Re: [asterisk-users] IAX configuration

[yonoko molomo]

Hi,
i think that is not the point.
the call works, what is not working is the IAX config.
somehow i need to put "manually" all users of the foreign asterisk
(user, password...).

if i put type=friend, it does not work in any case.
if i put type=peer it works only if i define the users of the foreign
asterisk and also an entry for the foreign server

that should not be the normal behaviour, i guess.

any ideas?
thanks



2007/9/27, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]>:
> when using variables, use ${variablename} instead of $(variablename) --
> (squiggly braces instead of parentheses) -- I'm not sure parentheses are
> allowed.
>
> yonoko molomo wrote:
> > Now I update the extensions.conf file accordingly.
> > exten => clientA_Number,1,Dial(sip/$(exten),10)
>
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Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828

[Olivier]

Strange !
We successfully used SuperMicro boards without any IRQ problems.

What is SuperMicro's reply, concerning this IRQ problems ?
They sure have interest to solve this or at least explain why it can't be
done.

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Re: [asterisk-users] 3-way calling

[Rilawich Ango]

Do u mean meetme?  It is total different from my case.
In meetme, everybody need to know and dial the conference room number
to get into the conference room.  In my case, party A,B,C may not know
the conference number.  A only knows B numbers and B only knows C
numbers.

On 9/28/07, Pamela Weis <[EMAIL PROTECTED]> wrote:
> it is probably not what you are looking for.
> but simply use a conference room of asterisk for those 1 line phones.
>
> pamela

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Re: [asterisk-users] Asterisk - Spandsp Fax not working?

[marco britannio]

On 9/27/07, Jonn R Taylor <[EMAIL PROTECTED]> wrote:
>
> marco britannio wrote:
> > Hi all,
> > I'm trying to setup an asterisk based fax receiving machine.
> > i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9
> > I have no problems with a modem-fax, but with the fax machines i have
> > tried almost every fax fails, both in sending and receive.
> > the machines are sending a receiving a lot of faxes every day and
> > working well, so i think the problem is on the spandsp side.
> > i have tried almost every spandsp version from 0.0.2 to the current one,
> > both with and without ECM, but without luck.
> > has anybody succeeded in receiving faxes with asterisk app_rxfax and
> > spandsp?
> >
> > I'm noticicing a lot of different behaviours: sending w ECM gave me an
> > OK, and the second half of the page was missing, other faxes fail with
> > Sep 26 17:26:18 DEBUG[4741] app_rxfax.c:
> >
> ==
> > Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: Fax receive not successful -
> > result (11) Unexpected message received.
> > Sep 26 17:26:18 DEBUG[4741] app_rxfax.c:
> >
> ==
> >
> >
> > can anybody help me?
> > thank you in advance,
> >
> >
> > marco
> >
> >
> > 
> >
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> Marco,
>
> First off, do not use any version over 0.0.3. I am using 0.0.3 on centos
> 4.5, asterisk 1.2.24 and freepbx 2.3 and it is working very well. One
> very important thing to keep in mind is that faxing over voip will only
> work reliably with ulaw or alaw and your internet connection MUST be
> able to sustain a constant data stream with low jitter. If your
> interested I have a shell script to install asterisk 1.2.24 and
> freepbx-2.3 with rxfax and txfax on centos 4 and working on centos 5.
>
> Jonn
>
> http://jonnt.users.taylortelephone.com/asterisk/centos-asterisk-install.sh
> and hylafax / iaxmodem
> http://jonnt.users.taylortelephone.com/asterisk/
>
>
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hi john,
thank you for your reply.
I've installed the configuration that you suggested (asterisk 1.2.24,
spandsp 0.0.3), but with no luck: I still cannot receive faxes.
Please note that I am not using T38: I am receiving faxes with app_rxfax and
storing them into a dir on the machine.
I have attached an isdn trunk to the card and I am trying to receive the
faxes straight from the trunk.
When receiving faxes without ecm, the sending fax machine completes with an
error, and this is the log I have in asterisk:
 Sep 27 11:54:46 DEBUG[8783] app_rxfax.c: FLOW FAX Set rx type 0
Sep 27 11:54:46 DEBUG[8783] app_rxfax.c: FLOW FAX Set tx type 4
Sep 27 11:54:48 DEBUG[8783] app_rxfax.c: FLOW FAX Set rx type 4
Sep 27 11:54:48 DEBUG[8783] app_rxfax.c: FLOW FAX Set tx type 0
Sep 27 11:54:50 DEBUG[8783] app_rxfax.c: FLOW FAX Set rx type 8
Sep 27 11:54:50 DEBUG[8783] app_rxfax.c: FLOW FAX Set tx type 0
Sep 27 11:54:50 DEBUG[8783] app_rxfax.c: FLOW FAX Switching from V.29 + V.21to
V.29 (-15.04dBm0)
Sep 27 11:54:52 DEBUG[8783] app_rxfax.c: FLOW FAX Set rx type 0
Sep 27 11:54:52 DEBUG[8783] app_rxfax.c: FLOW FAX Set tx type 4
Sep 27 11:54:53 DEBUG[8783] app_rxfax.c: FLOW FAX Set rx type 8
Sep 27 11:54:53 DEBUG[8783] app_rxfax.c: FLOW FAX Set tx type 0
Sep 27 11:54:53 DEBUG[8783] app_rxfax.c: FLOW FAX Switching from V.29 + V.21to
V.29 (-17.87dBm0)
Sep 27 11:55:17 DEBUG[8783] app_rxfax.c: FLOW FAX Set rx type 4
Sep 27 11:55:17 DEBUG[8783] app_rxfax.c: FLOW FAX Set tx type 0
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c:
==
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Pages transferred:  0
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Image size: 1728 x 540
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Image resolution8037 x 3850
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Transfer Rate:  9600
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Bad rows50
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Longest bad row run 10
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Compression type2
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c: Image size (bytes)  116841
Sep 27 11:55:19 DEBUG[8783] app_rxfax.c:
==

Re: [asterisk-users] 3-way calling

[Pamela Weis]

it is probably not what you are looking for.
but simply use a conference room of asterisk for those 1 line phones.

pamela

Rilawich Ango wrote:
> That's easy if phone supports 3 ways call.  However, phones in my
> company only have 1 line without join function.  Is it possible to
> implement 3 ways call using Asterisk without phone support in my case?
>
> On 9/28/07, Anthony Francis <[EMAIL PROTECTED]> wrote:
>   
>> Rilawich Ango wrote:
>> 
>>> What do you mean?  I just want to know whether there is a way to do
>>> the following.
>>>
>>> 1. A --calls --> B
>>> 2. A on hold, B --calls --> C
>>> 3. A, B and C connected to talk
>>>
>>> On 9/28/07, Paul Hales <[EMAIL PROTECTED]> wrote:
>>>
>>>   
 How are you going to do it without a phone?

 PaulH

 
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>>>   
>> your phone would need a "Join" feature or you could do it externally
>> with AMI but that would be clumsy. Most Sip phones have a 3way calling
>> option right on them.
>>
>> Anthony
>>
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>
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>   


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Re: [asterisk-users] call relation in call transfer

[Alex Balashov]

On Fri, 28 Sep 2007, Rilawich Ango wrote:

> In CDR, I found that there are 3 records after doing call transfer. 
> However, 3 of them are individual record that is very difficult to 
> identify they are related to call transfer.  My question is how to 
> identify the call with a clear flow, from CDR or by other means, is a 
> call transfer.

   Do they have a common criterion?  If they do not have a common 
criterion, it is probably not logically possible to associate them.
Asterisk is a back-to-back user agent, so it builds out distinct legs for 
every call with unique Call-IDs and dialogue tags.  This makes it hard to
meaningfully associate call flows like this inherently, unless you do
state tracking in the software to make this possible.

   This has been an ongoing topic of discussion periodically on the 
Asterisk Developers' List (asterisk-dev).  It seems there is considerable 
interest in reworking the CDR engine to account for this type of situation 
more meaningfully.  You may wish to search the list archives for greater 
insight into what core developers are thinking, or to join the list and 
add your two cents to what you want to see from it.  You're definitely not 
the first person to run into this or regard it as a serious impediment. :)

Cheers,

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

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[asterisk-users] call relation in call transfer

[Rilawich Ango]

In CDR, I found that there are 3 records after doing call transfer.
However, 3 of them are individual record that is very difficult to
identify they are related to call transfer.  My question is how to
identify the call with a clear flow, from CDR or by other means, is a
call transfer.

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Re: [asterisk-users] How to "busy out" zap channels

[Tomás Laureano Peralta Tormey]

Anthony:
Yes, you are right, sometimes that could happen but if Brian is going
to take out of service his box is possible that he is monitoring this
box and he could detect this behavior. Also, if you have a hung
channel in your box, this channel is actually in use and will
replicate the requirement that Brian needs (block the incoming calls
from the bearer channels of that PRI trunk).
Anyway, this could be a solution for me (that is better that
connecting the channels to empty meetme rooms) but maybe not for
Brian. I hope this works well for him.

Best regards, Tomás.

