RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?

[Salvatore Giudice]

I suspect that Jed has a substance abuse problem and that he may be in
rehab. I don't know for sure of course. This kind of silence is indicative
of people being hauled back to rehab. Anyway, maybe he just makes a habit of
running off with people's money.


--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Monday, April 30, 2007 2:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Anyone having trouble with claling US
Domesticon Sellvoip?

On 2007-03-26 01:46:40 -0700, "Salvatore Giudice" 
<[EMAIL PROTECTED]> said:

> 
> 
> This is a multi-part message in MIME format.
> 
> I opened up a ticket with them, but I'm not holding my breath. I think
it's
> time to start moving my DID's before the inbound stops working.

That seems like it was probably wise and I hope you followed through.  
I am now unable (for a week or so) to dial any outbound  calls, or 
receive any at my did.

Additionally when trying to call them at there local phone I get the 
disconnected message.

They provided by FAR the best call quality for me when they where 
working,  so I am going to miss them if they are gone forever. Also,  I 
still have like 24$ (us) credit with them...

I still hope they return, but wouldn't count on it.


Marty



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RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?

[Salvatore Giudice]

I have transitioned to other DID's. I think that company is out of business.

Sellvoip is best avoided at all costs.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Monday, April 30, 2007 2:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Anyone having trouble with claling US
Domesticon Sellvoip?

On 2007-03-26 01:46:40 -0700, "Salvatore Giudice" 
<[EMAIL PROTECTED]> said:

> 
> 
> This is a multi-part message in MIME format.
> 
> I opened up a ticket with them, but I'm not holding my breath. I think
it's
> time to start moving my DID's before the inbound stops working.

That seems like it was probably wise and I hope you followed through.  
I am now unable (for a week or so) to dial any outbound  calls, or 
receive any at my did.

Additionally when trying to call them at there local phone I get the 
disconnected message.

They provided by FAR the best call quality for me when they where 
working,  so I am going to miss them if they are gone forever. Also,  I 
still have like 24$ (us) credit with them...

I still hope they return, but wouldn't count on it.


Marty



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RE: [asterisk-users] IVR dictionary dial-plan

[Yuan LIU]

From: Steve Kennedy <[EMAIL PROTECTED]>
Date: Mon, 30 Apr 2007 19:33:43 +0100

Does anyone know of an (E)AGI or program to develop a IVR dial-plan
which will take a list of words and then do something when a unique
branch has been found.

i.e.

Say there's 3 words
demon
deacon
bishop

On a phone they'd be represented as
33666
332266
247467

So if the user enters "2" we know they want bishop
if they enter "336" they want demon and "332" they want deacon.


There was a similar discussion in the forum, 
http://forums.digium.com/viewtopic.php?t=14559.  Don't seem to have a ready 
answer.


Yuan Liu


Could run the dictionary through a script which could generate the
dial-plan or do it via some script interactively.

Any help appreciated.


Steve

--
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com



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Re: [asterisk-users] Re: Voicemail on Different Server (MySQL Replication split thread)

[Stephen Bosch]

JR Richardson wrote:
>> Your experience with database replication is not unique. I have seen
>> this happen with many flavours of database, not just MySQL. At the
>> "critical" sites where I've worked, database replication is not even on
>> the table as an option for precisely the reasons you state above: I have
>> yet to meet someone else who has had a positive experience with it.
> 
> I'm using MySQL replication for my realtime database, 1 master with
> 10+ slaves.  I have not had any data corruption or problems since
> implementation 6 mos ago.  To qualify that I must say, I only do
> one-way replication. Master and Slaves are on the same switch fabric,
>> .7msec latency between hosts.  I attended a MySQL HA class,
> Replication was covered in depth and discussed heavily, at no time did
> I receive cautionary information about possible data corruption or bad
> experiences from others in the class or from the instructors.

Having master and slave servers in the same switch fabric is the only
situation in which I would consider replication.

The cases that I described were with machines in separate subnets.
Replication simply doesn't work that well when there is significant
latency. Did they mention that in your HA class?

> I do daily backups of the master database and also regularly check bin
> file status between all servers to ensure no server is falling behind.
> 
> I'm not trying to dispute or start a flame war, I'm sure replication
> is not perfect and 100% reliable for every instance.  I'm sure there
> are many stories of failed replication or data corruption when
> replication is not implemented properly or setup in an environment not
> particularly suited well for replication.  Just wanted to add my own
> experience.

Thanks,

-Stephen-

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[asterisk-users] Segfaults in 1.2.18

[Grey Man]

Hi,

I have upgraded two production Asterisk servers to 1.2.18
from 1.2.17 in response to the security alert on April the 25th. Since
the upgrade one server has seg faulted 4 times and the other 2, nothing
else has changed on the two servers in the recent past. I use the
safe_asterisk script so have core dumps from all the seg faults but
running the backtrace only yeilds"unresolved symbol" messages so I have
not been able to isolate which module the seg fault is occurring in. I
did recompile Asterisk on one of the servers with the dont-optimize
option and waited two days for the next seg fault but still did got the
"unresolved symbol" messages, I'm probably not doing the build
correctly.

I'm going to downgrade to 1.2.17, which I had no seg
faults with on either server in approximately 2 months, and take the
risk that no malformed SIP responses will get sent to my servers. I run
90% of my SIP traffic through a SIP proxy anyway and the malformed SIP
response would be rejected by it but as my Asterisk servers do use some
SIP trunks for the other 10% of traffic I went ahead with the upgrade
on the hope of avoding any crashes :(.

If anyone else has
noticed the same thing it would be worth posting to see if it's a code
issue rather then a deployment issue on my side.

Regards,

The Grey Man


Send instant messages to your online friends http://au.messenger.yahoo.com 

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[asterisk-users] Re: Voicemail on Different Server (MySQL Replication split thread)

[JR Richardson]

Your experience with database replication is not unique. I have seen
this happen with many flavours of database, not just MySQL. At the
"critical" sites where I've worked, database replication is not even on
the table as an option for precisely the reasons you state above: I have
yet to meet someone else who has had a positive experience with it.


I'm using MySQL replication for my realtime database, 1 master with
10+ slaves.  I have not had any data corruption or problems since
implementation 6 mos ago.  To qualify that I must say, I only do
one-way replication. Master and Slaves are on the same switch fabric,

.7msec latency between hosts.  I attended a MySQL HA class,

Replication was covered in depth and discussed heavily, at no time did
I receive cautionary information about possible data corruption or bad
experiences from others in the class or from the instructors.

I do daily backups of the master database and also regularly check bin
file status between all servers to ensure no server is falling behind.

I'm not trying to dispute or start a flame war, I'm sure replication
is not perfect and 100% reliable for every instance.  I'm sure there
are many stories of failed replication or data corruption when
replication is not implemented properly or setup in an environment not
particularly suited well for replication.  Just wanted to add my own
experience.

JR
--
JR Richardson
Engineering for the Masses
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RE: [asterisk-users] CDR and Billing Issue

[Jonathan Barratt]

We have the same problem, and it also showed up when clients made three+
way calls. The CDRs would show them making the same call three times
simultaneously as the destination field for the second and third calls
was still showing the first number they had dialed. One of our clients
caught it and asked us how he could have been calling the same person at
the same time as he was already calling them...  Quite embarrassing. Not
sure if issue has been fixed in subsequent release or not...

 

Best,

Jonathan

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Monday, April 30, 2007 2:37 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR and Billing Issue

 

Hi Guys,

I am having an issue that I have been able to replicate and I want to
know if anyone else has this.

Extension 100 dials an external number. He speaks for 5 minutes and then
transfers the call to extension 200. Extension 200 speaks for 1 hour.
When we go through the call logs we see the five minute call to the
external number from extension 100. We then see a call from extension
100 to extension 200 for 1 hour. The issue we are having is that we are
billing the clients (100 and 200 are both the same client as ours) for
calls only that hit the PSTN and not internal calls. The issue comes in
that if the call is transfer from one extension to another since we see
it as a call from one extension to another we assume that  it is an
internal call. Is there any way to "fix asterisk" so that it doesn't do
this, am I doing some thing wrong or do all calls have to be attended
transfers ? (We don't want to tell this to the clients because then they
will figure out the loop hole).

 

Thanks a lot.

 

Dovid 

 

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Re: [asterisk-users] Zaptel kernel module load order

[Tzafrir Cohen]

On Mon, Apr 30, 2007 at 07:14:22PM -0500, Mitch Jackson wrote:
> Evening,
> 
> My latest asterisk box is having a difficult problem.  It is
> configured with one TE210P and TDM400P with four FXO modules.  I'm
> running FC6.
> 
> The TE210P only has a single PRI.

You have exactly two modules. either wctdm loads and registers its span 
first or wct4xxp loads first and registers its spans.

There are three cases to consider:

1. system boot 

2. Modules load in the zaptel init script

3. manual load

It would make much sense to put the T1 card first, o it will
automatically become the master, and you have indeed configured the
cards so,

The order in (1) is arbitrary but fixed. It is affected by the order of
scanning of the PCI bus by udev or someone near-by. It may be modified
by changing PCI slots and such.

(2) is normally mostly meaningful on "reastart": unloading and reloading
of the modules. The order in the init.d script is set manually.
The order is set by the value of MODULES, as set in
/etc/sysconfig/zaptel .

With (3) you do what you want. But this is no way for automation...

> 
> When the system boots, it is completely random what order the zaptel
> modules will get loaded in.  Sometimes zttool shows the FXO as the
> last span, sometimes as the first.  When it does load as the first,
> which happens more often, nothing will initialize properly.  When this
> happens, I have to unload all the zaptel modules, and re-load them
> over and over again, until the hardware comes up in the correct order.
> The order it is loaded is in no way related to what order I load the
> modules on the command line.  This problems makes it unlikely that
> asterisk will start properly if the system is rebooted.
> 
> Is there something I can do to ensure the modules get loaded in the
> correct order?
> 
> Here's my config files, if they will help...
> 
> # cat /etc/zaptel.conf
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> defaultzone=us
> loadzone=us
> 
> span=2,1,0,esf,b8zs

off-topic, but:

span=2,2,0,esf,b8zs

(give it lower priority), right?

> bchan=25-47
> dchan=48
> defaultzone=us
> loadzone=us
> 
> fxoks=49-52
> defaultzone=us
> loadzone=us

You only need to put defaultzone and loadzone once (harmless, though).

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Zaptel kernel module load order

[Joel Hill]

It's generally not recommended to put an analog and digital card in the
same box, however that being said Try this.
Write a little hack in /etc/rc.local

/sbin/modprobe wct4xxp

sleep 5

/sbin/modprobe wct4xxp

sleep 5

/sbin/ztcfg

sleep 5

/sbin/modprobe wctdm

sleep 5

/sbin/ztcfg

/usr/sbin/safe_asterisk

the rc.local script is loaded after all the others so it won't effect
anything else, and we had some trouble with some low heat VIA
motherboards so we did the modprobe twice for the PRI.
Hope this helps.

