to/log/file /dev/null
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Simone Cittadini
> Sent: Monday, February 13, 2006 5:51 AM
> To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: asterisk logger - urgent!!!
> 
> Dov Bigio ha scritto:
> 
> > I found the problem.
> >  
> > Master.csv reached 2.0GB and since the moment this happened 
> Asterisk 
> > went crazy!
> >  
> > Since I am using cdr-mysql, how do I disable the use of csvs?
> >  
> > Thank you
> > Dov
> 
> Why don't you simply rotate the logs with logrotate ?
> (no, I don't know how to disable them)
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RE: [Asterisk-Users] attended call transfer

[Alex Barnes]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Collins
> Sent: 13 February 2006 00:58
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
> Commercial Discussion
> Subject: RE: [Asterisk-Users] attended call transfer
> 
> Questions for the community: is an "integrated" transfer feature
> valuable to you?  If so, would you be willing to put out a bounty?
(In
> other words, is it just a nice feature or is it so important that
you'd
> be willing to pay a few bucks for it...)  Last question, but possibly
> the most important: what have you done, if anything, to get around the
> split between blind and attended transfers?
> 
> -MC
> 

To get around this issue we have had to only use attended transfer with
Snom phones which are easy to train our users on.

People with DECT / ??? phones have been told to only ever use blind
(only about 10 out of 50 extensions so not the end of the world).


We would definitely chip in some money for this to happen but again it
would have be in the stable release soon as we cant deploy anything
except stable.


Cheers

Alex


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RE: [Asterisk-Users] problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

[Michael Collins]

Nik,

Just curious - what is your telco setup?  Do you have PRI with the
specified D channels?  You need to make sure that your telco is set up
to have the D channels on 16 and 47.  When you first start Asterisk, or
when you log on to the CLI, do you ever see messages stating the B
channels are successfully started?

Let us know.

-MC



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Monday, February 13, 2006 3:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] problem with outgoing calls Unable to
createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)

hi

i've configured a TE205P on asterisk at home

this is my

zaptel.conf

span=1,0,0,ccs,hdb3,crc4,yellow
span=2,0,0,ccs,hdb3,crc4,yellow

bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47

loadzone= it
defaultzone = it

and my zapata.conf
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

faxdetect=both
faxdetect=incoming
faxdetect=outgoing
faxdetect=no
context=default ; Points to the default context of your extensions.conf
channel => 1-15,17-31,32-46,48-62; for E1


i've configured the outgoing calls
[outrt-001-9_outside]
include => outrt-001-9_outside-custom
exten => _9XX,1,Macro(dialout-trunk,1,${EXTEN:1},)
exten => _9XX,2,Macro(outisbusy); No available circuits

if i try to call i get:

Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Setting NAT on RTP to 0
Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response
65111: Match Found
Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Setting NAT on RTP to 0
Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Checking SIP call limits for
device 100
Feb 13 06:19:44 DEBUG[3637] chan_sip.c: build_route: Contact hop:

Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
Macro("SIP/100-611e", "dialout-trunk|1|333|") in new stack
Feb 13 06:19:44 DEBUG[5079] pbx.c: Expression result is '1'
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
GotoIf("SIP/100-611e", "1?3:2)") in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Goto
(macro-dialout-trunk,s,3)
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
Macro("SIP/100-611e", "user-callerid") in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
DBget("SIP/100-611e", "AMPUSER=DEVICE/100/user") in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- DBget: varname=AMPUSER,
family=DEVICE, key=100/user
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- DBget: set variable
AMPUSER to 100
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
DBget("SIP/100-611e", "AMPUSERCIDNAME=AMPUSER/100/cidname") in new
stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- DBget:
varname=AMPUSERCIDNAME, family=AMPUSER, key=100/cidname
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- DBget: set variable
AMPUSERCIDNAME to 100
Feb 13 06:19:44 DEBUG[5079] pbx.c: Expression result is '0'
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
GotoIf("SIP/100-611e", "0?5") in new stack
Feb 13 06:19:44 DEBUG[5079] pbx.c: Not taking any branch
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
SetCallerID("SIP/100-611e", ""100" <100>") in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
NoOp("SIP/100-611e", "Using CallerID "100" <100>") in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
Macro("SIP/100-611e", "record-enable|100|OUT") in new stack
Feb 13 06:19:44 DEBUG[5079] pbx.c: Function result is '0'
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
GotoIf("SIP/100-611e", "0 > 0?2:4") in new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Goto
(macro-record-enable,s,4)
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
AGI("SIP/100-611e", "recordingcheck|20060213-061944|1139829584.46") in
new stack
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
Feb 13 06:19:44 VERBOSE[5079] logger.c:  
recordingcheck|20060213-061944|1139829584.46: Outbound recording not
enabled
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- AGI Script
recordingcheck completed, returning 0
Feb 13 06:19:44 VERBOSE[5079] logger.c: -- Executing
NoOp("SIP/100-611e", "No recording needed") in new stack
Feb 13 06:19:44 VERBOSE[5079] l

RE: [Asterisk-Users] attended call transfer

[Michael Collins]

Yes - in a traditional PBX environment the transferring station has the
ability to pull the call back by pressing a sequence of keys.  In some
PBX's, pressing the transfer key twice, like a double-click of a mouse,
will pull the call back.  In some analog environments, pressing the
flash key twice will pull the call back.  (Pressing the flash key only
once will, in some PBX's, initiate a 3-way call.)

Thanks for bringing this up, Paul.  Any other suggestions about making
transfer even better?

-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Redstone
Sent: Monday, February 13, 2006 5:29 AM
To: Asterisk User
Subject: RE: [Asterisk-Users] attended call transfer

Useful discussion on this. There are some other functions in this which
need to 
be addressed. For example if doing an attended transfer and the
recipient phone 
number goes to voicemail, you have to wait for the timeout to reconnect
to the 
original caller - unless someone know differently. There should be a
reconnect 
hot key.

Again this is comparable to a conventional PABX where the attended
transfer 
puts the caller on hold and pushing a button reconnects.

Paul
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Re: [Asterisk-Users] Voicemail - direct call

[Jesus Mogollon]

http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMailIt's all in thereOn 2/13/06, 
Tomislav Parčina <[EMAIL PROTECTED]> wrote:
Hi list!How to send a call directly to voicemail recording?When I put thisexten => 313,n,VoiceMail,u221Or thisexten => 313,n,VoiceMail,b221In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible?
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