2007/9/27, Anthony Francis <[EMAIL PROTECTED]>:
> Tomás Laureano Peralta Tormey wrote:
> > Brian:
> > Maybe the CLI command "stop gracefully" is what are you looking for.
> > Basically, Asterisk will stop receiving incoming calls (of any channel
> > type) and stop itself when all the current calls finish.
> > I hope this help you.
> >
> > Best regards, Tomás.
> >
> > 2007/9/26, Brian Roy <[EMAIL PROTECTED]>:
> >
> >> I know this topic came up many months back and some discussions were
> being
> >> had on how to do this within the Zaptel drivers. However, I'm looking for
> >> even a crude hack that someone has put together to get this done.
> >>
> >> We have PRI's and LD T1's that are load balanced on two boxes. The hunt
> >> order goes from box to box as far as the spans are concerned. There are
> >> times that I would like to busy one out so that calls gradually role to
> the
> >> new box and I can eventually take one out of service. What I was thinking
> is
> >> to create a script that I could tell the specific channels and it would
> go
> >> through and initiate zap calls to an empty meetme. Basically bridging all
> of
> >> the available zap channels on a given span together. Then the trick is
> >> monitoring the hangups so that it can initiate a subsequent call
> immediately
> >> following. Once all of the channels in a span have been bridged, I can
> then
> >> bring the box down. Nasty huh?
> >>
> >> Anyone have a better idea? Or do they have anything like this so I'm not
> >> putting it together?
> >>
> >> Thanks,
> >>
> >> -Brian
> >>
> >>
> >
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> I used to do this automatically at night so that my script could svn
> update configs and then restart asterisk, what a found is that if you
> have one hung channel your ast box wil just sit there till someone
> issues a "Restart Now" in the morning.
>
> Anthony
>
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Re: [asterisk-users] 3-way calling

[Rilawich Ango]

That's easy if phone supports 3 ways call.  However, phones in my
company only have 1 line without join function.  Is it possible to
implement 3 ways call using Asterisk without phone support in my case?

On 9/28/07, Anthony Francis <[EMAIL PROTECTED]> wrote:
> Rilawich Ango wrote:
> > What do you mean?  I just want to know whether there is a way to do
> > the following.
> >
> > 1. A --calls --> B
> > 2. A on hold, B --calls --> C
> > 3. A, B and C connected to talk
> >
> > On 9/28/07, Paul Hales <[EMAIL PROTECTED]> wrote:
> >
> >> How are you going to do it without a phone?
> >>
> >> PaulH
> >>
> >
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> >
> your phone would need a "Join" feature or you could do it externally
> with AMI but that would be clumsy. Most Sip phones have a 3way calling
> option right on them.
>
> Anthony
>
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Re: [asterisk-users] Music On Hold

[Tilghman Lesher]

On Thursday 27 September 2007 17:00:33 Wayne wrote:
> I have noticed that the moh will start from where it left off from the
> previous caller, not from the beginning of the sound file. So going back
> to what Joal asked originally, having one file will mean that - yes
> things will be played in the correct order - but as each caller gets the
> moh - it would 'carry on' at the position it was last at when the
> /previous/ caller left moh - not from the start.

That's true if you use mpg123 for MOH... that's the old way.  The recommended
method now is to use native file format, which is saved per channel.  So every
channel gets the message started from the beginning.

-- 
Tilghman

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Re: [asterisk-users] 3-way calling

[Anthony Francis]

Rilawich Ango wrote:
> What do you mean?  I just want to know whether there is a way to do
> the following.
>
> 1. A --calls --> B
> 2. A on hold, B --calls --> C
> 3. A, B and C connected to talk
>
> On 9/28/07, Paul Hales <[EMAIL PROTECTED]> wrote:
>   
>> How are you going to do it without a phone?
>>
>> PaulH
>> 
>
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your phone would need a "Join" feature or you could do it externally 
with AMI but that would be clumsy. Most Sip phones have a 3way calling 
option right on them.

Anthony

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Re: [asterisk-users] 3-way calling

[Paul Hales]

Your procedure as written below, is perfect and works fine.

I have used Snom, Aastra and Polycom phones at various times to do
exactly as you describe.

PaulH

On Fri, 2007-09-28 at 09:49 +0800, Rilawich Ango wrote:
> What do you mean?  I just want to know whether there is a way to do
> the following.
> 
> 1. A --calls --> B
> 2. A on hold, B --calls --> C
> 3. A, B and C connected to talk
> 
> On 9/28/07, Paul Hales <[EMAIL PROTECTED]> wrote:
> >
> > How are you going to do it without a phone?
> >
> > PaulH
> 
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Re: [asterisk-users] How to "busy out" zap channels

[Anthony Francis]

Tomás Laureano Peralta Tormey wrote:
> Brian:
> Maybe the CLI command "stop gracefully" is what are you looking for.
> Basically, Asterisk will stop receiving incoming calls (of any channel
> type) and stop itself when all the current calls finish.
> I hope this help you.
>
> Best regards, Tomás.
>
> 2007/9/26, Brian Roy <[EMAIL PROTECTED]>:
>   
>> I know this topic came up many months back and some discussions were being
>> had on how to do this within the Zaptel drivers. However, I'm looking for
>> even a crude hack that someone has put together to get this done.
>>
>> We have PRI's and LD T1's that are load balanced on two boxes. The hunt
>> order goes from box to box as far as the spans are concerned. There are
>> times that I would like to busy one out so that calls gradually role to the
>> new box and I can eventually take one out of service. What I was thinking is
>> to create a script that I could tell the specific channels and it would go
>> through and initiate zap calls to an empty meetme. Basically bridging all of
>> the available zap channels on a given span together. Then the trick is
>> monitoring the hangups so that it can initiate a subsequent call immediately
>> following. Once all of the channels in a span have been bridged, I can then
>> bring the box down. Nasty huh?
>>
>> Anyone have a better idea? Or do they have anything like this so I'm not
>> putting it together?
>>
>> Thanks,
>>
>> -Brian
>>
>> 
>
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I used to do this automatically at night so that my script could svn 
update configs and then restart asterisk, what a found is that if you 
have one hung channel your ast box wil just sit there till someone 
issues a "Restart Now" in the morning.

Anthony

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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[Patrick]

On Thu, 2007-09-27 at 14:58 -0500, Erick Perez wrote:
> Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
> handle the 7940G ?
> The 7941G does conform to the standard but it only support SCCP (shame
> on cisco).

The 7941 & 7961 also support SIP if you load the appropriate firmware
from the Cisco website (login required).

Regards,
Patrick




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Re: [asterisk-users] 3-way calling

[Rilawich Ango]

What do you mean?  I just want to know whether there is a way to do
the following.

1. A --calls --> B
2. A on hold, B --calls --> C
3. A, B and C connected to talk

On 9/28/07, Paul Hales <[EMAIL PROTECTED]> wrote:
>
> How are you going to do it without a phone?
>
> PaulH

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[asterisk-users] Asterisk Appliance with VoIPStreet

[Cory Andrews]

Anyone using VoIPStreet with the Asterisk Appliance?  Having some
trouble getting a test trunk working with them, not sure how to properly
refer to them in the "Host" field under Custom VoIP.  

 

Thanks

 

Cory J Andrews

 

 

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Re: [asterisk-users] Asterisk Redundancy

[Douglas Garstang]

>- Original Message 
>From: Scott Moseman <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion 
>
>Sent: Wednesday, September 26, 2007 6:07:06 AM
>Subject: Re: [asterisk-users] Asterisk Redundancy
>
>On 9/26/07, SIP <[EMAIL PROTECTED]> wrote:
>>
>> No. It's not. But there still exists the possibility even in a
>> relatively stable situation that the software could crash or that
>> hardware could fail.  It's best, when planning a highly-available
>> solution, to plan for the unforeseen and not assume you can
>> avoid all mishaps. Let's assume, for the sake of argument, that
>> the software will NEVER fail. Hardware still might, and that would
>> still mean a lost call unless there's a way to switch running calls
>> over to a new server seamlessly.
>>
>
>Also be sure that you have a very redundant network configuration.
>Too often I see people spend a great deal of time and money to get
>redundant servers when their switches, firewalls, routers, etc are not
>even capable of handling a failed network element.

You can achieve this at the application level.







   

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Re: [asterisk-users] Point-to-Point SIP link without registration

[Eric Chamberlain]

Correct you want to set those settings to yes.  Search the Voxilla  
Linksys forums  for hotline or ringdown and  
you will find several examples.  The examples are mostly for the  
spa3000, but the configuration is mostly the same.

You are basically setting up ip or sip uri speeddials that are  
automatically dialed when the line goes off-hook.

--
Eric Chamberlain

On Sep 27, 2007, at 11:14 AM, "Mojo with Horan & Company, LLC" <[EMAIL 
PROTECTED] 
 > wrote:

> err... you'd set them to 'yes', right?  Sorry if I'm missing the  
> obvious.
>
> Eric Chamberlain wrote:
>> You can do this with any of the Linksys SPA series ATA's or phones,  
>> just set "Make Call Without Reg" and "Ans Call Without Reg" to no.
>>
>
>
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Re: [asterisk-users] Polycom 501 won't take new bootrom.ld or sip.ld...