Cheers,

Joel Hill
Support Engineer
Asterisk IT


On Mon, 2007-04-30 at 19:14 -0500, Mitch Jackson wrote:
> Evening,
> 
> My latest asterisk box is having a difficult problem.  It is
> configured with one TE210P and TDM400P with four FXO modules.  I'm
> running FC6.
> 
> The TE210P only has a single PRI.
> 
> When the system boots, it is completely random what order the zaptel
> modules will get loaded in.  Sometimes zttool shows the FXO as the
> last span, sometimes as the first.  When it does load as the first,
> which happens more often, nothing will initialize properly.  When this
> happens, I have to unload all the zaptel modules, and re-load them
> over and over again, until the hardware comes up in the correct order.
>  The order it is loaded is in no way related to what order I load the
> modules on the command line.  This problems makes it unlikely that
> asterisk will start properly if the system is rebooted.
> 
> Is there something I can do to ensure the modules get loaded in the
> correct order?
> 
> Here's my config files, if they will help...
> 
> # cat /etc/zaptel.conf
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> defaultzone=us
> loadzone=us
> 
> span=2,1,0,esf,b8zs
> bchan=25-47
> dchan=48
> defaultzone=us
> loadzone=us
> 
> fxoks=49-52
> defaultzone=us
> loadzone=us
> 
> # cat /etc/asterisk/zapata.conf
> [channels]
> language=en
> switchtype=national
> context=incoming
> faxdetect=none
> signalling=pri_cpe
> group=1
> echocancel=yes
> resetinterval=never
> channel => 1-23
> 
> language=en
> switchtype=national
> context=incoming
> faxdetect=none
> signalling=pri_cpe
> group=3
> echocancel=yes
> resetinterval=never
> channel => 25-47
> 
> 
> signalling=fxo_ks
> usecallerid=yes
> callerid=Fidelity Reserves
> group=2
> threewaycalling=no
> context=outgoing
> channel => 49-52
> 
> 
> 
> 
> 
> Thanks for any help,
> 
> /Mitch
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RE: [asterisk-users] ADSL routers with integrated SIP QoS for other devices

[Salvatore Giudice]

Try the intertex gateways http://www.intertex.se/

Here their page outlining the their QoS settings: 
http://www.intertex.se/products/page.asp?iPageID=143

They have models with ADSL models and wireless access point components.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: Saturday, April 28, 2007 11:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] ADSL routers with integrated SIP QoS for other devices

Greetings list,

Thanks to all who replied to my thread a few days ago "SIP devices with packet 
loss tolerance". One of the suggestions that came out of that thread was to 
replace routers at users' premises with ones that support QoS.

I've used m0n0wall's QoS in the past with reasonable success, but it's quite a 
bulky and complex setup for deploying to remote sites which I'll never visit 
(minimum 3 boxes - ADSL modem, m0n0, WiFi AP).

So, does anyone have any recommendations for a wireless ADSL router with 
integrated QoS for SIP/RTP? I've looked at some of the Draytek units (e.g. 
Vigor 2700V), but I can't find reference as to whether the integrated QoS 
applies only to the FXS ports in the router itself, or to all SIP traffic (most 
of the users will have separate SIP hardphones). These are all to be used in 
the UK, so the device in question needs to support PPPoA.

Any suggestions gratefully appreciated.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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Re: [asterisk-users] Zaptel kernel module load order

[dave cantera]

mitch,
not that I can answer your problem but is this ver 1.4.1?  I had a 
similiar problem in that zapscan was updating the zaptel.conf and 
nothing would work until I mucked with zaptel.conf.zapscan... I might 
have the filename wrong as I have multiple files now :(...  it has 
zapscan in the filename...

daveC

Mitch Jackson wrote:

Evening,

My latest asterisk box is having a difficult problem.  It is
configured with one TE210P and TDM400P with four FXO modules.  I'm
running FC6.

The TE210P only has a single PRI.

When the system boots, it is completely random what order the zaptel
modules will get loaded in.  Sometimes zttool shows the FXO as the
last span, sometimes as the first.  When it does load as the first,
which happens more often, nothing will initialize properly.  When this
happens, I have to unload all the zaptel modules, and re-load them
over and over again, until the hardware comes up in the correct order.
The order it is loaded is in no way related to what order I load the
modules on the command line.  This problems makes it unlikely that
asterisk will start properly if the system is rebooted.

Is there something I can do to ensure the modules get loaded in the
correct order?

Here's my config files, if they will help...

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

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
defaultzone=us
loadzone=us

fxoks=49-52
defaultzone=us
loadzone=us

# cat /etc/asterisk/zapata.conf
[channels]
language=en
switchtype=national
context=incoming
faxdetect=none
signalling=pri_cpe
group=1
echocancel=yes
resetinterval=never
channel => 1-23

language=en
switchtype=national
context=incoming
faxdetect=none
signalling=pri_cpe
group=3
echocancel=yes
resetinterval=never
channel => 25-47


signalling=fxo_ks
usecallerid=yes
callerid=Fidelity Reserves
group=2
threewaycalling=no
context=outgoing
channel => 49-52





Thanks for any help,

/Mitch
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[asterisk-users] Zaptel kernel module load order

[Mitch Jackson]

Evening,

My latest asterisk box is having a difficult problem.  It is
configured with one TE210P and TDM400P with four FXO modules.  I'm
running FC6.

The TE210P only has a single PRI.

When the system boots, it is completely random what order the zaptel
modules will get loaded in.  Sometimes zttool shows the FXO as the
last span, sometimes as the first.  When it does load as the first,
which happens more often, nothing will initialize properly.  When this
happens, I have to unload all the zaptel modules, and re-load them
over and over again, until the hardware comes up in the correct order.
The order it is loaded is in no way related to what order I load the
modules on the command line.  This problems makes it unlikely that
asterisk will start properly if the system is rebooted.

Is there something I can do to ensure the modules get loaded in the
correct order?

Here's my config files, if they will help...

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

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
defaultzone=us
loadzone=us

fxoks=49-52
defaultzone=us
loadzone=us

# cat /etc/asterisk/zapata.conf
[channels]
language=en
switchtype=national
context=incoming
faxdetect=none
signalling=pri_cpe
group=1
echocancel=yes
resetinterval=never
channel => 1-23

language=en
switchtype=national
context=incoming
faxdetect=none
signalling=pri_cpe
group=3
echocancel=yes
resetinterval=never
channel => 25-47


signalling=fxo_ks
usecallerid=yes
callerid=Fidelity Reserves
group=2
threewaycalling=no
context=outgoing
channel => 49-52





Thanks for any help,

/Mitch
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[asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help

[Bruce McAlister]

Hi All,

I have an issue with the ODBC voicemail storage option with asterisk. All
appears to work fine, however, I get several sql execute warnings. I was
wondering if anyone out there could help me get to the bottom of what is
causing this and how I could possibly go about rectifying it.

The warning message we are getting is as follows:

WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error!
[DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?]

This warning occurs whenever a user leaves a message for an extension. It
also occurs when someone dials in to listen to their messages when they hang
up.

These messages do actually exist within the database, and asterisk does
extract them from the database when playing back or recording messages.

Here is an example when someone leaves a message for someone:


---

-- Executing [EMAIL PROTECTED]:1] AGI("SIP/bruce.mcalister-09051118",
"agi://10.7.0.136:4573?app=getvoicemailexten") in new stack
-- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed,
returning 0
-- Executing [EMAIL PROTECTED]:2] NoOp("SIP/bruce.mcalister-09051118", "Caller
VoiceMail Extension = 3031") in new stack
-- Executing [EMAIL PROTECTED]:3] VoiceMail("SIP/bruce.mcalister-09051118",
"[EMAIL PROTECTED]") in new stack
--  Playing
'/usr/local/asterisk/var/spool/voicemail/users/3031/temp' (language 'en')
--  Playing 'vm-intro' (language 'en')
--  Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/usr/local/asterisk/var/spool/voicemail/users/3031/tmp/hGkNG0 format: wav49,
0x90539c8
-- User ended message by pressing #
--  Playing 'auth-thankyou' (language 'en')
[Apr 30 23:56:03] WARNING[30123]: app_voicemail.c:1280 delete_file: SQL
Execute error!
[DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?]

== Parsing
'/usr/local/asterisk/var/spool/voicemail/users/3031/INBOX/msg0002.txt':
Found
Length is 20600
-- Executing [EMAIL PROTECTED]:4] Hangup("SIP/bruce.mcalister-09051118", "") in
new stack
== Spawn extension (base-out, 170, 4) exited non-zero on
'SIP/bruce.mcalister-09051118'


---

Here is an example when someone listens to their voicemail messages without
deleting any:


---

-- Executing [EMAIL PROTECTED]:1] AGI("SIP/bruce.mcalister-09051118",
"agi://10.7.0.136:4573?app=getvoicemailexten") in new stack
-- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed,
returning 0
-- Executing [EMAIL PROTECTED]:2] NoOp("SIP/bruce.mcalister-09051118",
"voicemail extension=3031") in new stack
-- Executing [EMAIL PROTECTED]:3]
VoiceMailMain("SIP/bruce.mcalister-09051118", "[EMAIL PROTECTED]") in new stack
--  Playing 'vm-password' (language 'en')
--  Playing 'vm-youhave' (language 'en')
--  Playing 'digits/19' (language 'en')
--  Playing 'vm-INBOX' (language 'en')
--  Playing 'vm-and' (language 'en')
--  Playing 'digits/20' (language 'en')
--  Playing 'vm-Old' (language 'en')
--  Playing 'vm-messages' (language 'en')
--  Playing 'vm-onefor' (language 'en')
--  Playing 'vm-INBOX' (language 'en')
--  Playing 'vm-messages' (language 'en')
--  Playing 'vm-first' (language 'en')
--  Playing 'vm-message' (language 'en')
  == Parsing
'/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg.txt':
Found
--  Playing 'vm-unknown-caller' (language
'en')
--  Playing
'/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg' (language
'en')
--  Playing 'vm-message' (language 'en')
--  Playing 'digits/2' (language 'en')
  == Parsing
'/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0001.txt':
Found
--  Playing 'vm-from-phonenumber'
(language 'en')
--  Playing 'digits/4' (language 'en')
--  Playing 'digits/4' (language 'en')
--  Playing 'digits/2' (language 'en')
--  Playing 'digits/0' (language 'en')
--  Playing 'digits/8' (language 'en')
--  Playing 'digits/4' (language 'en')
--  Playing 'digits/4' (language 'en')
--  Playing 'digits/9' (language 'en')
--  Playing 'digits/2' (language 'en')
--  Playing 'digits/2' (language 'en')
--  Playing 'digits/5' (language 'en')
--  Playing 'digits/5' (language 'en')
--  Playing
'/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0001' (language
'en')
--  Playing 'vm-prev' (language 'en')
--  Playing 'vm-message' (language 'en')
--  Playing 'digits/3' (language 'en')
  == Parsing
'/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0002.txt':
Found
--  Playing 'vm-from-phonenumber

RE: [asterisk-users] 100 users - voip lan security and qos ?

[Salvatore Giudice]

This is a pretty common setup. Just make sure you have ACL's restricting
traffic between your data and voice vlan's. Generally, we recommend more
than two VLAN's for QoS and security. Usually customers setup the following:

1.) Voice VLAN's for Phones
2.) Data VLAN's for workstations
3.) Voice server VLAN's for IP telephony servers (anything that handles
communications media)
4.) Data server VLAN's for intranet services
5.) converged communications VLAN's - Remote access VLAN's and workstation
endpoints that have soft phones or IPTV clients fall into this category -
802.1p is recommended for these types of VLAN's
6.) wireless VLAN's - These are seldom built for QoS or streaming media, so
they should be segmented and treated differently.