[Anthony Rodgers]

Hi Doug,

What combination of bootrom, sip version and FTP server are you using?
There is a combination with vsFTPd which can cause this sort of problem.

CP 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: Thursday, September 27, 2007 3:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom 501 won't take new bootrom.ld or
sip.ld...

...even when I do the "factory reset" (4-6-8-* then 456).

I tried using FTP and TFTP, but even though the phone uploads the log, I
get these errors:

0927211350|app1 |3|00|Time has been set from
0.us.pool.ntp.org(69.60.124.59).
0927211350|cfg  |4|00|Could not get all 512 bytes of the header.
0927211351|cfg  |4|00|Could not get all 512 bytes of the header.
0927211422|app1 |4|00|Loaded application sip.ld successfully, errors
0x20.
0927211422|app1 |6|00|Uploading boot log, time is THU SEP 27 21:14:22 
0927211422|2007

Has anyone seen this before?


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Re: [asterisk-users] 3-way calling

[Paul Hales]

How are you going to do it without a phone?

PaulH


On Thu, 2007-09-27 at 18:34 +0800, Rilawich Ango wrote:
> From the web site said: 3-way Calling: Normally implemented by the
> phone.  Can I do it in asterisk?  How?
> 
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[asterisk-users] Polycom 501 won't take new bootrom.ld or sip.ld...

[Doug]

...even when I do the "factory reset" (4-6-8-* then 456).

I tried using FTP and TFTP, but even though the phone
uploads the log, I get these errors:

0927211350|app1 |3|00|Time has been set from 0.us.pool.ntp.org(69.60.124.59).
0927211350|cfg  |4|00|Could not get all 512 bytes of the header.
0927211351|cfg  |4|00|Could not get all 512 bytes of the header.
0927211422|app1 |4|00|Loaded application sip.ld successfully, errors 0x20.
0927211422|app1 |6|00|Uploading boot log, time is THU SEP 27 21:14:22 2007

Has anyone seen this before?


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Re: [asterisk-users] Problems Connecting Two Asterisk Installs ViaISDN PRI Cards

[Wai Wu]

Have you tried to load the driver with ec disable? Last time (long time
ago) when I was working on a quad card, we weren't able to get ec to
work with hardware ec on.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Alexander
Sent: Thursday, September 27, 2007 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problems Connecting Two Asterisk Installs
ViaISDN PRI Cards


Okay. I ordered a commercially made T1 crossover cable, connected all of
the cables and rebooted both computers. 

I no longer get the 'Got S-frame while link down' or 'HDLC Bad FCS'
errors. However, I still receive the series of 'Detected alarm on
channel NN: Red Alarm' and 'Unable to disable echo cancellation on
channel NN'. Followed by the alarms clearing. 

At this point I am confident that the cabling is not the problem. I have
included my zaptel and zapata conf files again below. 

Is there a way to make the second TE120P card pass on the timing
received from the first? (rather than using software timing for the
pri_net signalling) 

The errors all seem to be about echo cancellation... What do I need to
do to force asterisk to never disable echo cancellation?

Thanks again for all of the help. Even though we have not found a
solution yet I appreciate the help - and am still confident we will
succeed! 

-Brian



Machine1
=

zaptel.conf
---
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

zapata.conf
---
[trunkgroups]
[channels]
group=1
context=fromtelco
signalling=pri_cpe
switchtype=national
channel=>1-23

group=1
context=frommachine2
signalling=pri_net
switchtype=national
channel=>25-47

Machine2
=

zaptel.conf
---
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

zapata.conf
---
[trunkgroups]
[channels]
group=1
context=frommachine1
signalling=pri_cpe
switchtype=national
channel=>1-23

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Re: [asterisk-users] ChanSpy issue

[Wai Wu]

I got an idea. If you only have 1 sip trunk, just do chanspy(SIP/) 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, September 27, 2007 10:17 AM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] ChanSpy issue

Good point, but the deal is that I have a remote call center with their own 
Nortel PBX.  I get these calls from my DID provided via Zap and I send them 
VoIP to the gateway connected to the Nortel PBX.  This is what I refer to my
SIP trunk.  When I specify Sip/SIPTRUNK(SIPTRUNK) is the name of the
trunk.  Asterisk only monitors one call at a time in the whole trunk, and you 
can Cycle through the calls by pressing "*". 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Wednesday, September 26, 2007 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy issue

I am not an expert on chanspy, but it seems to me spying on the trunk would not 
work very well, would not you hear multiple conversations mixed if more than 
one extension were calling?  Seems best to me to spy on an extension.  YOu also 
can do a show channels to see who is talking to whom.

on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote  > The parameter to 
Chanspy should be the whole or part of the channel name.
I do not understand what you mean by "sip trunk". It make perfect sense that 
you can hear both streams of voice when you use the phone's extension as 
Asterisk usually uses "SIP/extension+xxx" as the channel name of the call.
 >
 >
 > -Original Message-
 > From: [EMAIL PROTECTED] on behalf of Ed Nuñez  > Sent: Wed 9/26/2007 4:48 PM 
 >  > To: [EMAIL PROTECTED]
 > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 > Subject: Re: [asterisk-users] ChanSpy issue  >  >  >  > Hello list  >  >  >  
 > > I am having an issue with Chanspy/SIP that I'm hoping someone has come  > 
 > across and resolved in the past.
 >
 >
 >
 > I am sending calls that come in TDM through T1 ZAP channels and go out to a  
 > > SIP trunk.
 >
 >
 >
 > If I spy on the SIP channel, I can hear the person on the SIP side of the  > 
 > call just fine, but the person on the ZAP channel fades in and out.
 >
 > If I spy on the ZAP channel, and can hear both sides just fine, but I don't  
 > > know who I am spying on since I have other calls coming in on the same T1.
 >
 >
 >
 > If I spy on a SIP extension instead of a SIP trunk, I hear both sides just  
 > > fine.
 >
 >
 >
 > I am using a recent version of Asterisk 1.2 and I am using g729 licenses.
 >
 >
 >
 > This is the command I am using to spy.
 >
 >
 >
 > exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))
 >
 >
 >
 >
 >
 >
 >
 >
 >
 >
 >   >   >   > 
 >   > 
 >   > 
 > RE: [asterisk-users] ChanSpy issue  >   >   > 
 >   >  > The 
 > parameter to Chanspy should be the whole or part of the channel name. I do 
 > not understand what you mean by "sip trunk". It make perfect sense 
 > that you can hear both streams of voice when you use the phone's extension 
 > as Asterisk usually uses "SIP/extension+xxx" as the channel name 
 > of the call.  >   >   > -Original Message-  > From: 
 > [EMAIL PROTECTED] on behalf of Ed Nuñez  > Sent: Wed 9/26/2007 4:48 
 > PM  > To: [EMAIL PROTECTED]
 > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'  > 
 > Subject: Re: [asterisk-users] ChanSpy issue  >   >   >   > 
 > Hello list  >   >   >   > I am having an issue with 
 > Chanspy/SIP that I'm hoping someone has come  > across and resolved in 
 > the past.  >   >   >   > I am sending calls that come in TDM 
 > through T1 ZAP channels and go out to a  > SIP trunk.  >   > 
 >   >   > If I spy on the SIP channel, I can hear the person on the 
 > SIP side of the  > call just fine, but the person on the ZAP channel 
 > fades in and out.  >   > If I spy on the ZAP channel, and can hear 
 > both sides just fine, but I don't  > know who I am spying on since I 
 > have other calls coming in on the same T1.  >   >   >   > If 
 > I spy on a SIP extension instead of a SIP trunk, I hear both sides just  
 > > fine.  >   >   >   > I am using a recent version of 
 > Asterisk 1.2 and I am using g729 licenses.  >   >   >   > 
 > This is the command I am using to spy.  >   >   >   > exten 
 > => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 >
 > 
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--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

Re: [asterisk-users] Music On Hold

[Wayne]

Hiya all,
Please excuse me if I'm a bit out of date with my Asterisk version here 
but... :)

I have noticed that the moh will start from where it left off from the 
previous caller, not from the beginning of the sound file. So going back 
to what Joal asked originally, having one file will mean that - yes 
things will be played in the correct order - but as each caller gets the 
moh - it would 'carry on' at the position it was last at when the 
/previous/ caller left moh - not from the start.


Cheery
Wayne.



David Gomillion wrote:
> > > Hi All,
> > >
> > > I need to have the same file played from MoH every time someone gets
> > > to
> > > MoH from a Dial. I want to play marketing messages from it and I
> > > want it
> > > to start from file 1 every time.
> > >
> > > Anyone know if/how this can be done?
>
> On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote:
> > Make the file the only one in the /var/lib/asterisk/moh directory.
> >
> > Forrest Beck
> > [EMAIL PROTECTED] 
> > www.shift8.biz 
> Thanks for the suggestion, but I need it to play multiple messages.
> Always starting with the same one.
>
> Cheers,
>
> Joel.
>
>
> Create a new MOH class with one large file consisting of every message 
> you want heard, in the order you want them heard. Since there will be 
> only one file, you know which will be first ;)
>
> We actually do this with some of our queues, so I know it works.
>
> 
>
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Re: [asterisk-users] What's the deal with ATAcomm?

[Jerry Jones]

I will miss them. It was nice having a local company with a few  
Polycoms in stock most of the time. A month or so ago we needed some  
quick and were unable to contact them, either through their toll free  
or local numbers. I swung by their office last week and nocticed it  
was vacant.


On Sep 27, 2007, at 1:49 PM, Darrick Hartman (lists) wrote:

> Doug wrote:
>> http://www.atacomm.com/
>>
>> ATACOMM
>>
>> Dear Atacomm Customers,
>> We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
>> and its parent company Ataractic Corporation has ceased
>> operations.  We appreciate the 7 years of loyalty and support from
>> our customers.  We sincerely regret any adverse effects this may  
>> have caused.
>>
>
> I'd say that's pretty self-explanatory.  My credit card company is
> trying to recover about $800 in fraudulent charges for duplicate
> transactions and failing to send the merchandise for a transaction  
> that
> dates back to late August.
>
> Normally I'd say this sort of thing belongs only on the biz list, but
> this sort of issue may affect so many people it's worth noting here  
> (but
> not dragging out with hundreds of "me toos").
> -- 
> Darrick Hartman
> DJH Solutions, LLC
> http://www.djhsolutions.com
>
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Re: [asterisk-users] Zap channel stuck in conference

[Tzafrir Cohen]

On Thu, Sep 27, 2007 at 03:07:54PM -0400, Jason Martin wrote:
> Hello, I have a strange problem with one of my Zap channels. A user told me 
> that he was in a voicemail system during a call, hit the Flash button, and 
> the call hung up. Now we get no dialtone on the phone hooked up to the 
> channel. Here's the status of the channel:

What is the output of:

  show channels

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

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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[Sasa]

Hi, sorry for my intrusion... I have the same problem with Cisco 7941G, can 
I do buy the the Smartnet registration also for 7941G or this license is 
available only for 7940G ?
Thanks.

--
   Salvatore.

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

Sent: Thursday, September 27, 2007 6:48 PM
Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk


> You need to purchase a Smartnet license for your phone, and have it 
> registered by a Cisco authorized reseller.
>
> The Smartnet registration will run you $10-$20 per phone, depending upon 
> the reseller.  The registration process typically takes around 24-48 hours 
> to process.
>
> Once registered, you will receive an email from Cisco with instruction on 
> obtaining a Cisco TAC login.  Once you have your login, you will be able 
> to access and download the SIP firmware.
>
> If you look around on Google or on the Cisco website, there is a lot of 
> documentation out there that describes the process for migrating the 
> firmware.
>
> I agree, it is a lot of work.  I do not see Cisco shipping phones with SIP 
> firmware on them anytime soon, as obviously their vested interest is in 
> their CCM and CCME platforms, and their native Skinny protocol.
>
> They are being dragged reluctantly into SIP and platforms such as Asterisk 
> present a threatthey are not going to tailor their tools and channel 
> practices toward folks using a non Cisco platform.
>
> Cory J Andrews
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of yonoko 
> molomo
> Sent: Thursday, September 27, 2007 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk
>
> Hi,
> i bought this device and the cost of the 7040G itself was similar to
> the license. if im not wrong, the telephone cost around 80€. the sip
> license was around 80€ as well
>
> however, i am quite annoyed because the phone did not come with sip,
> but callmanager so i cant use it as i planned.
> i have read somewhere that I need to change the firmware, but i
> require a cisco account to download the firmware (but nobody provided
> me this account). we paid for the SIP license, but we did not get a
> SIP-capable device, and we do not have the way to download the
> firmware (yet).
>
> Regarding the power adapter, I had to buy them sepparately. since i do
> not have POE devices i cant answer your last question.
>
>
>
> 2007/9/27, Erick Perez <[EMAIL PROTECTED]>:
>> Hi there,
>> In Cisco web site
>> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
>> It says that regardless of the technology used you have to buy a 
>> licencse.
>> Does the license apply to use the phone with asterisk, or, can i just
>> buy the phone?
>>
>> Also, the phone does not requiere to use an AC adapter if used with
>> PoE injectors/switches.
>> Can non-Cisco PoE injectors/switches be used with this phone?
>>
>> Thanks,
>>
>> --
>> 
>> Erick Perez
>> 
>>
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>
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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[Peder @ NetworkOblivion]

I'm pretty sure that any Cisco switch that has PoE supports pre-standard 
PoE.  However there are only certain ones that do support the standard. 
  If you are looking for the cheapest used ones, then a 3524-PWR will 
work.  If you want new, then a 3560 "ps" version will work.

Erick Perez wrote:
> Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
> handle the 7940G ?
> The 7941G does conform to the standard but it only support SCCP (shame
> on cisco).
> 
> 
> 
> On 9/27/07, Peder @ NetworkOblivion <[EMAIL PROTECTED]> wrote:
>> Yes, you need to buy a license if you use it with ANY pbx, whether it is
>> Callmangler or Asterisk or whatever.  If you buy one used, then you need
>> to pay to re-license it as well.
>>
>> The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you
>> will need a switch that provides Cisco PoE for it to work.
>>
>>
>> Erick Perez wrote:
>>> Hi there,
>>> In Cisco web site
>>> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
>>> It says that regardless of the technology used you have to buy a licencse.
>>> Does the license apply to use the phone with asterisk, or, can i just
>>> buy the phone?
>>>
>>> Also, the phone does not requiere to use an AC adapter if used with
>>> PoE injectors/switches.
>>> Can non-Cisco PoE injectors/switches be used with this phone?
>>>
>>> Thanks,
>>>
>>
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> 
> 


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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[Erick Perez]

Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
handle the 7940G ?
The 7941G does conform to the standard but it only support SCCP (shame
on cisco).



On 9/27/07, Peder @ NetworkOblivion <[EMAIL PROTECTED]> wrote:
> Yes, you need to buy a license if you use it with ANY pbx, whether it is
> Callmangler or Asterisk or whatever.  If you buy one used, then you need
> to pay to re-license it as well.
>
> The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you
> will need a switch that provides Cisco PoE for it to work.
>
>
> Erick Perez wrote:
> > Hi there,
> > In Cisco web site
> > http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
> > It says that regardless of the technology used you have to buy a licencse.
> > Does the license apply to use the phone with asterisk, or, can i just
> > buy the phone?
> >
> > Also, the phone does not requiere to use an AC adapter if used with
> > PoE injectors/switches.
> > Can non-Cisco PoE injectors/switches be used with this phone?
> >
> > Thanks,
> >
>
>
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-- 

Erick Perez

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[asterisk-users] Timeout issues

[Kutman.DK]

Hello, 

I have a softphone which I am using with Asterisk. Sometimes when I place a 
call it works fine and sometimes the SipListener comes back with a timeout.  
The timeout is a Retransmission timeout and it seems to be occurring when the 
INVITE is sent.  The thing is about 70% of the time it works, but the other 30% 
or so it comes back with the timeout message.  Is there any reason why it does 
this occasionally?  I am not sure what to look for or how to get rid of these 
timeouts. 

Any help or advice would be appreciated. 

Thanks, 

Denis 


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[asterisk-users] Zap channel stuck in conference

[Jason Martin]

Hello, I have a strange problem with one of my Zap channels. A user told me 
that he was in a voicemail system during a call, hit the Flash button, and 
the call hung up. Now we get no dialtone on the phone hooked up to the 
channel. Here's the status of the channel:

[EMAIL PROTECTED]:~$ sudo asterisk -r -x "zap show channel 7"
Parsing /etc/asterisk/extconfig.conf
Channel: 7
File Descriptor: 27
Span: 1
Extension:
Dialing: no
Context: batoutbound
Caller ID: 5852311542
Calling TON: 0
Caller ID name: Metrix Matrix, Inc.
Destroy: 0
InAlarm: 0
Signalling Type: FXO Kewlstart
Radio: 0
Owner: Zap/7-1
Real: Zap/7-1 (Confed)
Callwait: 
Threeway: Zap/7-1 (Confed)
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: yes
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently ON
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Onhook
[Sep 27 15:02:38] -- Remote UNIX connection
Asterisk ending (0).

There's no channel for Zap/7-1 so soft hangup isn't an option. I can't do a 
full zap restart because the system is in use. Any thoughts?

Thanks!

-- 
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679


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Re: [asterisk-users] Which Asterisk version to use?

[Razza]

On 27/09/2007, Eric B. <[EMAIL PROTECTED]> wrote:
>
> For starters, what is the difference btwn the 1.2 and 1.4 branches of
> Asterisk?  I can't seem to find a document that describes the changes.
>
Anyone?
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Re: [asterisk-users] What's the deal with ATAcomm?