All VLAN's should be properly segmented from each other. Ie. Data VLAN's
should be restricted from accessing voice VLAN's. All network ingress/egress
points should have appropriate SBC's and application layer gateways
installed. The network should always be constructed to preserve voice
services in the event of a network crisis. If you lose the data side of the
network, 95% of large enterprises will always fall back on their telephone
and conferencing systems for crisis management.

Good luck. 

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Finkelstein
Sent: Sunday, April 29, 2007 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 100 users - voip lan security and qos ?

If you are using a cisco switch (2950, 3560, CE500, 4000, 6500, or 3750)
then you will be able to setup the phone and have the computer daisy
chained to it.

I have a similar setup on mine. Here's how I configure my switch ports
in order to achieve the desired effect:

switchport access vlan 5
switchport voice vlan 6
auto qos voip cisco-phone

This is assuming your data VLAN is configured as VLAN 5, and your VoIP
VLAN is on VLAN 6. This will allow the phone to create a trunk port and
facilitate both end nodes through one switch port.

HTH

- sf

A_ Navone wrote:
> i have a customer that needs to plug the phones into the pc's
> using the pass-through rj45 available on most sip phones
> 
> the question they are asking me is how to keep the data network
> separate from / secure from the voip network
> 
> i understand they can set up vlans but i am hazy on a few details
> 
> 1
> since the phones are plugged into the pc's how will the phones
> be segmented into their own vlan ?
> 
> 2
> assuming the phone sends out a tos bit, how can we confirm
> that the customer's switch can read the tos bit and correctly
> prioritize it ?
> 
> 3
> to prioritize voip in the router (coming from the switch)
> we are looking at the wrtg54L and have
> found these 2 juicy websites
> http://openwrt.org
> and
> http://www.dd-wrt.com/dd-wrtv2/index.php
> 
> has anyone downloaded and flashed the "voip" firmware ?
> does it give worthwhile advantages over the default firmware ?
> does the wrtg54L have any advantages over other routers ?
> 
> any other advice to offer ?
> 
> thank you so much in advance
> 
> _
> Exercise your brain! Try Flexicon.
>
http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglineapr
il07
> 
> 
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Re: [asterisk-users] Confference function

[Edoardo Serra]

Hi Ed

Ed Nuñez ha scritto:


I would like to know if anyone here knows the answer to the following 
question


I need to implement the following conferencing feature for my agents.

 


1.   Agent receives call from caller

2.   Agent conferences a verification service


No problem since here


3.   After finishing the verification, agent needs to drop third 
party (Verification service) and continue on the line with caller.



What is your Verification service ?
A VoIP UA which is called for each received call ?? In this case you 
should kick it from the conference


Here are my suggestion:
- Use MeetM Web Control 
(http://www.voip-info.org/wiki/view/MeetMe-Web-Control)


- Use MeetMe b option and write an AGI which react to DTMF pressed by 
the agent (Pay attention to it: 
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe#MoreonoptionbAGI_BACKGROUND)


- Implement some dirty hack in app_meetme.c (you can define a key which 
kicks every user markned with 'A' option)


Hope it helps

Regards

Edoardo Serra

 

My problem right now is being able to disconnect the third party and 
keeping the caller on the line.  Would this be a function of Asterisk 
or the SIP / IAX phone?  Any comments would be appreciated.


 


Thank you

 


Ed Nuñez



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--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

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Re: [asterisk-users] automatically close a meetme

[Edoardo Serra]

Jerry Geis ha scritto:

I am looking for a way to automatically close a meetme conference
when either a user hangs up or through an agi call?

Look at MeetMe docs.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe

create the MeetMe with the 'x' flag and then put inside it some marked 
users ('A')

when the last marked user leaves the conference is closed

Hope it helps

Edoardo


Some method that would automatically terminate the meetme.

Is there a way to do that?

Jerry
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--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

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Re: [asterisk-users] Improving Asterisk's DNS support

[Remco Post]

Kristian Kielhofner wrote:
> Hello everyone,
> 
>  After several years of using Asterisk I have always been frustrated
> by the support for DNS.  I have seen all kinds of strange behavior
> when Asterisk is used on a system with "iffy" DNS servers:
> 
> - no failover to other DNS servers in /etc/resolv.conf (might be a C
> library thing)

wasn't there some setting for that? I run a dns caching deamon om my *
box (speeds up enum lookups big time), but i seem to recall that some
dns settings could be made

> - chan_sip will sometimes mark even local SIP peers as unreachable
> during/after any DNS problems - why?

because your * can't resolve the names any more?

> - dnsmgr doesn't support SIP (yikes!):
> http://bugs.digium.com/view.php?id=9153
> - other randomness (please contribute your own experiences)
> 
>  What can we do about improving this situation?  At the very least we
> need to extend DNS manager support to SIP.  I'm willing to pay for
> this and any other Asterisk DNS improvements.  Any other ideas?
> 


-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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Re: [asterisk-users] ZAPTEL PROBLEM

[Tzafrir Cohen]

On Mon, Apr 30, 2007 at 11:37:22PM +0300, Tzafrir Cohen wrote:

> The logs show:
> 
> Apr 30 14:50:53 DEBUG[6003] chan_zap.c: Message status for 401 changed from 
> -1 to 0 on 1
> Apr 30 14:51:00 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying 
> channel 3
> Apr 30 14:51:16 DEBUG[6004] chan_sip.c: Stopping retransmission on [snipped 
> by Tzafrir]
> 
> This seems to happen when you disconnect an incoming FXO call.
> 
> 
> 
> ZT_EVENT_REMOVED is defined to be 20 . That package includes a patch
> that has been applied to trunk in rev 58321.

oops.

With one minor difference: the patch that went into trunk included an
extra break that prevented polarity reversal events from falling through
to the ZT_EVENT_REMOVED .

In short: not a problem with trunk, only with my package. I'll upload fixed 
packages shortly.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Improving Asterisk's DNS support

[Kristian Kielhofner]

Hello everyone,

 After several years of using Asterisk I have always been frustrated
by the support for DNS.  I have seen all kinds of strange behavior
when Asterisk is used on a system with "iffy" DNS servers:

- no failover to other DNS servers in /etc/resolv.conf (might be a C
library thing)
- chan_sip will sometimes mark even local SIP peers as unreachable
during/after any DNS problems - why?
- dnsmgr doesn't support SIP (yikes!): http://bugs.digium.com/view.php?id=9153
- other randomness (please contribute your own experiences)

 What can we do about improving this situation?  At the very least we
need to extend DNS manager support to SIP.  I'm willing to pay for
this and any other Asterisk DNS improvements.  Any other ideas?

--
Kristian Kielhofner
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Re: [asterisk-users] ADSL routers with integrated SIP QoS for other devices

[Andrew Kohlsmith]

On Monday 30 April 2007 4:14 pm, bails wrote:
> > I'm still looking for a miniPCI ADSL chipset that Linux can use, or an
> > actual "raw" ADSL non-PCI chipset that I can design into an embedded
> > system. If anyone has any leads, please don't hesitate to contact me!
>
> Any chance we can get to see this as it sounds just what i'm looking for?

Once I find something, yes.  :-)

> > If you're curious, I have my rc.tc script for Linux up on
> > http://mixdown.ca/~andrew/rc.tc.
>
> Forbidden
> You don't have permission to access /~andrew/rc.tc on this server.

Fixed.

-A.
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Re: [asterisk-users] ZAPTEL PROBLEM

[Tzafrir Cohen]

On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote:
> Hi all,
> I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
> seems nice, but i'm not able to make calls nor to receive any. When I
> try to make a call, I keep receiven the "all circuits are busy now"
> message, and when I receive calls, asterisk doesn't seems to care
> (don't get anything on the CLI)
> 
> I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository
> 
> voip:~# asterisk -rx 'zap show status'
> Description  Alarms IRQbpviol 
> CRC4
> Wildcard TDM400P REV I Board 1   OK 0  0  0
> 
> voip:~# asterisk -rx 'zap show channels'
>   Chan Extension  Context Language   MusicOnHold
> pseudofrom-internal   es
>  1from-internal   es
>  2from-internal   es
>  3from-pstn   es
>  4from-pstn   es

After some further digging:

it turns out that when there' a problem, the output of the above command
is actually:
> 
> voip:~# asterisk -rx 'zap show channels'
>   Chan Extension  Context Language   MusicOnHold
> pseudofrom-internal   es
>  1from-internal   es
>  2from-internal   es
>  4from-pstn   es

that is: channel 3 has been destroyed.

The logs show:

Apr 30 14:50:53 DEBUG[6003] chan_zap.c: Message status for 401 changed from -1 
to 0 on 1
Apr 30 14:51:00 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying 
channel 3
Apr 30 14:51:16 DEBUG[6004] chan_sip.c: Stopping retransmission on [snipped by 
Tzafrir]

This seems to happen when you disconnect an incoming FXO call.



ZT_EVENT_REMOVED is defined to be 20 . That package includes a patch
that has been applied to trunk in rev 58321.

Anybody encountered this in Asterisk/trunk with a wctdm card with an FXO
module?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Simple dial plan inquiry

[Steve Finkelstein]

Howdy Noah,

I just re-read my original inquiry and noticed my original purpose for
mailing the list was not simple to dig out of the message.

Ultimately, the dialout macro works fabulous. My issue is that I'd like
to be able to override one particular SIP endpoint with its own unique
callerID versus what is passed in $ARG1. So any exten that hits the
dialout macro will get set to the callerID in $ARG1. My one particular
SIP handset, for argument sake, SIP/123 .. should be set to CallerID = 234.

Does that clear up what I'm trying to accomplish some?

Thanks!

- sf

Noah Miller wrote:
> Hi Steve -
> 
>> [macro-dialout];
>> arg1 = callerid number;
>> arg2 = phone numberl
>> exten => s,1,Set(CALLERID(number)=${ARG1})
>> exten => s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
>> exten => s,3,Set(ARG2=1${ARG2})
>> exten => s,4,Dial(${TRUNK}/${ARG2},,m)
>> exten => s,5,Congestion()
>> exten => s,105,Busy()
>>
>> This macro overrides one SIP endpoint which I use for personal usage and
>> do not wish to contain our default CID which is passed through arg1. Is
>> there anyway I can combine GotoIf/Goto to set it otherwise? I was
>> thinking in terms of pseudo code to do something similar to the
>> following:
> 
> Looks like it should work.  Does it?  Dialplan logic is fairly terse.
> I don't think you'll be able to clean it up much more than that.  If
> you're looking for something that looks prettier, you could always use
> AEL/AEL2.  Of course, in the end AEL code will compile down to
> Dialplan code.
> 
> - Noah
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Re: [asterisk-users] ADSL routers with integrated SIP QoS for other devices

[bails]

Andrew Kohlsmith wrote:

On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote:

Thanks to all who replied to my thread a few days ago "SIP devices with
packet loss tolerance". One of the suggestions that came out of that thread
was to replace routers at users' premises with ones that support QoS.


Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD 
with pf.  These are the best solutions, IMO.


The latest Linux kernels also have SIP connection tracking/matching, so it 
should be possible to mark packets and prioritize based on iptables matching.  
I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not 
play nice with the wanrouter drivers.


(note: there was a recent patch to 2.6.20.4 which apparently has much better 
SIP matching, and has been tested successfully with Asterisk.  I have not 
tested it yet, and the iptables guys have rejected the patch as their 
direction for packet matching is shifting significantly in the near future.  
It can be found at 
http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18860.)