[Darrick Hartman (lists)]

Doug wrote:
> http://www.atacomm.com/
> 
> ATACOMM
> 
> Dear Atacomm Customers,
> We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm 
> and its parent company Ataractic Corporation has ceased 
> operations.  We appreciate the 7 years of loyalty and support from 
> our customers.  We sincerely regret any adverse effects this may have caused.
> 

I'd say that's pretty self-explanatory.  My credit card company is 
trying to recover about $800 in fraudulent charges for duplicate 
transactions and failing to send the merchandise for a transaction that 
dates back to late August.

Normally I'd say this sort of thing belongs only on the biz list, but 
this sort of issue may affect so many people it's worth noting here (but 
not dragging out with hundreds of "me toos").
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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Re: [asterisk-users] What's the deal with ATAcomm?

[Erik Anderson]

On 9/27/07, Doug <[EMAIL PROTECTED]> wrote:
> http://www.atacomm.com/

Heh - yah I pulled up their website earlier today with the hopes of
purchasing a Polycom SIP conference phone.  Oh well...

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Re: [asterisk-users] Which Asterisk version to use?

[Joseph]

Use whatever stable version your distro, in your case Debian" provides, it is 
your best options; especially when starting.
As to GUI - it is not a good option, you will not learn much, in addition if 
your GUI will not work and you need to fix something 
you are stuck.
Go the way everybody does, use command line and your editor; it is not that 
hard; there are only few files you have to edit.
It is your best chance to learn something.

#Joseph

On 09/27/07 13:01, Jim Canfield wrote:
>Eric B. wrote:
> 
>  site and got to chapter 4 or 5 and decided to take a break.  Which is when I
>  found AsteriskNow and TriBox and then started wondering if it was really
>  necessary / worthwhile to figure out all the intricacies of the application
>  if someones have already created the appliance version of it.  In which
>  case, I was very confused as to the difference btwn AsteriskNow and TriBox.
> 
>  Thanks!
[snip]

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Re: [asterisk-users] Point-to-Point SIP link without registration

[Mojo with Horan & Company, LLC]

err... you'd set them to 'yes', right?  Sorry if I'm missing the obvious.

Eric Chamberlain wrote:
> You can do this with any of the Linksys SPA series ATA's or phones, just set 
> "Make Call Without Reg" and "Ans Call Without Reg" to no.
>   


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[asterisk-users] What's the deal with ATAcomm?

[Doug]

http://www.atacomm.com/

ATACOMM

Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm 
and its parent company Ataractic Corporation has ceased 
operations.  We appreciate the 7 years of loyalty and support from 
our customers.  We sincerely regret any adverse effects this may have caused.



Sincerely,

Atacomm Sales & Support Staff


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Re: [asterisk-users] Which Asterisk version to use?

[Jim Canfield]

   Eric B. wrote:

 site and got to chapter 4 or 5 and decided to take a break.  Which is when I
 found AsteriskNow and TriBox and then started wondering if it was really
 necessary / worthwhile to figure out all the intricacies of the application
 if someones have already created the appliance version of it.  In which
 case, I was very confused as to the difference btwn AsteriskNow and TriBox.

 Thanks!
  

   Last week I posed a similar question to the list as a "noob". 
   Specifically, I was curious why every one was so adverse to GUI
   implementations.   Like you, I entered the asterisk world quite idealistic
   and oblivious to what is actually required to create a functional system
   (still am).  I spent the good part of last week trying to make heads or
   tails of the AsteriskNOW distro, but finally gave up in favor of a plain
   jane Debian install with asterisk and wish I would have never wasted so
   much time trying to figure out how the users.conf  worked.

   
   [TK]D-Fender - The users.conf is a flaming piece of sh**!
   <\quote>

   I actually thought that was a bit harsh when I read it...turns out to be
   quite accurate.  Long story short, I'm learning to be quite comfortable in
   the CLI and finding myself more productive in nano (yes..nano) than I was
   in the GUI.

   Good luck!

   -jc
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[asterisk-users] SIP interface status

[James Fromm]

I've discovered that the status of a SIP device doesn't get passed as 
in-use when on an outbound call.  Viewing the debug log the status is 
always passed as 'not in use' when on the outbound call.  The 
sip_devicestate function doesn't appear to check the user object at all.

The devices are configured as friends in sip.conf.  Being both a peer 
and a user, the device is found as a peer in the sip_devicestate 
function but then not found in use because only the peer object is 
checked.  If the device is configured as a user in sip.conf, then the 
status is returned as INVALID because in the sip_devicestate function it 
doesn't find a peer to check.

Looking at the SVN repository, the function appears to have never 
checked the user object.  Shouldn't the device be defined as in-use even 
when on an outbound call?  Does this function need to be rewritten? 
Anyone have a solution?

Thanks,
James


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Re: [asterisk-users] Digium acquires Switchvox

[Michael Collins]

>  I also think this is a
> positive thing for the Asterisk community as well, as key pieces of
the
> Switchvox system will be rolled into the open-source version of
Asterisk.
> (I've personally heard of two or three things that the Switchvox team
has
> done to improve Asterisk, and I'm sure there are lots more I'm not
aware
> of yet.)
> 
Thanks for the update.  Many of us are curious as to what those features
are and when they might be made available.  We are looking forward to
hearing more... 

-MC

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Re: [asterisk-users] Which Asterisk version to use?

[Michael Collins]

> I'm a complete newbie to Asterisk and have been reading through
> documentation and sites for the last couple of hours trying to
understand
> what to do to start my learning curve with Asterisk, and am very
confused.

It's a big world, so take a deep breath and don't worry about being
overwhelmed at first.

> 
> For starters, what is the difference btwn the 1.2 and 1.4 branches of
> Asterisk?  I can't seem to find a document that describes the changes.
> 
> Secondly, what is the best way to start off with Asterisk?  Should I
> install
> a Linux distro from scratch and then install Asterisk on top of that,
> start
> with AsteriskNOW and go from there, or start with Tribox?  What
advantage
> do
> I get installing Linux/Asterisk vs. installing AsteriskNow or Tribox
and
> starting my learning curve from there?  It would seem as the most
> reasonable
> to start with a prepackaged "appliance" installation - no?

One advantage to using the prepackage method is that you get straight
into what Asterisk can do while skipping most of the how-do-I-set-it-up
drama.  I personally like the Trixbox distro for getting a quick setup
into operation.  It is relatively easy to get started with and let's you
play with the system.  Once you get your feet wet then it's a good
exercise to learn the steps of doing a manual install.


> 
> Can someone please explain the difference between AsteriskNow and
Tribox?
> They seem to be filling the same need - a one-step easy installation
of
> Asterisk on a brand new PC.  Am I missing something?  Both have GUIs,
but
> TriBox seems to be more complete with more features.  Is this not
correct?
> 

Trixbox is, essentially, Asterisk + Asterisk 3rd party add-ons + decent
preconfiguration.  I'm not sure about AsteriskNOW.  Just remember that
Asterisk has an ecosystem, so there are lots of different things you can
plug into it and lots of different apps that can interface with it.
Just take it slow and steady and you'll do very well.


> Thanks so much for any information to help set me on the right path.
As
> you
> can see, I am extermely confused and lost in the maze of Asterisk docs
and
> struggling to find a little headway here.
> 

Someone already mentioned it, but get the O'Reilly book - "Asterisk -
The Future Of Telephony."  (Abbreviated TFOT in many places.)  Be sure
to get the 2nd edition if you're going to buy it off the shelf.  It
should be out any time if it isn't already available.

-MC

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Re: [asterisk-users] Asterisk Redundancy

[Douglas Garstang]

>- Original Message 
>From: SIP <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]>
>Sent: Wednesday, September 26, 2007 4:31:08 AM
>Subject: Re: [asterisk-users] Asterisk Redundancy
>
>Per Jessen wrote:
>> Atis Lezdins wrote:
>>
>>   
>>> This seems nice way of sharing settings, however it wouldn't take over
>>> calls in progress. For us, currently the greatest problem is that
>>> whenever Asterisk crashes, calls are lost, and that means - lost
>>> money. Are there any ideas?
>>> 
>>
>> Perhaps investigate/diagnose the craches?  Software instability is not
>> solved with a high-availability solution. IMHO.  
>>
>>
>> /Per Jessen, Zürich
>>
>>   
>No. It's not. But there still exists the possibility even in a 
>relatively stable situation that the software could crash or that 
>hardware could fail.  It's best, when planning a highly-available 
>solution, to plan for the unforeseen and not assume you can avoid all 
>mishaps. Let's assume, for the sake of argument, that the software will 
>NEVER fail. Hardware still might, and that would still mean a lost call 
>unless there's a way to switch running calls over to a new server 
>seamlessly.
>
>Are there such ways? IP calls are especially troublesome in that regard.

Don't set your goals too high. I've worked for a few companies with Asterisk 
now and just having an architecture that can recover within a few seconds and 
process new calls almost seamlessly is a workable goal. Having an architecture 
that can seamlessly fail over and keep calls up is kinda like the whole grail 
of redundancy with Asterisk. Hint... you might be able to do it with SIP 
reinvites...

Doug.







   

Yahoo! oneSearch: Finally, mobile search 
that gives answers, not web links. 
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC___

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Re: [asterisk-users] Digium acquires Switchvox

[Jared Smith]

On Thu, 2007-09-27 at 12:21 -0400, Aubrey Wells wrote:
> I take it you mean to insert this: http://www.digium.com/en/company/ 
> switchvox-acquisition-faq.php URL there? :-)

Yes, that was a mistake on my part.  I shouldn't be allowed to post
before breakfast.  The URL I meant to insert is:
http://www.digium.com/en/company/switchvox-acquisition-faq.php


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] IAX configuration

[Mojo with Horan & Company, LLC]

when using variables, use ${variablename} instead of $(variablename) -- 
(squiggly braces instead of parentheses) -- I'm not sure parentheses are 
allowed.

yonoko molomo wrote:
> Now I update the extensions.conf file accordingly.
> exten => clientA_Number,1,Dial(sip/$(exten),10)

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Re: [asterisk-users] Which Asterisk version to use?

[Eric B.]

"Bob Pierce" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote:
>> I'm a complete newbie to Asterisk and have been reading through
>> documentation and sites for the last couple of hours trying to
>> understand what to do to start my learning curve with Asterisk, and am
>> very confused.
>
> The best starting point IMHO is simply to buy the new O'Reilly Book
> "Asterisk - The Future of Telephony" and follow the instructions there
> to install and configure Asterisk 1.4 on top of your favourite Linux
> Distro.

Oh - I've already started that as well; I downloaded the PDF from the doc 
site and got to chapter 4 or 5 and decided to take a break.  Which is when I 
found AsteriskNow and TriBox and then started wondering if it was really 
necessary / worthwhile to figure out all the intricacies of the application 
if someones have already created the appliance version of it.  In which 
case, I was very confused as to the difference btwn AsteriskNow and TriBox.

Thanks!

Eric




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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[Cory Andrews]

You need to purchase a Smartnet license for your phone, and have it registered 
by a Cisco authorized reseller.  

The Smartnet registration will run you $10-$20 per phone, depending upon the 
reseller.  The registration process typically takes around 24-48 hours to 
process.

Once registered, you will receive an email from Cisco with instruction on 
obtaining a Cisco TAC login.  Once you have your login, you will be able to 
access and download the SIP firmware.

If you look around on Google or on the Cisco website, there is a lot of 
documentation out there that describes the process for migrating the firmware.

I agree, it is a lot of work.  I do not see Cisco shipping phones with SIP 
firmware on them anytime soon, as obviously their vested interest is in their 
CCM and CCME platforms, and their native Skinny protocol.

They are being dragged reluctantly into SIP and platforms such as Asterisk 
present a threatthey are not going to tailor their tools and channel 
practices toward folks using a non Cisco platform.

Cory J Andrews




-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yonoko molomo
Sent: Thursday, September 27, 2007 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk

Hi,
i bought this device and the cost of the 7040G itself was similar to
the license. if im not wrong, the telephone cost around 80€. the sip
license was around 80€ as well

however, i am quite annoyed because the phone did not come with sip,
but callmanager so i cant use it as i planned.
i have read somewhere that I need to change the firmware, but i
require a cisco account to download the firmware (but nobody provided
me this account). we paid for the SIP license, but we did not get a
SIP-capable device, and we do not have the way to download the
firmware (yet).

Regarding the power adapter, I had to buy them sepparately. since i do
not have POE devices i cant answer your last question.



2007/9/27, Erick Perez <[EMAIL PROTECTED]>:
> Hi there,
> In Cisco web site
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
> It says that regardless of the technology used you have to buy a licencse.
> Does the license apply to use the phone with asterisk, or, can i just
> buy the phone?
>
> Also, the phone does not requiere to use an AC adapter if used with
> PoE injectors/switches.
> Can non-Cisco PoE injectors/switches be used with this phone?
>
> Thanks,
>
> --
> 
> Erick Perez
> 
>
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[asterisk-users] TE120p and music on hold

[[EMAIL PROTECTED]]

We buyed a te120p. We are using asterisk 1.2.24 on linux 2.6.18 ( config.gz is 
the configuration of the kernel). Our customer tells they ramdomly are put on 
hold, earing mosiconhold, both legs of the call. So, when this happens, 
neither can resume from musiconhold and the only thing to do is to end the 
call.
Also, time to time, calls are ended ramdomly, without any reason, at first 
glance.

Zaptel.conf is
---
loadzone=it
deafultzone=it

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

they have only ten lines on pri

Zapata.conf is



[trunkgroups]

[channels]
language=it
immediate=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
callerid=asreceived
switchtype=euroisdn

context=from-mypstn-custom
;context=from-zapata
signalling=pri_cpe

overlapdial=yes
;pridialplan=national
pridialplan=unknown
prilocaldialplan=unknown


internationalprefix=00
nationalprefix=0
localprefix=02
privateprefix=*** ; for privacy j naked this value. 
;If You need it j can unake

priindication=inband

;immediate=no

usecallerid=yes
hidecallerid=yes
;usecallingpres=yes

group=1
channel=>1-15,17-31
-

J need help as soon as it is possible. 

Salvatore Cardinale
OpenSys srl
Via P. Del Vaga 12
Milano (ITALY)
tel +39 0233405257
fax +39 33400655



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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[yonoko molomo]

Hi,
i bought this device and the cost of the 7040G itself was similar to
the license. if im not wrong, the telephone cost around 80€. the sip
license was around 80€ as well

however, i am quite annoyed because the phone did not come with sip,
but callmanager so i cant use it as i planned.
i have read somewhere that I need to change the firmware, but i
require a cisco account to download the firmware (but nobody provided
me this account). we paid for the SIP license, but we did not get a
SIP-capable device, and we do not have the way to download the
firmware (yet).

Regarding the power adapter, I had to buy them sepparately. since i do
not have POE devices i cant answer your last question.



2007/9/27, Erick Perez <[EMAIL PROTECTED]>:
> Hi there,
> In Cisco web site
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
> It says that regardless of the technology used you have to buy a licencse.
> Does the license apply to use the phone with asterisk, or, can i just
> buy the phone?
>
> Also, the phone does not requiere to use an AC adapter if used with
> PoE injectors/switches.
> Can non-Cisco PoE injectors/switches be used with this phone?
>
> Thanks,
>
> --
> 
> Erick Perez
> 
>
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Re: [asterisk-users] Cisco 7940G licensing with asterisk

[Peder @ NetworkOblivion]

Yes, you need to buy a license if you use it with ANY pbx, whether it is 
Callmangler or Asterisk or whatever.  If you buy one used, then you need 
to pay to re-license it as well.

The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you 
will need a switch that provides Cisco PoE for it to work.


Erick Perez wrote:
> Hi there,
> In Cisco web site
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
> It says that regardless of the technology used you have to buy a licencse.
> Does the license apply to use the phone with asterisk, or, can i just
> buy the phone?
> 
> Also, the phone does not requiere to use an AC adapter if used with
> PoE injectors/switches.
> Can non-Cisco PoE injectors/switches be used with this phone?
> 
> Thanks,
> 


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Re: [asterisk-users] IAX configuration

[yonoko molomo]

hi,
it does not help.
at first i already tried using type=friend.
but i am not able to make calls.
in the 'caller' asterisk get:

WARNING[18541]: chan_iax2.c:7101 socket_process: Call rejected by
x.x.x.x: No authority found
-- Hungup 'IAX2'

in the 'called' asterisk i get following error:
chan_iax2.c:7584 socket_process: Host x.x.x.x failed to authenticate as serverB

but in the CLI, i type iax2 show registry and i see that servers are registered.
Host  dnsmgr  UsernamePerceived Refresh  State
x.x.x.x:4569   N   servery.y.y.y:4569 60  Registered
i asume registered means also authenticated but it complaints.


I already followed that link but it does not work. i need to change some things.
first, i need to change friend to peer. if i use type=friend it does
not work for me.

then i need to define all users of the foreign asterisk, which is not
normal, i guess.

I also need to define the ip address of the servers as host=dynamic.
if i put host=192.168.1.10 or similar, i get the message in the CLI:
"Peer 'server' is not dynamic (from 192.168.1.10)"

somehow i feel that there is "something" missing in the configuracion
examples, but i do not know what is it.
any ideas?

2007/9/27, Gordon Henderson <[EMAIL PROTECTED]>:
> On Thu, 27 Sep 2007, Anthony Messina wrote:
>
> > On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
> >> Hi,
> >>
> >> I have some problems and doubts connecting two asterisk servers.
> >>
> >> I have one asterisk (serverA), with 1 sip client registered (clientA).
> >> I have another asterisk (sever B), with another client (clientB).
> >> Now I want to call from client A to B and from B to A.
> >>
> >> Searching in google i find many configuration examples. For instance:
> >> http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_us
> >> ing_IAX There are hundreds of pages like this. Also the book "Asterisk the
> >> future of telephony" describes how to configure IAX.
> >> But it seems that there are many things missing.
> >>
> >> I understand I need to configure iax.conf in server A and add an entry
> >> for server B as "peer".
> >> I also need to include a "register" line in iax.conf of server B. If i
> >> do this, i have server B registered in server A.
> >>
> >> I do the same in server B.
> >>
> >
> > you'll need to define each as a "friend" of each other.
>
> Start here:
>
>http://astrecipes.net/index.php?n=204
>
> Gordon
>
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Re: [asterisk-users] Which Asterisk version to use?