I'm still looking for a miniPCI ADSL chipset that Linux can use, or an 
actual "raw" ADSL non-PCI chipset that I can design into an embedded system.  
If anyone has any leads, please don't hesitate to contact me!


Any chance we can get to see this as it sounds just what i'm looking for?



If you're curious, I have my rc.tc script for Linux up on 
http://mixdown.ca/~andrew/rc.tc. 


Forbidden
You don't have permission to access /~andrew/rc.tc on this server.

 It's loosely based off of wondershaper, but
works much better, IMO.  It does host-based prioritization for VOIP, puts 
mail just underneath bulk traffic, and P2P beyond that (if you have the p2p 
connmark stuff set).  I can completely saturate DSL links with the S518 with 
this config without appreciable VOIP degradation.


Even without an S518, this script works well with external ADSL/cable modems.  
You may have to play with the upload rate; some cheap ADSL modems will 
start "blocking" your upstream traffic beyond as little as 50% of the 
upstream rate.


-A.
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Thanks

Bails
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Re: [asterisk-users] ZAPTEL PROBLEM

[Diego Quintana Cruz]

2007/4/30, Tzafrir Cohen <[EMAIL PROTECTED]>:

On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote:
> Hi all,
> I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
> seems nice, but i'm not able to make calls nor to receive any. When I
> try to make a call, I keep receiven the "all circuits are busy now"
> message, and when I receive calls, asterisk doesn't seems to care
> (don't get anything on the CLI)

set verbose 3?

Call from where? To where?



From PSTN to Asterisk and viceversa

Do you see the relevant channel as offhook in 'zap show channel N' ?


I'm not able to to see the channel anymore.

voip*CLI> zap show channel 3
Unable to find given channel 3

I found that this error happens every time i receive an inbound call:
Apr 30 15:08:39 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED.
Destroying channel 3




Sanity check:

  asterisk -rx 'show channels'


voip*CLI> show channels
Channel  Location State   Application(Data)
0 active channels
0 active calls


voip*CLI> zap show channels
  Chan Extension  Context Language   MusicOnHold
pseudofrom-internal   es
 1from-internal   es
 2from-internal   es
 4from-zaptel es




(hmm... asterisk -n -rx 'show channels'hangs for you as well?)

>
> I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository

asterisk-classic or asterisk-bristuff?

asterisk-classic



>
> voip:~# asterisk -rx 'zap show status'
> Description  Alarms IRQbpviol
> CRC4
> Wildcard TDM400P REV I Board 1   OK 0  0  0
>
> voip:~# asterisk -rx 'zap show channels'
>   Chan Extension  Context Language   MusicOnHold
> pseudofrom-internal   es
>  1from-internal   es
>  2from-internal   es
>  3from-pstn   es
>  4from-pstn   es
>
> I thought it could be an IRQ problem, but everything seems fine
>
> voip:~# cat /proc/interrupts
>   CPU0
>  0:  118621819  XT-PIC  timer
>  1:811  XT-PIC  i8042
>  2:  0  XT-PIC  cascade
>  5:  0  XT-PIC  uhci_hcd:usb3, via82cxxx
>  6:  5  XT-PIC  floppy
>  7:  0  XT-PIC  parport0
>  8:  1  XT-PIC  rtc
>  9:  0  XT-PIC  acpi
> 10:2879759  XT-PIC  uhci_hcd:usb2, eth0
> 11:3048189  XT-PIC  uhci_hcd:usb1, eth1
> 12:  474378440  XT-PIC  ehci_hcd:usb4, wctdm

ehci_hcd:usb4 does normally take all the USB interrupts. However this
issue is probably not related to missed interrupts , if there are any.

> 14:1074418  XT-PIC  ide0
> 15:4239765  XT-PIC  ide1
> NMI:  0
> LOC:  0
> ERR:  0
> MIS:  0



Any help would be appreciated

--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext. 1011
FWD # 764839 Ext. 1011
http://routerman.blogsome.com
http://gst.telecom.pucp.edu.pe
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Re: [asterisk-users] Simple dial plan inquiry

[Noah Miller]

Hi Steve -


[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten => s,1,Set(CALLERID(number)=${ARG1})
exten => s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten => s,3,Set(ARG2=1${ARG2})
exten => s,4,Dial(${TRUNK}/${ARG2},,m)
exten => s,5,Congestion()
exten => s,105,Busy()

This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the following:


Looks like it should work.  Does it?  Dialplan logic is fairly terse.
I don't think you'll be able to clean it up much more than that.  If
you're looking for something that looks prettier, you could always use
AEL/AEL2.  Of course, in the end AEL code will compile down to
Dialplan code.

- Noah
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[asterisk-users] CDR and Billing Issue

[Dovid B]

Hi Guys,
I am having an issue that I have been able to replicate and I want to know if 
anyone else has this.
Extension 100 dials an external number. He speaks for 5 minutes and then 
transfers the call to extension 200. Extension 200 speaks for 1 hour. When we 
go through the call logs we see the five minute call to the external number 
from extension 100. We then see a call from extension 100 to extension 200 for 
1 hour. The issue we are having is that we are billing the clients (100 and 200 
are both the same client as ours) for calls only that hit the PSTN and not 
internal calls. The issue comes in that if the call is transfer from one 
extension to another since we see it as a call from one extension to another we 
assume that  it is an internal call. Is there any way to "fix asterisk" so that 
it doesn't do this, am I doing some thing wrong or do all calls have to be 
attended transfers ? (We don't want to tell this to the clients because then 
they will figure out the loop hole).

Thanks a lot.

Dovid 
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Re: [asterisk-users] Polycom 430 , 501 and 550

[Stephen Bosch]

Jeff Davis wrote:
> The IP 501 supports both Cisco and 802.11af with different cables. While
> there are pin assignments differences, there are also electrical
> differences in the discovery protocols. The special cable is an artifact
> of this.
> 
> I don't know of anyone who was able to make the phone work without the
> cable, and the 501 is not designed to work without it. If it were just a
> matter of pin assignments, then people would be selling cables on eBay.
> 
> I notice that now that 802.11af is THE standard, Polycom is supporting
> it on their new phones without any special cables.

The IEEE PoE standard -- for those who care -- is actually IEEE 802.3af,
not 802.11af. The important one is the 802.3af-2003, because that
introduced provisions for preventing ground loops between the PoE
midspan and the switch.

There's a white paper on the topic at Polycom's website which explains
the need for non-standard cable (the newer version of which does in fact
have some additional electronics in it):

http://polycom.com/common/pw_item_show_doc/1,1276,2766,00.pdf

I suspect that the 330, 430 and 550 phones have some sort of built-in
ground loop detection and prevention.

For anybody contemplating running the 501/301 phones with a straight
Cat5 cable: in most cases, it won't work; in some cases, it will work,
but you can bake your PoE injector, unless said injector has a ground
loop prevention circuit built-in; all the newer ones are required to
have this to be truly 802.3af compliant. Remember that a label
indicating 802.3af compliance is no assurance, since the standard has
been through numerous revisions in the last few years.

The moral of the story -- it is probably safer to use the special cable
if you have 301s or 501s.

Thanks for the feedback!

-Stephen-
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[asterisk-users] IVR dictionary dial-plan

[Steve Kennedy]

Does anyone know of an (E)AGI or program to develop a IVR dial-plan
which will take a list of words and then do something when a unique
branch has been found.

i.e.

Say there's 3 words
demon
deacon
bishop

On a phone they'd be represented as
33666
332266
247467

So if the user enters "2" we know they want bishop
if they enter "336" they want demon and "332" they want deacon.

Could run the dictionary through a script which could generate the
dial-plan or do it via some script interactively.

Any help appreciated.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] ZAPTEL PROBLEM

[Tzafrir Cohen]

On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote:
> Hi all,
> I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
> seems nice, but i'm not able to make calls nor to receive any. When I
> try to make a call, I keep receiven the "all circuits are busy now"
> message, and when I receive calls, asterisk doesn't seems to care
> (don't get anything on the CLI)

set verbose 3?

Call from where? To where?

Do you see the relevant channel as offhook in 'zap show channel N' ?

Sanity check:

  asterisk -rx 'show channels'

(hmm... asterisk -n -rx 'show channels'hangs for you as well?)

> 
> I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository

asterisk-classic or asterisk-bristuff?

> 
> voip:~# asterisk -rx 'zap show status'
> Description  Alarms IRQbpviol 
> CRC4
> Wildcard TDM400P REV I Board 1   OK 0  0  0
> 
> voip:~# asterisk -rx 'zap show channels'
>   Chan Extension  Context Language   MusicOnHold
> pseudofrom-internal   es
>  1from-internal   es
>  2from-internal   es
>  3from-pstn   es
>  4from-pstn   es
> 
> I thought it could be an IRQ problem, but everything seems fine
> 
> voip:~# cat /proc/interrupts
>   CPU0
>  0:  118621819  XT-PIC  timer
>  1:811  XT-PIC  i8042
>  2:  0  XT-PIC  cascade
>  5:  0  XT-PIC  uhci_hcd:usb3, via82cxxx
>  6:  5  XT-PIC  floppy
>  7:  0  XT-PIC  parport0
>  8:  1  XT-PIC  rtc
>  9:  0  XT-PIC  acpi
> 10:2879759  XT-PIC  uhci_hcd:usb2, eth0
> 11:3048189  XT-PIC  uhci_hcd:usb1, eth1
> 12:  474378440  XT-PIC  ehci_hcd:usb4, wctdm

ehci_hcd:usb4 does normally take all the USB interrupts. However this
issue is probably not related to missed interrupts , if there are any.

> 14:1074418  XT-PIC  ide0
> 15:4239765  XT-PIC  ide1
> NMI:  0
> LOC:  0
> ERR:  0
> MIS:  0

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Polycom 430 , 501 and 550

[William McCloskey]

If it supports the old Cisco POE, you might be able to try this:

568b
1 OrWh
2 Or
3 GrWh
4 Bl
5 WhBl
6 Gr
7 BrWh
8 Br

Phone Side
1 OrWh
2 Or
3 GrWh
4 BrWh
5 Br
6 Gr
7 Bl
8 WhBl

(From voip-info.org wiki, Cisco POE)

That config has allowed me to run 7940g's on a standard Dell POE switch.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Davis
Sent: Monday, April 30, 2007 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 430 , 501 and 550

Stephen Bosch wrote:
> Justin Hamade wrote:
>> The 501 is more "weird" then that.  The cat5 cable with the built in
>> power injector is cool but to use it with a PoE (802.3af) switch you
>> need a "special" cable (the pairs are just different you can probably
>> look it up and make your own).
> 
> Is this true? I read earlier on the list that there's some sort of
logic
> unit in the cable -- but if it's just a matter of pin assignments,
I'll
> make my own effing cable ;)
> 
> Does anybody have a clear answer on this?

The IP 501 supports both Cisco and 802.11af with different cables. While

there are pin assignments differences, there are also electrical 
differences in the discovery protocols. The special cable is an artifact

of this.

I don't know of anyone who was able to make the phone work without the 
cable, and the 501 is not designed to work without it. If it were just a

matter of pin assignments, then people would be selling cables on eBay.

I notice that now that 802.11af is THE standard, Polycom is supporting 
it on their new phones without any special cables.



--
Jeff Davis
Netsource Consulting
Polycom Certified Reseller
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[asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?

[Martin Joseph]

On 2007-03-26 01:46:40 -0700, "Salvatore Giudice" 
<[EMAIL PROTECTED]> said:





This is a multi-part message in MIME format.

I opened up a ticket with them, but I'm not holding my breath. I think it's
time to start moving my DID's before the inbound stops working.


That seems like it was probably wise and I hope you followed through.  
I am now unable (for a week or so) to dial any outbound  calls, or 
receive any at my did.


Additionally when trying to call them at there local phone I get the 
disconnected message.


They provided by FAR the best call quality for me when they where 
working,  so I am going to miss them if they are gone forever. Also,  I 
still have like 24$ (us) credit with them...


I still hope they return, but wouldn't count on it.


Marty



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Re: [asterisk-users] Polycom 430 , 501 and 550

[Jeff Davis]

Stephen Bosch wrote:

Justin Hamade wrote:

The 501 is more "weird" then that.  The cat5 cable with the built in
power injector is cool but to use it with a PoE (802.3af) switch you
need a "special" cable (the pairs are just different you can probably
look it up and make your own).


Is this true? I read earlier on the list that there's some sort of logic
unit in the cable -- but if it's just a matter of pin assignments, I'll
make my own effing cable ;)

Does anybody have a clear answer on this?


The IP 501 supports both Cisco and 802.11af with different cables. While 
there are pin assignments differences, there are also electrical 
differences in the discovery protocols. The special cable is an artifact 
of this.


I don't know of anyone who was able to make the phone work without the 
cable, and the 501 is not designed to work without it. If it were just a 
matter of pin assignments, then people would be selling cables on eBay.


I notice that now that 802.11af is THE standard, Polycom is supporting 
it on their new phones without any special cables.




--
Jeff Davis
Netsource Consulting
Polycom Certified Reseller
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Re: [asterisk-users] Voicemail on Different Server

[Stephen Bosch]

Anthony Rodgers wrote:
> That's the way we want to go, but have been unable to divine the correct
> settings for getting it working with MS Exchange.

Just for laughs...

what sort of problem do you have?

(Stinky, stinky MS Exchange... worst IMAP support -- but hell, maybe we
can find a solution)

-Stephen-
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Re: [asterisk-users] Send Variable in Dial

[Remco Post]

Andres Gomez wrote:
> Hello to all
> 
> I need send a data to sofphones screen when I use a Dial () .
> 

There is the applications SendText, SendImage or SendURL. Also, for SIP
phones you could possibly use SipAddHeader...

> 
> Thanks a lot
> 
> 
> Regards
> 
> Andres Gomez
> 
> 
> 
> 
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-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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[asterisk-users] automatically close a meetme

[Jerry Geis]

I am looking for a way to automatically close a meetme conference
when either a user hangs up or through an agi call?

Some method that would automatically terminate the meetme.

Is there a way to do that?

Jerry
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[asterisk-users] Confference function

[Ed Nuñez]

I would like to know if anyone here knows the answer to the following question

 

I need to implement the following conferencing feature for my agents.

 

1.   Agent receives call from caller

2.   Agent conferences a verification service

3.   After finishing the verification, agent needs to drop third party 
(Verification service) and continue on the line with caller.

 

My problem right now is being able to disconnect the third party and keeping 
the caller on the line.  Would this be a function of Asterisk or the SIP / IAX 
phone?  Any comments would be appreciated.

 

Thank you

 

Ed Nuñez 

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Re: [asterisk-users] Send Variable in Dial

[Philipp Kempgen]

Andres Gomez wrote:

> I need send a data to sofphones screen when I use a Dial () .

SendText()?


Regards,
  Philipp

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

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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[asterisk-users] ZAPTEL PROBLEM

[Diego Quintana Cruz]

Hi all,
I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
seems nice, but i'm not able to make calls nor to receive any. When I
try to make a call, I keep receiven the "all circuits are busy now"
message, and when I receive calls, asterisk doesn't seems to care
(don't get anything on the CLI)

I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository

voip:~# asterisk -rx 'zap show status'
Description  Alarms IRQbpviol CRC4
Wildcard TDM400P REV I Board 1   OK 0  0  0

voip:~# asterisk -rx 'zap show channels'
  Chan Extension  Context Language   MusicOnHold
pseudofrom-internal   es
 1from-internal   es
 2from-internal   es
 3from-pstn   es
 4from-pstn   es

I thought it could be an IRQ problem, but everything seems fine

voip:~# cat /proc/interrupts
  CPU0
 0:  118621819  XT-PIC  timer
 1:811  XT-PIC  i8042
 2:  0  XT-PIC  cascade
 5:  0  XT-PIC  uhci_hcd:usb3, via82cxxx
 6:  5  XT-PIC  floppy
 7:  0  XT-PIC  parport0
 8:  1  XT-PIC  rtc
 9:  0  XT-PIC  acpi
10:2879759  XT-PIC  uhci_hcd:usb2, eth0
11:3048189  XT-PIC  uhci_hcd:usb1, eth1
12:  474378440  XT-PIC  ehci_hcd:usb4, wctdm
14:1074418  XT-PIC  ide0
15:4239765  XT-PIC  ide1
NMI:  0
LOC:  0
ERR:  0
MIS:  0

Hope you can help me with my problem.
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext. 1011
FWD # 764839 Ext. 1011
http://routerman.blogsome.com
http://gst.telecom.pucp.edu.pe
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[asterisk-users] Send Variable in Dial

[Andres Gomez]

Hello to all

I need send a data to sofphones screen when I use a Dial () .


Thanks a lot


Regards

Andres Gomez
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Re: OT: Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Paul]

Richard Lyman wrote:

> Paul wrote:
>
>> Richard Lyman wrote:
>>
>>  
>>
>>> Paul wrote:
>>>
>>>
>>>
 Joe acquisto wrote:

  

  

> Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
>  
>
>   
>
>
>> Joe acquisto wrote:
>>
>>   
>>  
>>
>>> I have dual posted this to the user and biz lists.
>>>
>>> Has anyone ever heard of someone running an Asterisk based system,
>>> yet Has abandoning SugarCRM, and opting to develop their own
>>> Visual FoxPro Has database/CRM?
>>>
>>> Please don't dump on me now, this is not my idea, I am just asking
>>> for Please comments, to see if my own initial thoughts are
>>> reasonably accurate.
>>>
>>>
>>>
>>> 
>>
>> I'll answer it on the user list. I don't think the idea is developed
>> enough to discuss on biz.
>>
>> First - vtiger is available for those who don't like the SugarCRM
>> licensing.
>>   
>
> It's not a licensing complaint.  At least that has not surfaced.  It
> is more that the programmer does not seem to be comfortable with
> SugarCRM, MySQL and php.
> Biggest compliant about sugar is - hard to configure, does not work
> with latest
> php.
>  
>
>   
>
>
>> Second -  developing your own CRM is an ambitious undertaking.  You
>> need
>> good reasons to go in that direction.
>>
>> Third -  I have enough exposure to  Visual FoxPro to quickly rule
>> it out
>> as  a choice for anything new. The fact that somebody is
>> proposing to
>> use it might give you the idea that they don't know what they are
>> talking about at all. BTW - my exposure to it did include things
>> like
>> access from linux apps using ODBC so I know enough to hate the
>> product.
>>   
>
 It still makes me wonder why the programmer chooses Visual Foxpro.
 Sounds like he also rejects many other language and database options.
 
>>>
>>> original poster didn't say it was 'visual foxpro'.
>>> 
>>
>>
>> Yes he did say that. Check the indent levels above to see OP is Joe
>> acquisto.
>>   
>
> yep, i somehow skipped over that part.  more coffee!

Sometimes more coffee is what makes me skip over things

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Re: [asterisk-users] Voicemail on Different Server

[Anthony Rodgers]

That's the way we want to go, but have been unable to divine the correct
settings for getting it working with MS Exchange.

CP

Tim Panton wrote:
>
> If I were starting a project now, I'd
> take a look at the (newish) support for IMAP storage for voicemail.
>
>
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Paul]

Stephen Bosch wrote:

>Paul wrote:
>  
>
Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
as  a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did include things like
access from linux apps using ODBC so I know enough to hate the product.
   



>>It still makes me wonder why the programmer chooses Visual Foxpro.
>>Sounds like he also rejects many other language and database options.
>>
>>
>
>What it *sounds* like is that's all he knows and he's too terrified (or
>unmotivated) to learn anything new.
>
>I have met people who are still programming in OS/360 (using an x86
>emulation environment). Yes, you read that right. These people sell
>*software* to *customers*.
>
>There comes a time in the life of any technology when you have to take
>it behind the barn...
>  
>
Do you mean a trip to the woodshed where I apply the "board of
education" to some technologies I started out with? I don't have any
"tough love" for things like FORTRAN. I'll just disinherit those
children. :)

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[asterisk-users] Re: headsets for linksys/sipura phones?

[PGentilini]

We are using VXI headsets with our Asterisk PBX as well as our legacy
PBX's.  A nice feature of these headsets is that you can use the same
headset in either a USB port with their DSP translator cord or in a
traditional rj-11 port with another cord.  This adds some redundancy to
your system if you want to enable both hard and softphones without buying
all kinds of headsets.  They work great with x-lite as well!
http://www.vxicorp.com/

Paul

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OT: Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Richard Lyman]

Paul wrote:

Richard Lyman wrote:

  

Paul wrote:



Joe acquisto wrote:

 

  

Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
 

   



Joe acquisto wrote:

  
 

  

I have dual posted this to the user and biz lists.

Has anyone ever heard of someone running an Asterisk based system,
yet Has abandoning SugarCRM, and opting to develop their own
Visual FoxPro Has database/CRM?

Please don't dump on me now, this is not my idea, I am just asking
for Please comments, to see if my own initial thoughts are
reasonably accurate.







I'll answer it on the user list. I don't think the idea is developed
enough to discuss on biz.

First - vtiger is available for those who don't like the SugarCRM
licensing.
  
  
  

It's not a licensing complaint.  At least that has not surfaced.  It
is more that the programmer does not seem to be comfortable with
SugarCRM, MySQL and php.
Biggest compliant about sugar is - hard to configure, does not work
with latest
php.
 

   



Second -  developing your own CRM is an ambitious undertaking.  You
need
good reasons to go in that direction.

Third -  I have enough exposure to  Visual FoxPro to quickly rule
it out
as  a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did include things like
access from linux apps using ODBC so I know enough to hate the
product.
  
  
  

It still makes me wonder why the programmer chooses Visual Foxpro.
Sounds like he also rejects many other language and database options.
  
  

original poster didn't say it was 'visual foxpro'.



Yes he did say that. Check the indent levels above to see OP is Joe
acquisto.
  

yep, i somehow skipped over that part.  more coffee!



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[asterisk-users] Simple dial plan inquiry

[Steve Finkelstein]

Hi all,

This is a simple concept, however I'm not entirely comfortable with
available applications and functions available to me to make this happen.

I have a simple dialout macro such as the following:

[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten => s,1,Set(CALLERID(number)=${ARG1})
exten => s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten => s,3,Set(ARG2=1${ARG2})
exten => s,4,Dial(${TRUNK}/${ARG2},,m)
exten => s,5,Congestion()exten => s,105,Busy()

This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the following:

if ($arg1 = SIP/MyPersonal)
{
 set caller ID to mypersonal
 goto s,2
}
else
{
 Set(CALLERID(number)=${ARG1}) ; leave as is
 goto s,2 ; leave as is
}

Thanks for any insight.