[Bob Pierce]

On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote:
> I'm a complete newbie to Asterisk and have been reading through 
> documentation and sites for the last couple of hours trying to
> understand what to do to start my learning curve with Asterisk, and am
> very confused.

The best starting point IMHO is simply to buy the new O'Reilly Book
"Asterisk - The Future of Telephony" and follow the instructions there
to install and configure Asterisk 1.4 on top of your favourite Linux
Distro.

Bob

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Re: [asterisk-users] Digium acquires Switchvox

[Aubrey Wells]

On Sep 27, 2007, at 11:25 AM, Jared Smith wrote:

> putting up a question and answer page at .  Obviously


I take it you mean to insert this: http://www.digium.com/en/company/ 
switchvox-acquisition-faq.php URL there? :-)



--
Aubrey Wells
Senior Engineer
Shelton | Johns Technology Group
404.478.2790
www.sheltonjohns.com






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[asterisk-users] Which Asterisk version to use?

[Eric B.]

Hi,

I'm a complete newbie to Asterisk and have been reading through 
documentation and sites for the last couple of hours trying to understand 
what to do to start my learning curve with Asterisk, and am very confused.

For starters, what is the difference btwn the 1.2 and 1.4 branches of 
Asterisk?  I can't seem to find a document that describes the changes.

Secondly, what is the best way to start off with Asterisk?  Should I install 
a Linux distro from scratch and then install Asterisk on top of that, start 
with AsteriskNOW and go from there, or start with Tribox?  What advantage do 
I get installing Linux/Asterisk vs. installing AsteriskNow or Tribox and 
starting my learning curve from there?  It would seem as the most reasonable 
to start with a prepackaged "appliance" installation - no?

Can someone please explain the difference between AsteriskNow and Tribox? 
They seem to be filling the same need - a one-step easy installation of 
Asterisk on a brand new PC.  Am I missing something?  Both have GUIs, but 
TriBox seems to be more complete with more features.  Is this not correct?

Thanks so much for any information to help set me on the right path.  As you 
can see, I am extermely confused and lost in the maze of Asterisk docs and 
struggling to find a little headway here.

Eric




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[asterisk-users] Cisco 7940G licensing with asterisk

[Erick Perez]

Hi there,
In Cisco web site
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
It says that regardless of the technology used you have to buy a licencse.
Does the license apply to use the phone with asterisk, or, can i just
buy the phone?

Also, the phone does not requiere to use an AC adapter if used with
PoE injectors/switches.
Can non-Cisco PoE injectors/switches be used with this phone?

Thanks,

-- 

Erick Perez


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Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards

[Richard Lyman]

Brian Alexander wrote:
> *snipped
> The errors all seem to be about echo cancellation... What do I need to do to
> force asterisk to never disable echo cancellation?
>   
*snipped

there used to be this in ../zaptel/zconfig.h

#define NO_ECHOCAN_DISABLE

check if whatever version you are running has this.


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[asterisk-users] Digium acquires Switchvox

[Jared Smith]

As you may have heard, Digium announced this morning that it's acquired 
Switchvox, a well known provider of Asterisk-based phone systems.  Since 
several people have already asked me about the deal, I figured I'd let you all 
know my feelings on the matter.  First of all, let me say that I personally 
think this is a great thing for all the parties involved.  Obviously this gives 
Digium a more polished product aimed primarily at small- and medium-sized 
businesses, and it allows the Switchvox team to join a larger organization with 
a greater reach.  I also think this is a positive thing for the Asterisk 
community as well, as key pieces of the Switchvox system will be rolled into 
the open-source version of Asterisk.  (I've personally heard of two or three 
things that the Switchvox team has done to improve Asterisk, and I'm sure there 
are lots more I'm not aware of yet.)

The full text of the press release can be found at 
http://www.digium.com/en/mediacenter/news/viewpress.php?id=digium-acquires-switchvox,
 and we've tried to answer the common questions we've already received by 
putting up a question and answer page at .  Obviously this 
won't answer everyone's questions, so I'm more than happy to field questions 
(either in private or here on the mailing list) and have them answered by 
Digium management.  We're also planning an audio conference next Tuesday in 
which you'll be able to dial in and ask any questions you may have concerning 
the acquisition.  (I'll post the exact time and details as soon as I have them.)


---
Jared Smith
Community Relations Manager
Digium, Inc.

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Re: [asterisk-users] IAX configuration

[Gordon Henderson]

On Thu, 27 Sep 2007, Anthony Messina wrote:

> On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
>> Hi,
>>
>> I have some problems and doubts connecting two asterisk servers.
>>
>> I have one asterisk (serverA), with 1 sip client registered (clientA).
>> I have another asterisk (sever B), with another client (clientB).
>> Now I want to call from client A to B and from B to A.
>>
>> Searching in google i find many configuration examples. For instance:
>> http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_us
>> ing_IAX There are hundreds of pages like this. Also the book "Asterisk the
>> future of telephony" describes how to configure IAX.
>> But it seems that there are many things missing.
>>
>> I understand I need to configure iax.conf in server A and add an entry
>> for server B as "peer".
>> I also need to include a "register" line in iax.conf of server B. If i
>> do this, i have server B registered in server A.
>>
>> I do the same in server B.
>>
>
> you'll need to define each as a "friend" of each other.

Start here:

   http://astrecipes.net/index.php?n=204

Gordon

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Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards

[Brian Alexander]

Okay. I ordered a commercially made T1 crossover cable, connected all of the
cables and rebooted both computers.

I no longer get the 'Got S-frame while link down' or 'HDLC Bad FCS' errors.
However, I still receive the series of 'Detected alarm on channel NN: Red
Alarm' and 'Unable to disable echo cancellation on channel NN'. Followed by
the alarms clearing.

At this point I am confident that the cabling is not the problem. I have
included my zaptel and zapata conf files again below.

Is there a way to make the second TE120P card pass on the timing received
from the first? (rather than using software timing for the pri_net
signalling)

The errors all seem to be about echo cancellation... What do I need to do to
force asterisk to never disable echo cancellation?

Thanks again for all of the help. Even though we have not found a solution
yet I appreciate the help - and am still confident we will succeed!

-Brian



Machine1
=

zaptel.conf
---
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

zapata.conf
---
[trunkgroups]
[channels]
group=1
context=fromtelco
signalling=pri_cpe
switchtype=national
channel=>1-23

group=1
context=frommachine2
signalling=pri_net
switchtype=national
channel=>25-47

Machine2
=

zaptel.conf
---
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

zapata.conf
---
[trunkgroups]
[channels]
group=1
context=frommachine1
signalling=pri_cpe
switchtype=national
channel=>1-23
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Re: [asterisk-users] IAX configuration

[Anthony Messina]

On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
> Hi,
>
> I have some problems and doubts connecting two asterisk servers.
>
> I have one asterisk (serverA), with 1 sip client registered (clientA).
> I have another asterisk (sever B), with another client (clientB).
> Now I want to call from client A to B and from B to A.
>
> Searching in google i find many configuration examples. For instance:
> http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_us
>ing_IAX There are hundreds of pages like this. Also the book "Asterisk the
> future of telephony" describes how to configure IAX.
> But it seems that there are many things missing.
>
> I understand I need to configure iax.conf in server A and add an entry
> for server B as "peer".
> I also need to include a "register" line in iax.conf of server B. If i
> do this, i have server B registered in server A.
>
> I do the same in server B.
>

you'll need to define each as a "friend" of each other.


-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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[asterisk-users] IAX configuration

[yonoko molomo]

Hi,

I have some problems and doubts connecting two asterisk servers.

I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with another client (clientB).
Now I want to call from client A to B and from B to A.

Searching in google i find many configuration examples. For instance:
http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_using_IAX
There are hundreds of pages like this. Also the book "Asterisk the
future of telephony" describes how to configure IAX.
But it seems that there are many things missing.

I understand I need to configure iax.conf in server A and add an entry
for server B as "peer".
I also need to include a "register" line in iax.conf of server B. If i
do this, i have server B registered in server A.

I do the same in server B.

The problem is that I see some "rejected" messages, and the servers
are not registering. I do not see anything else that i need to do in
the configuration examples.
Finally, after doing a lot of tests, I tried adding in "users.conf"
some lines like:
[serverA]
type=peer
callwaiting = yes
fullname = serverA
hasagent = yes
hasdirectory = no
hasiax = yes
hasmanager = no
hassip = no
hasvoicemail = yes
host = dynamic
secret = serverA
threewaycalling = yes
vmsecret = 1234
registeriax = yes
registersip = no
canreinvite = no
nat = no
dtmfmode = rfc2833

im not sure of some of these parameter. but basicalli i define a new
user for the server. if i do not define a user like this in users.conf
the registration of the server is rejected.

I type in the CLI> iax2 show registry
and i see that server A is registered in B and B in A. so it seems fine.