- sf
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Stephen Bosch]

Paul wrote:
>>> Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
>>> as  a choice for anything new. The fact that somebody is proposing to
>>> use it might give you the idea that they don't know what they are
>>> talking about at all. BTW - my exposure to it did include things like
>>> access from linux apps using ODBC so I know enough to hate the product.
>>>
>>>
> It still makes me wonder why the programmer chooses Visual Foxpro.
> Sounds like he also rejects many other language and database options.

What it *sounds* like is that's all he knows and he's too terrified (or
unmotivated) to learn anything new.

I have met people who are still programming in OS/360 (using an x86
emulation environment). Yes, you read that right. These people sell
*software* to *customers*.

There comes a time in the life of any technology when you have to take
it behind the barn...

-Stephen-

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Re: [asterisk-users] Polycom 650

[Stephen Bosch]

Klaverstyn, David C wrote:
> All,
> 
> I have a Polycom 650 phone, when turned on displays “Checking application”.
> 
> Can any give me some information as to what is wrong?  I have copied the
> CFG files from a 601 phone to work with this 650.

1. You need at least SIP 2.0.1 (2.1.0 recommended minimum, 2.1.1 now
available)

2. Don't use CFG files from another phone. Follow the Polycom SIP
Administrator's Guide (available on the website); make your changes in
phone-specific config files and use the sip.cfg and phone1.cfg files in
their default state.

-Stephen-
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Paul]

Richard Lyman wrote:

> Paul wrote:
>
>> Joe acquisto wrote:
>>
>>  
>>
>>> Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
>>>  
>>>
>>>
>>>
 Joe acquisto wrote:

   
  

> I have dual posted this to the user and biz lists.
>
> Has anyone ever heard of someone running an Asterisk based system,
> yet Has abandoning SugarCRM, and opting to develop their own
> Visual FoxPro Has database/CRM?
>
> Please don't dump on me now, this is not my idea, I am just asking
> for Please comments, to see if my own initial thoughts are
> reasonably accurate.
>
>
>
> 
> 

 I'll answer it on the user list. I don't think the idea is developed
 enough to discuss on biz.

 First - vtiger is available for those who don't like the SugarCRM
 licensing.
   
   
>>>
>>> It's not a licensing complaint.  At least that has not surfaced.  It
>>> is more that the programmer does not seem to be comfortable with
>>> SugarCRM, MySQL and php.
>>> Biggest compliant about sugar is - hard to configure, does not work
>>> with latest
>>> php.
>>>  
>>>
>>>
>>>
 Second -  developing your own CRM is an ambitious undertaking.  You
 need
 good reasons to go in that direction.

 Third -  I have enough exposure to  Visual FoxPro to quickly rule
 it out
 as  a choice for anything new. The fact that somebody is proposing to
 use it might give you the idea that they don't know what they are
 talking about at all. BTW - my exposure to it did include things like
 access from linux apps using ODBC so I know enough to hate the
 product.
   
   
>>>
>> It still makes me wonder why the programmer chooses Visual Foxpro.
>> Sounds like he also rejects many other language and database options.
>>   
>
>
> original poster didn't say it was 'visual foxpro'.

Yes he did say that. Check the indent levels above to see OP is Joe
acquisto.

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Re: [asterisk-users] Polycom 430 , 501 and 550

[Stephen Bosch]

Justin Hamade wrote:
> The 501 is more "weird" then that.  The cat5 cable with the built in
> power injector is cool but to use it with a PoE (802.3af) switch you
> need a "special" cable (the pairs are just different you can probably
> look it up and make your own).

Is this true? I read earlier on the list that there's some sort of logic
unit in the cable -- but if it's just a matter of pin assignments, I'll
make my own effing cable ;)

Does anybody have a clear answer on this?

-Stephen-
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Re: [asterisk-users] Viable using purchasing sip lines

[Stephen Bosch]

Hi:

I can try and answer some of your questions.

[EMAIL PROTECTED] wrote:
> Hello All,
> 
> We have been doing Asterisk and CME implementations recently but we
> almost always exlusively bring in analog lines and or PRI for PSTN
> access to our systems.  I have known about providers providing SIP based
> lines and SIP trunks to end users for PSTN access.  I am curious about
> the following:
> 
> - How practical is this?  The idea of terminating pstn calls to across
> the Internet which is an unguarenteed medium concerns me.  Even if our
> access to it is quazi stable T1 data type of access.  Do any of you do
> systems where this is soley the method used for incoming calls from the
> pstn?  If this is done are there things to look for in a SIP provider,
> as in their presence on the Internet latency ..etc?

You're absolutely right to be concerned. If you need critical service,
leave the VoIP terminations for the time being.

We *do* use it as an adjunct to primary PSTN lines; so, for example, we
might dump a bunch of extra PSTN lines and replace them with SIP or IAX
connections; in one case, we just "forward on busy" from the main PSTN
number to the SIP numbers, and that works quite well. When it's working
users even like the sound quality better. When it's working ;)

It does work most of the time, but when it comes to phone service, user
expectations are way different (hell, my expectations are way different,
so I can understand).

Here are the things I would recommend you pay close attention to when
choosing a SIP provider:

1. There are numerous fly-by-night operations. Providing a stable, high
availability service is not trivial and costs some money; you want a
provider that offers as close to round-the-clock support as you can get.
One of the best tests is just to call the provider directly and see what
kind of experience you have. When calling some SIP carriers, I have had
dropouts, chirping, clicking, calls terminating nowhere, etc. Those are
the ones you want to avoid :) Also, if you are spending a long time in
queue... be suspicious. There are outfits run by "two guys in a
basement" -- and those two guys are often away skiing or windsurfing,
depending on the season. Just be vigilant.

2. Ask for the IP of the PSTN POP for the provider and check the
latency. I wouldn't tolerate anything higher than 75 ms, and shoot for
something under 50 ms if you can manage it. Sometimes it's not possible
and depends where you're located.

> - What are the major advantages?  I know some places provide all you can
> eat plans which could be seen as a plus and some others provide really
> low rates. Are there others?

The major advantage is that, in general, it is easier to support more
channels (when you need them) and the cost per channel is, on average,
lower than for a PSTN connection. For example, with certain providers,
it's possible to get two channels for slightly more than the cost of a
single business line with the PSTN provider.

Another advantage: if you only need a few channels, and your PSTN lines
are analog, then VoIP connections offer you some call progress
detection, which can be useful when you're trying to do "follow me"
ringing off the premises.

-Stephen-
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Richard Lyman]

Paul wrote:

Joe acquisto wrote:

  

Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
 



Joe acquisto wrote:

   

  

I have dual posted this to the user and biz lists.

Has anyone ever heard of someone running an Asterisk based system, yet 
Has abandoning SugarCRM, and opting to develop their own Visual FoxPro 
Has database/CRM?


Please don't dump on me now, this is not my idea, I am just asking for 
Please comments, to see if my own initial thoughts are reasonably accurate.




 



I'll answer it on the user list. I don't think the idea is developed
enough to discuss on biz.

First - vtiger is available for those who don't like the SugarCRM 
licensing.
   

  
It's not a licensing complaint.  At least that has not surfaced.  It is more that the 
programmer does not seem to be comfortable with SugarCRM, MySQL and php.

Biggest compliant about sugar is - hard to configure, does not work with latest
php. 

 



Second -  developing your own CRM is an ambitious undertaking.  You need
good reasons to go in that direction.

Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
as  a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did include things like
access from linux apps using ODBC so I know enough to hate the product.
   

  

It still makes me wonder why the programmer chooses Visual Foxpro.
Sounds like he also rejects many other language and database options.
  


original poster didn't say it was 'visual foxpro'.

who knows, maybe this is that guy that wrote that http server in foxpro 
dos years ago. 


as for the topic, most people tend to use what they know or *feel* will 
be easiest to integrate.
(it is that simple) 




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Re: [asterisk-users] Voicemail on Different Server

[Stephen Bosch]

Noah Miller wrote:
> At the time I set this up, MySQL replication was really designed for
> one-way replication.  Two way replication was possible, but required
> somewhat unorthodox methods.  (Maybe this has changed, I don't know).
> Configuration is also a little tricky.  It's not too bad to set it up
> between two machines, but 3 machines is more tricky, and 4 is even
> more tricky, etc, etc.  This client had only 3 offices at the time,
> but I knew they would be expanding.  They now have 6.
> 
> Anyway, after getting everything working, I found that replication
> would periodically stop after some time.  I'd have to re-create the
> setup, and then replication would work for a time, and then stop again
> later.  This occurred across several different version of MySQL.  I
> suppose I could have fixed this issue with persistence, but
> unfortunately this was only an annoyance compared to the major issue
> of data corruption.

Your experience with database replication is not unique. I have seen
this happen with many flavours of database, not just MySQL. At the
"critical" sites where I've worked, database replication is not even on
the table as an option for precisely the reasons you state above: I have
yet to meet someone else who has had a positive experience with it.

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[asterisk-users] voicemail + Dynamic mailbox

[mohammad mirzaee]

HI All;

I want to use Asterisk for just Voicemail Server and I need a Dynamic creation 
of Mailboxes.
My users 's Mailboxes are same as "Extensions" but I donot want to add 
mailboxes in
"Voicemail.conf"

Is there any way to create mailbox from Asterisk dial-plan ?

Appreciate any suggestions
Mohammad Mirzaee

Mohammad Mirzaee
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Paul]

I fully understand that but the OP says the programmer "does not seem to
be comfortable with SugarCRM, MySQL and php". That is quite different
from "It's easier to build on the code I already have". If I had
resisted growth and change over the years, I might be looking for ways
to integrate FORTRAN and RPG2 with asterisk today. How about punchcards
for the CDR's?

Bruce Reeves wrote:

> I have heard of people rejecting Sugar for their existing CRM/ERP
> product based on VS Foxpro. I'm not a huge fan of Foxpro myself, but
> if the system already exist then a lot of people see little advantage
> in changing.
>
> On 4/30/07, *Paul* <[EMAIL PROTECTED] > wrote:
>
> Joe acquisto wrote:
>
> >Paul <[EMAIL PROTECTED] > Wrote: 4/30/2007
> 8:53 AM:
> >
> >
> >>Joe acquisto wrote:
> >>
> >>
> >>
> >>>I have dual posted this to the user and biz lists.
> >>>
> >>>Has anyone ever heard of someone running an Asterisk based
> system, yet
> >>>Has abandoning SugarCRM, and opting to develop their own Visual
> FoxPro
> >>>Has database/CRM?
> >>>
> >>>Please don't dump on me now, this is not my idea, I am just
> asking for
> >>>Please comments, to see if my own initial thoughts are
> reasonably accurate.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>I'll answer it on the user list. I don't think the idea is developed
> >>enough to discuss on biz.
> >>
> >>First - vtiger is available for those who don't like the SugarCRM
> >>licensing.
> >>
> >>
> >
> >It's not a licensing complaint.  At least that has not
> surfaced.  It is more that the
> >programmer does not seem to be comfortable with SugarCRM, MySQL
> and php.
> >Biggest compliant about sugar is - hard to configure, does not
> work with latest
> >php.
> >
> >
> >
> >>Second -  developing your own CRM is an ambitious
> undertaking.  You need
> >>good reasons to go in that direction.
> >>
> >>Third -  I have enough exposure to  Visual FoxPro to quickly
> rule it out
> >>as  a choice for anything new. The fact that somebody is
> proposing to
> >>use it might give you the idea that they don't know what they are
> >>talking about at all. BTW - my exposure to it did include things
> like
> >>access from linux apps using ODBC so I know enough to hate the
> product.
> >>
> >>
> It still makes me wonder why the programmer chooses Visual Foxpro.
> Sounds like he also rejects many other language and database options.
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>
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> To UNSUBSCRIBE or update options visit:
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>
>
>
>
> -- 
> Bruce Reeves
> Nortex Networks
>
>
>
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SV: [asterisk-users] Early audio(progress) and MOH

[Håkon Nessjøen]

Hi,
 
This does not work with early audio (the use of Progress() on a Zap channel
before Dial(,20,m)).
 
The caller will not need to pay anything before anyone answers(). But I want
to play music or audio, while the call is progressing.
 
Håkon


  _  

Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Steve Totaro
Sendt: 29. april 2007 19:06
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [asterisk-users] Early audio(progress) and MOH



The m switch should play music just as r will generate a ring.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  


  _  


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Håkon Nessjøen
Sent: Sunday, April 29, 2007 12:28 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Early audio(progress) and MOH

 

Hi,

 

Is it possible to have MOH in early audio, while waiting for someone to pick
up a Dial() call?

(When using zap channels, I have early audio working with playback)

 

Håkon Nessjøen

Loopback Systems AS

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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Bruce Reeves]

I have heard of people rejecting Sugar for their existing CRM/ERP product
based on VS Foxpro. I'm not a huge fan of Foxpro myself, but if the system
already exist then a lot of people see little advantage in changing.

On 4/30/07, Paul <[EMAIL PROTECTED]> wrote:


Joe acquisto wrote:

>Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
>
>
>>Joe acquisto wrote:
>>
>>
>>
>>>I have dual posted this to the user and biz lists.
>>>
>>>Has anyone ever heard of someone running an Asterisk based system, yet
>>>Has abandoning SugarCRM, and opting to develop their own Visual FoxPro
>>>Has database/CRM?
>>>
>>>Please don't dump on me now, this is not my idea, I am just asking for
>>>Please comments, to see if my own initial thoughts are reasonably
accurate.
>>>
>>>
>>>
>>>
>>>
>>I'll answer it on the user list. I don't think the idea is developed
>>enough to discuss on biz.
>>
>>First - vtiger is available for those who don't like the SugarCRM
>>licensing.
>>
>>
>
>It's not a licensing complaint.  At least that has not surfaced.  It is
more that the
>programmer does not seem to be comfortable with SugarCRM, MySQL and php.
>Biggest compliant about sugar is - hard to configure, does not work with
latest
>php.
>
>
>
>>Second -  developing your own CRM is an ambitious undertaking.  You need
>>good reasons to go in that direction.
>>
>>Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
>>as  a choice for anything new. The fact that somebody is proposing to
>>use it might give you the idea that they don't know what they are
>>talking about at all. BTW - my exposure to it did include things like
>>access from linux apps using ODBC so I know enough to hate the product.
>>
>>
It still makes me wonder why the programmer chooses Visual Foxpro.
Sounds like he also rejects many other language and database options.
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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] Test

[Dovid B]

I love these :)
- Original Message - 
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Friday, April 27, 2007 7:54 PM
Subject: Re: [asterisk-users] Test



Failed

On 4/26/07, gc <[EMAIL PROTECTED]> wrote:




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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Paul]

Joe acquisto wrote:

>Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
>  
>
>>Joe acquisto wrote:
>>
>>
>>
>>>I have dual posted this to the user and biz lists.
>>>
>>>Has anyone ever heard of someone running an Asterisk based system, yet 
>>>Has abandoning SugarCRM, and opting to develop their own Visual FoxPro 
>>>Has database/CRM?
>>>
>>>Please don't dump on me now, this is not my idea, I am just asking for 
>>>Please comments, to see if my own initial thoughts are reasonably accurate.
>>>
>>> 
>>>
>>>  
>>>
>>I'll answer it on the user list. I don't think the idea is developed
>>enough to discuss on biz.
>>
>>First - vtiger is available for those who don't like the SugarCRM 
>>licensing.
>>
>>
>
>It's not a licensing complaint.  At least that has not surfaced.  It is more 
>that the 
>programmer does not seem to be comfortable with SugarCRM, MySQL and php.
>Biggest compliant about sugar is - hard to configure, does not work with latest
>php. 
>
>  
>
>>Second -  developing your own CRM is an ambitious undertaking.  You need
>>good reasons to go in that direction.
>>
>>Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
>>as  a choice for anything new. The fact that somebody is proposing to
>>use it might give you the idea that they don't know what they are
>>talking about at all. BTW - my exposure to it did include things like
>>access from linux apps using ODBC so I know enough to hate the product.
>>
>>
It still makes me wonder why the programmer chooses Visual Foxpro.
Sounds like he also rejects many other language and database options.
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Per Jessen]

Time Bandit wrote:

>> > First - vtiger is available for those who don't like the SugarCRM
>> > licensing.
>>
>> It's not a licensing complaint.  At least that has not surfaced.  It
>> is more that the programmer does not seem to be comfortable with
>> SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to
>> configure, does not work with latest php.
> Have a look at vTiger then (fork of SugarCRM). Works with latest PHP
> and MySQL, easy to configure and is free : http://www.vtiger.com/

This is going somewhat OT, but we have pretty much had to shelve our
vtiger project due lack of maturity - support for multiple
countries/languages/currencies/VAT-rates is very poor.  Or at least it
was last time I looked. 


/Per Jessen, Zürich

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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Time Bandit]

> First - vtiger is available for those who don't like the SugarCRM
> licensing.

It's not a licensing complaint.  At least that has not surfaced.  It is more 
that the
programmer does not seem to be comfortable with SugarCRM, MySQL and php.
Biggest compliant about sugar is - hard to configure, does not work with latest
php.

Have a look at vTiger then (fork of SugarCRM). Works with latest PHP
and MySQL, easy to configure and is free : http://www.vtiger.com/


hth
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[asterisk-users] Remodified Asterisk brute force blockers..

[J. Oquendo]

Top of the morning all... So I reworked the pseudo IDS/Brute Force 
Asterisk script for those who want to either use it, or use it as a 
baseline to build a better one...


The script now does a few things... It logs those with password issues, 
and blocks them as well. This was done to ensure that a remote user who 
was blocked can be found in the log. E.g., Sally the homemaker keeps 
fiddling with her ATA or phone... Toasts her password... She will be 
blocked, and her username and IP address will be logged in the home 
directory of the admin running the script. This was done to ensure you 
don't go blowing away legitimate 
(01110011011101000111010101110110100101100100 / PEBKAC) users. It 
also double checks the entries to make sure no one is injecting false 
parameters into Asterisk which would log say... Your own domain...


Some may need to tweak their columns under awk... Test before using on a 
production machine... Works fine for me under Debian and FC5, results 
may vary so test it on your own. If you have to ask about what it does, 
please don't use it... Comments on the awk/sed/grep nightmare... Fire 
away... It was started as a oneliner that spiraled out of control


http://www.infiltrated.net/scripts/ashtray

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g' 


"Wise men talk because they have something to say;
fools, because they have to say something." -- Plato




smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Joe acquisto]

Paul <[EMAIL PROTECTED]> Wrote: 4/30/2007 8:53 AM:
> Joe acquisto wrote:
> 
>>I have dual posted this to the user and biz lists.
>>
>>Has anyone ever heard of someone running an Asterisk based system, yet 
>>Has abandoning SugarCRM, and opting to develop their own Visual FoxPro 
>>Has database/CRM?
>>
>>Please don't dump on me now, this is not my idea, I am just asking for 
>>Please comments, to see if my own initial thoughts are reasonably accurate.
>>
>>  
>>
> I'll answer it on the user list. I don't think the idea is developed
> enough to discuss on biz.
> 
> First - vtiger is available for those who don't like the SugarCRM 
> licensing.

It's not a licensing complaint.  At least that has not surfaced.  It is more 
that the 
programmer does not seem to be comfortable with SugarCRM, MySQL and php.
Biggest compliant about sugar is - hard to configure, does not work with latest
php. 

> Second -  developing your own CRM is an ambitious undertaking.  You need
> good reasons to go in that direction.
> 
> Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
> as  a choice for anything new. The fact that somebody is proposing to
> use it might give you the idea that they don't know what they are
> talking about at all. BTW - my exposure to it did include things like
> access from linux apps using ODBC so I know enough to hate the product.
> 

Thanks. 

joe a.

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Re: [asterisk-users] don't want call to get answered

[Remco Post]

Arun Kumar wrote:
> In my * box I've configured two queues and incoming number and whenever
> any one calls those number call comes to my *box and it sends call to my
> agents in queue. but if no agent is available it still answer the call.
> Is there any why when my agents are not available I don't want call to
> get answered. Here is my dialplan:

I think that you want asterisk not to pick up the call when all of your
agents are busy, right? I guess that in this case you do not want to use
queues, but you want to build some dialgroup.

So you'll need to do a few things

1- have some extension that agents can call to log on. When they do,
append their account (Technology/resource) to a database record (or
global var).
2- have some extension that agents can call to log off. Reverse of the
above.
3- When somebody calls the extension '' from below, you use Dial on
the database entry or global var to call all of your agents. If the dail
fails, you check the dialstatus to see why and possibly retry after so
many seconds (for a limited amount of tries) and then maybe answer the
call to play an announcement that nobody is available and please try
again later, or would they want to wait an be placed in a queue.

> 
> exten => ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
> exten => ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
> exten => ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
> exten => ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
> exten => ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
> exten => ,6,Goto(out-of-hours,5003,1)
> exten => ,7,Answer()
> exten => ,8,Playback(custom/next-avail-advisor)
> exten =>
> ,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
> 
> exten => ,10,Monitor(wav,${MONITOR_FILENAME},mb)
> exten => ,11,Queue(kbsupport,t)
> exten => ,12,Hangup()
> 
> 
> 
> thanks
> arun
> 
> 
> 
> 
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-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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RE: [asterisk-users] TDM400P and Junghanns QuadBRI issue

[Backup e-mail]

The solution to this issue is to edit /etc/sysconfig/zaptel and add the 
following line:
   
  MODULES="$MODULES qozap"# BRISTUFF driver

  Costa

Henk Dick <[EMAIL PROTECTED]> wrote:
v\:* {behavior:url(#default#VML);}  o\:* {behavior:url(#default#VML);}  
w\:* {behavior:url(#default#VML);}  .shape {behavior:url(#default#VML);}
I would check:
   
  Cat /proc/zaptel/
   
  To make shure that the cards are activated in the order that you programmed 
them.
   
   
  Henk
   
  
-
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Backup e-mail
Sent: maandag 30 april 2007 13:26
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TDM400P and Junghanns QuadBRI issue

   
Hi List,

 

I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one 
Junghanns QuadBRI
on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). 
I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, 
libpri-1.2.4 
and asterisk-1.2.17

When it's the time for ztcfg to do its job it complains with 


 

I'm out of ideas what to do to make it to work. Your help is very much 
appreciated.

The config files and results from various commands follow.

 

Thanks,


Costa.

---
Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Loopstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Loopstart (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: D-channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: D-channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: D-channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)

16 channels configured.