Now I update the extensions.conf file accordingly.
in server A i define something like
exten => clientA_Number,1,Dial(sip/$(exten),10)
exten => clientA_Number,2,hangup()
exten => clientB_Number,1,Dial(iax2/serverB,$(exten))
exten => clientB_Number,2,hangup()

i do the similar in serverB.

i am very frustrated because the call does not work. the error is Call
rejected by x.x.x.x  No authority found

Again, after doing a lot of tests, i noticed that if i add a new user
in iax.conf like:
[clientB]
type=friend
username=clientB
secret=userB
...

and i do the following in serverB's iax.conf:
register => clientB:[EMAIL PROTECTED]

then it seems to work.
but i have some doubts on it.
i have:
- clientA registered in serverA
- clientB registered in serverB
- serverA registered in serverB
- serverB registered in serverA

then, why do i need to do an explicit register of clientB in server A
in the iax.conf and in the users.conf?
i am not sure if this is mandatory (i did not find this configuration
in any example, but if i do not do this, the call is rejected in the
remote asterisk server)

why should server A know all users in server B?
is it not enough if server B is registered in server A?

i am not an expert in asterisk and i am not sure if this is the
correct configuration, but i am a bit tired testing and this is the
only way i found to make the calls work, but i dont find it is
reasonable, especially when the system grows and then my asterisk
server peers with many other servers (do i need to configure my
users.conf and iax.conf for *each* of the users of other asterisks?)

could someone give me some hints?
thanks

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Re: [asterisk-users] ChanSpy issue

[Ed Nuñez]

Good point, but the deal is that I have a remote call center with their own
Nortel PBX.  I get these calls from my DID provided via Zap and I send them
VoIP to the gateway connected to the Nortel PBX.  This is what I refer to my
SIP trunk.  When I specify Sip/SIPTRUNK(SIPTRUNK) is the name of the
trunk.  Asterisk only monitors one call at a time in the whole trunk, and
you can Cycle through the calls by pressing "*". 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Wednesday, September 26, 2007 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy issue

I am not an expert on chanspy, but it seems to me spying on the trunk
would not work very well, would not you hear multiple conversations
mixed if more than one extension were calling?  Seems best to me to
spy on an extension.  YOu also can do a show channels to see who is
talking to whom.

on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote
 > The parameter to Chanspy should be the whole or part of the channel name.
I do not understand what you mean by "sip trunk". It make perfect sense that
you can hear both streams of voice when you use the phone's extension as
Asterisk usually uses "SIP/extension+xxx" as the channel name of the call.
 > 
 > 
 > -Original Message-
 > From: [EMAIL PROTECTED] on behalf of Ed Nuñez
 > Sent: Wed 9/26/2007 4:48 PM
 > To: [EMAIL PROTECTED]
 > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 > Subject: Re: [asterisk-users] ChanSpy issue
 >  
 >  
 > 
 > Hello list
 > 
 >  
 > 
 > I am having an issue with Chanspy/SIP that I'm hoping someone has come
 > across and resolved in the past.
 > 
 >  
 > 
 > I am sending calls that come in TDM through T1 ZAP channels and go out to
a
 > SIP trunk.
 > 
 >  
 > 
 > If I spy on the SIP channel, I can hear the person on the SIP side of the
 > call just fine, but the person on the ZAP channel fades in and out.
 > 
 > If I spy on the ZAP channel, and can hear both sides just fine, but I
don't
 > know who I am spying on since I have other calls coming in on the same
T1.
 > 
 >  
 > 
 > If I spy on a SIP extension instead of a SIP trunk, I hear both sides
just
 > fine.
 > 
 >  
 > 
 > I am using a recent version of Asterisk 1.2 and I am using g729 licenses.
 > 
 >  
 > 
 > This is the command I am using to spy.
 > 
 >  
 > 
 > exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))
 > 
 >  
 > 
 >  
 > 
 > 
 > 
 >  
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > RE: [asterisk-users] ChanSpy issue
 > 
 > 
 > 
 > 
 > The parameter to Chanspy should be the whole or part of
the channel name. I do not understand what you mean by "sip
trunk". It make perfect sense that you can hear both streams of voice
when you use the phone's extension as Asterisk usually uses
"SIP/extension+xxx" as the channel name of the call.
 > 
 > 
 > -Original Message-
 > From: [EMAIL PROTECTED] on behalf of Ed Nuñez
 > Sent: Wed 9/26/2007 4:48 PM
 > To: [EMAIL PROTECTED]
 > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 > Subject: Re: [asterisk-users] ChanSpy issue
 > 
 > 
 > 
 > Hello list
 > 
 > 
 > 
 > I am having an issue with Chanspy/SIP that I'm hoping someone has
come
 > across and resolved in the past.
 > 
 > 
 > 
 > I am sending calls that come in TDM through T1 ZAP channels and go out to
a
 > SIP trunk.
 > 
 > 
 > 
 > If I spy on the SIP channel, I can hear the person on the SIP side of
the
 > call just fine, but the person on the ZAP channel fades in and out.
 > 
 > If I spy on the ZAP channel, and can hear both sides just fine, but I
don't
 > know who I am spying on since I have other calls coming in on the same
T1.
 > 
 > 
 > 
 > If I spy on a SIP extension instead of a SIP trunk, I hear both sides
just
 > fine.
 > 
 > 
 > 
 > I am using a recent version of Asterisk 1.2 and I am using g729
licenses.
 > 
 > 
 > 
 > This is the command I am using to spy.
 > 
 > 
 > 
 > exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > ___
 > 
 > Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/ 
 > 
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 > 
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] Music On Hold - How to increase volume ?

[Tzafrir Cohen]

On Thu, Sep 27, 2007 at 11:54:04PM +1200, Richard wrote:
> I processed mine by using foobar2000 with the equalizer set with some gain
> and then the advanced limiter DSP plugin.  You could do the same with winamp
> and the diskwriter output plugin.
> 

or something of the sort of:

  sox -v 0.5 input_file output_file

Or:

  sox -v 1.5 input_file output_file

If you need limitergain:

  sox input_file output_file vol 1.5 amplitude 0.05

(Here: a limiter value of 0.05).

Sadly I have no time to check this myself, and have to take the word of
the sox man page for it.

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

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[asterisk-users] astcc sometimes doesnt write on mysql

[wassim darwish]

Hi:
I noticed that astcc on my asterisk server sometimes it doesnt write on mysql 
,example :when the caller hangup the call its didnt written on cdrs table nor 
subtract the cost of the call  from the face value of caller card number.This 
problem occured sometimes and not always.

Regards;
wassim
_
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[asterisk-users] ADIT TDM T1 <> Asterisk MGCP

[Barton Fisher]

I have this idea to use an old ADIT 600 with a CMG card to convert two T1
TDM circuits to MGCP towards asterisk.  The basics I've found on the net,
but there is not much available.

 

I've cut and pasted the mgcp.conf details I could find, but there not much
as far as CMG setup.

 

I was hoping I could hook-up with someone that's tried this so I could pick
your brain about the finer details.

 

Thanks, Bart

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Re: [asterisk-users] Music On Hold - How to increase volume ?

[Richard]

I processed mine by using foobar2000 with the equalizer set with some gain
and then the advanced limiter DSP plugin.  You could do the same with winamp
and the diskwriter output plugin.

 

Still not loud and distorted, but certainly loud enough that you can put the
phone down on the desk and still hear it without needing speakerphone.

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rogerio Pazini
Sent: Thursday, September 27, 2007 11:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Music On Hold - How to increase volume ?

 

Hi all,

 

I´ve searched many Internet pages to see how to increase music on hold
volume and I got no success so far. Does anyone have any hint on how to do
that ?

 

 

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Re: [asterisk-users] help with channelbank audiocodes MP-124

[Leonardo Silva]

Carlos,

 What's the help do you need?

Leonardo Silva



2007/9/26, Carlos Hernandez <[EMAIL PROTECTED]>:
>
> Hi:
>
> We're offering some sort of reward to that one who can help us
> For this site we are using trixbox and Asterisk
> 1.2
>
> More info off list.
>
> Many thanks,
> Carlos
>
>
>
>
> ___
>
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-- 
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fone: 16 8143-1146
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Re: [asterisk-users] Music On Hold - How to increase volume ?

[Dean Collins]

http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf 
 
Specifying the Music 

The sample music on hold file (/etc/asterisk/musiconhold.conf) will contain: 

[classes]
;default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rogerio Pazini
Sent: Thursday, 27 September 2007 7:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Music On Hold - How to increase volume ?

 

Hi all,

 

I´ve searched many Internet pages to see how to increase music on hold volume 
and I got no success so far. Does anyone have any hint on how to do that ?

 

Tks !

 

Rogério.

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[asterisk-users] Music On Hold - How to increase volume ?

[Rogerio Pazini]

Hi all,

I´ve searched many Internet pages to see how to increase music on hold
volume and I got no success so far. Does anyone have any hint on how to do
that ?

Tks !

Rogério.
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[asterisk-users] 3-way calling

[Rilawich Ango]

>From the web site said: 3-way Calling: Normally implemented by the
phone.  Can I do it in asterisk?  How?

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