ZT_SPANCONFIG failed on span 2: No such device or address (6)
---

 

The /etc/modprobe.d/blacklist file contains, amongst others, the following 
lines:
--
blacklist hisax
blacklist hisax_fcpcipnp
blacklist 8139cp
blacklist hfc4s8s_l1
--

 

The /etc/zaptel.conf file looks like this:
- zaptel.conf -
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
fxsks=1
fxols=2
fxsks=3
fxols=4

# Span 2-5: Junghans
span=2,1,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
span=5,0,3,ccs,ami

bchan=5,6
dchan=7
bchan=8,9
dchan=10
bchan=11,12
dchan=13
bchan=14,15
dchan=16

# Global data

loadzone= fr
defaultzone = fr
---

 

The  command gives the following:
---
zaptel  182820  8 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm
crc_ccitt 6337  2 zaptel,irda
---

 

The lspci -vv command returns the following info in relation to the 
Junghanns card:

00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller 
[HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST]
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- 
Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- 
SERR- http://lists.digium.com/mailman/listinfo/asterisk-users


   
-
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Paul]

Joe acquisto wrote:

>I have dual posted this to the user and biz lists.
>
>Has anyone ever heard of someone running an Asterisk based system, yet 
>abandoning SugarCRM, and opting to develop their own Visual FoxPro 
>database/CRM?
>
>Please don't dump on me now, this is not my idea, I am just asking for 
>comments, to see if my own initial thoughts are reasonably accurate.
>
>  
>
I'll answer it on the user list. I don't think the idea is developed
enough to discuss on biz.

First - vtiger is available for those who don't like the SugarCRM licensing.

Second -  developing your own CRM is an ambitious undertaking.  You need
good reasons to go in that direction.

Third -  I have enough exposure to  Visual FoxPro to quickly rule it out
as  a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did include things like
access from linux apps using ODBC so I know enough to hate the product.

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RE: [asterisk-users] headsets for linksys/sipura phones?

[Nabeel Jafferali]

Most of the headsets at http://preview.tinyurl.com/38ow27 should work. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Per Jessen
> Sent: April 28, 2007 11:02 AM
> To: asterisk-users@lists.digium.com
> Subject: RE: [asterisk-users] headsets for linksys/sipura phones?
> 
> Nabeel Jafferali wrote:
> 
> > You can look for headsets made for Motorola cell phones. Also, 
> > Plantronics has some compatible models - I can dig up part 
> numbers if 
> > you're interested.
> > 
> 
> Yes, please - Plantronics is in my regular suppliers catalog, 
> but still only with 3.5mm jacks.  If you've got part#s or 
> URLs, that would be very helpful.
> 
> 
> /Per Jessen, Zürich
> 
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[asterisk-users] SugarCRM, NO!, Foxpro, SI?

[Joe acquisto]

I have dual posted this to the user and biz lists.

Has anyone ever heard of someone running an Asterisk based system, yet 
abandoning SugarCRM, and opting to develop their own Visual FoxPro database/CRM?

Please don't dump on me now, this is not my idea, I am just asking for 
comments, to see if my own initial thoughts are reasonably accurate.

joe a.

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RE: [asterisk-users] TDM400P and Junghanns QuadBRI issue

[Henk Dick]

I would check:

 

Cat /proc/zaptel/

 

To make shure that the cards are activated in the order that you programmed
them.

 

 

Henk

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Backup
e-mail
Sent: maandag 30 april 2007 13:26
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TDM400P and Junghanns QuadBRI issue

 

Hi List,

 

I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns
QuadBRI
on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). 
I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16,
libpri-1.2.4 
and asterisk-1.2.17

When it's the time for ztcfg to do its job it complains with 


 

I'm out of ideas what to do to make it to work. Your help is very much
appreciated.

The config files and results from various commands follow.

 

Thanks,


Costa.

---
Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Loopstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Loopstart (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: D-channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: D-channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: D-channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)

16 channels configured.

ZT_SPANCONFIG failed on span 2: No such device or address (6)
---

 

The /etc/modprobe.d/blacklist file contains, amongst others, the following
lines:
--
blacklist hisax
blacklist hisax_fcpcipnp
blacklist 8139cp
blacklist hfc4s8s_l1
--

 

The /etc/zaptel.conf file looks like this:
- zaptel.conf -
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
fxsks=1
fxols=2
fxsks=3
fxols=4

# Span 2-5: Junghans
span=2,1,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
span=5,0,3,ccs,ami

bchan=5,6
dchan=7
bchan=8,9
dchan=10
bchan=11,12
dchan=13
bchan=14,15
dchan=16

# Global data

loadzone= fr
defaultzone = fr
---

 

The  command gives the following:
---
zaptel  182820  8
wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm
crc_ccitt 6337  2 zaptel,irda
---

 

The lspci -vv command returns the following info in relation to the
Junghanns card:

00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller
[HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST]
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort-
SERR- http://us.rd.yahoo.com/evt=48245/*http:/autos.yahoo.com/new_cars.html;_ylc=
X3oDMTE1YW1jcXJ2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDbmV3LWNhcnM->  cars
at Yahoo! Autos. 

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Re: [asterisk-users] don't want call to get answered

[Jeff Davis]

Arun Kumar wrote:
In my * box I've configured two queues and incoming number and whenever 
any one calls those number call comes to my *box and it sends call to my 
agents in queue. but if no agent is available it still answer the call. 
Is there any why when my agents are not available I don't want call to 
get answered. Here is my dialplan:


From:
http://voip-info.org/wiki/index.php?page=Asterisk+config+queues.conf

--
joinempty=strict

"joinempty" set to "strict" will keep incoming callers from being placed
in queues where there are no agents to take calls. The Queue()
application will return, and the dial plan can detemine what to do next.
--

You should also pick up a copy of Asterisk: The Future of Telephony.
pp.326-328 contain information on queues.conf.


exten => ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten => ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten => ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten => ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten => ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten => ,6,Goto(out-of-hours,5003,1)
exten => ,7,Answer()
exten => ,8,Playback(custom/next-avail-advisor)
exten => 
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID}) 


exten => ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten => ,11,Queue(kbsupport,t)
exten => ,12,Hangup()


You may need to put a handler after priority 11. Your current logic will
unconditionally hangup the channel if no agents are in the queue.

--
Jeff Davis
Netsource Consulting

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[asterisk-users] TDM400P and Junghanns QuadBRI issue

[Backup e-mail]

Hi List,
   
  I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns 
QuadBRI
on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). 
I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, 
libpri-1.2.4 
and asterisk-1.2.17
  When it's the time for ztcfg to do its job it complains with 

   
  I'm out of ideas what to do to make it to work. Your help is very much 
appreciated.
  The config files and results from various commands follow.
   
  Thanks,
  
Costa.
  ---
Zaptel Configuration
==
  SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1)
  Channel map:
  Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Loopstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Loopstart (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: D-channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: D-channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: D-channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
  16 channels configured.
  ZT_SPANCONFIG failed on span 2: No such device or address (6)
---
   
  The /etc/modprobe.d/blacklist file contains, amongst others, the following 
lines:
--
blacklist hisax
blacklist hisax_fcpcipnp
blacklist 8139cp
blacklist hfc4s8s_l1
--
   
  The /etc/zaptel.conf file looks like this:
- zaptel.conf -
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
fxsks=1
fxols=2
fxsks=3
fxols=4
  # Span 2-5: Junghans
span=2,1,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
span=5,0,3,ccs,ami
  bchan=5,6
dchan=7
bchan=8,9
dchan=10
bchan=11,12
dchan=13
bchan=14,15
dchan=16
  # Global data
  loadzone= fr
defaultzone = fr
---
   
  The  command gives the following:
---
zaptel  182820  8 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm
crc_ccitt 6337  2 zaptel,irda
---
   
  The lspci -vv command returns the following info in relation to the Junghanns 
card:

00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller 
[HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST]
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- 
Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- 
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[asterisk-users] don't want call to get answered

[Arun Kumar]

In my * box I've configured two queues and incoming number and whenever any
one calls those number call comes to my *box and it sends call to my agents
in queue. but if no agent is available it still answer the call. Is there
any why when my agents are not available I don't want call to get answered.
Here is my dialplan:

exten => ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten => ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten => ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten => ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten => ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten => ,6,Goto(out-of-hours,5003,1)
exten => ,7,Answer()
exten => ,8,Playback(custom/next-avail-advisor)
exten =>
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
exten => ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten => ,11,Queue(kbsupport,t)
exten => ,12,Hangup()



thanks
arun
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[asterisk-users] Priority in ACD

[Mantu Jha]

  
Hi all,

Can someone help me in reslving issue with priority in ACD
I am using Asterisk 1.4 and also ACD but when my agent login using priority 1 
and 2 or 1 and 3 call come to both the priority which is unusual if anyone 
encounter this issue please let me know also help me how to comeout of this 
issue


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Re: [asterisk-users] Poor man's High Availability solution

[Vicente Aguilar]

Hi

We've got a redfone here and it's working great so far, despite all the
TDMoE bad press.

The 4-span version is slightly more expensive than a TE410P, so in the
end it's gonna be a more affordable solution as you'd need two digium
cards (plus maybe the ISDN guard).

The downside is that it doesn't have echo cancellation, so you'll have
to do it via software while some cards do it on hardware (faster,
better, less CPU-intensive...)

El sáb, 28-04-2007 a las 23:22 +0200, Laurent CARON escribió:
> Hi,
> 
> I'm wondering what the best option to obtain a high availability
> asterisk server is.
> 
> I currently use a TE410P (4 x E1) card.
> 
> I'm thinking of 2 different solutions:
> 
> - 2 servers configured with Heartbeat + DRBD (drbd mainly for
> voicemail) and the E1 span plugged to the 2 servers (with a TE410P
> in each server).
> 
> - 2 servers configures with Heartbeat + DRBD with the E1 span hooked to
> an ISDN guard connected to the main server and the backup one.
> 
> Here comes the real question.
> 
> Is it "technically" good to connect an E1 span to 2 cards at the same
> time (with only one accepting the calls).
> 
> Since it is possible with BRI cards, i'm wondering if it could be done
> with PRI.
> 
> Thanks
> 
> Laurent
> 
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-- 
 Vicente Aguilar <[EMAIL PROTECTED]>
 Dpto. de Infraestructuras
 Tlf.: 965 98 71 92

 Recursos en la Red, S.L.U.
 http://www.renr.es

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Re: [asterisk-users] headsets for linksys/sipura phones?

[Per Jessen]

Andrew Joakimsen wrote:

> On 4/27/07, Per Jessen <[EMAIL PROTECTED]> wrote:
> 
>> > Try your local mobile phone supplier.  I used a headset that came
>> > with one of my cell phones, and it worked great w/ my SPA-941.
>>
>> Not a bad idea  - which make was this for?  None of my phones
>> (Ericsson, Nokia) have a 2.5mm socket, they're all
>> special/proprietary.
>>
> 
> The headset for any other mobile will work. And I thought Nokia did
> use 2.5mm but reverse polarity

I have 4-5 different Nokias, none have a 2.5mm jack.  Nothing that even
remotely resembles a jack.  

So far, what I've found is a Plantronics M175 headset.  The local
plantronics website itself is not very informative wrt what kind of
connectors they use. 


/Per Jessen, Zürich

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