RE: [asterisk-users] Asterisk stops responding to SIP/ZAP

[Yuan LIU]

From: "Ken Williams" <[EMAIL PROTECTED]>
Date: Fri, 20 Apr 2007 07:27:05 -0600

About once a week or so my Asterisk box stops responding to all phones.
I can pull up the console, do whatever I want at the CLI but the only
way to get things working again is to restart Asterisk altogether.

I finally cranked verbose & debugging way up (and watched my log files
go from 1mb/day to 100mb/day), but below I believe contains my problem.
The next line is 1.5 minutes later where I restart Asterisk.


As a general troubleshooting procedure, you want to ask yourself if you have 
made any changes before it stopped working.  If not, and especially if you 
can restart and get it working again, I'd suspect some hardware failure. 
(Assuming the problem is reproduceable - I had times when TDM card stopped 
working with no trace of error.)  Try installing on another box.


Yuan Liu


SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in
place here).  Zap/3-1 is a Digium TDM400.

I can't quite figure out where my problem is, is it the initial
exception, is it not getting hung up completely, does it have to do with
the call limit on the SIP channel, perhaps 'no provider found'
statements?

Any help would be appreciated, I have a relatively simple dial-plan, I
can send over relevant bits of it if necessary.

Thanks,
Ken

[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on
channel 3 (index 0)
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on
channel 3
[Apr 19 13:51:13] DEBUG[27722] channel.c: Didn't get a frame from
channel: Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging channels
SIP/701-08ee6120 and Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'Zap/3-1'
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1)
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0,
normal = 12, callwait = -1, thirdcall = -1
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on
channel 3
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3,
with 0 conference users
[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Hungup 'Zap/3-1'
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension
(from-internal,201,2) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
(from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing
[EMAIL PROTECTED]:1] Hangup("SIP/701-08ee6120", "") in new stack
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension
(from-internal,h,1) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
(from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel
'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call SIP/701-08ee6120,
SIP callid [EMAIL PROTECTED])
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for
incoming call
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701' removed
from call limit 6
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel SIP/701
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel SIP/701-08ee6120
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for Zap - 3
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for Zap/3 -
state 0 (Unknown)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701
- state 1 (Not in use)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701
- state 1 (Not in use)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701



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RE: [asterisk-users] Softphone that supports central provisioning?

[Yuan LIU]

From: "Salvatore Giudice" <[EMAIL PROTECTED]>
Date: Sat, 21 Apr 2007 01:46:20 -0400

A complete provisioning system for soft phones could impart some of the 
same

authentication models used for popular IM clients. Imagine a large
enterprise who wants to give out several thousand soft phones to employees
in a turnkey fashion requiring the employee's network credentials to
authenticate at the start of each session. Generally, it is not acceptable
to use employee credentials to perform SIP digest authentication. Employee
credentials are meant for employees, not devices or software that sets up a
session on behalf of an employee.

The solution to this kind of setup is to use a soft phone that can be
downloaded on demand and presents the employee with a simple
username/password/domain login box. In one such system that I worked on, 
the

client would take the credentials from the employee and authenticate via
HTTPS to a simple CGI script that authenticates the credentials against an
Active Directory setup. Once the employee is authenticated, the CGI script
sets a temporary password in a database that is accessible by a radius
server and sends back all the provisioning information including the
employee's office number and the temporary session password via XML in the
HTTPS POST response. The client then logs into the SIP service using the
session credentials.


Thought the OP wanted the name of a soft phone that was capable of using CGI 
or whatever mechanism to pull such provisioning info, or one that could be 
reconfigured on demand (outside of itself).  I'd like to know which one(s), 
too.  Wouldn't imagine pushing user credentials to end points.


Yuan Liu


The employee is required to re-authenticate at the start of each soft phone
session or after a timed interval when the temporary session password is
expired from radius.

The advantages to this kind of setup are:
1.) you don't have employee credentials stored in soft phones
2.) you avoid locking out employee credentials when policy-based password
changes are required because of rapid authentication failures from a SIP
device with stored credentials
3.) no SIP service credentials are stored in the soft phones
4.) in the event that the temporary session password is stolen from a soft
phone installation, it is only good for a short period of time usually
limited to 12 hours
5.) HTTPS is a significantly better provisioning method than TFTP (cough
Cisco...) because it is encrypted and you have the opportunity to validate 
a

cert from the provisioning server to ensure that the soft phone client is
talking directly to the provisioning server. Man in the middle attacks 
suck.

6.) it's a lot easier to change provisioning information for all clients
without requiring employees to download a new soft phone with hardcoded
settings or trying to get employees to implement changes on their phones
manually. For the same reason, it reduces initial setup complexity and also
eliminates the bulk of setup related support calls

We have put together implementations of this kind of system before for
clients. Usually, this kind of scenario is not something we discuss outside
our training classes or at conventions. Generally, this kind of system is
commonly requested by enterprise and government customers when they seek to
extend their phone system to employees for road warrior, pandemic, disaster
recovery, or occasional work at home scenarios.



--
Salvatore Giudice
[EMAIL PROTECTED]

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

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, April 20, 2007 9:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Softphone that supports central provisioning?

On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
> Has anyone found a softphone that supports pulling it's configuration 
from

a
> central server via TFTP/FTP/HTTP, much like hard desk phones use?

Why would you want to do that?

There are well-known and established tools to "provision" (centrally
configure) software running on computers in a entwork. Why should the
soft phones be configured any differently?

What OS do you use on the desktops?



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RE: [asterisk-users] Softphone that supports central provisioning?

[Salvatore Giudice]

If it's a law firm, they are probably using Windows. I believe the initial
post mentioned they were using Counterpath  as well. BTW, if you are using
X-ten, you can script a launcher application which can perform your
provisioning download/authentication and provision the client by setting the
appropriate registry entries. The part that sucks with Counterpath is that
it's difficult to generate the encrypted string they use to store the
password in the registry key. The work around is to generate sample
passwords and capture those from the registry. Use the plain text password
in your sip service and set the client to the encrypted string with the
launcher script.


--
Salvatore Giudice
[EMAIL PROTECTED]

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

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, April 20, 2007 9:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Softphone that supports central provisioning?

On Fri, Apr 20, 2007 at 09:21:54PM -0400, Michelle Dupuis wrote:
> Tzafrir,
> 
> I don't know if you do many large deployments, but this would be a
godsend!
> We did an install for a large law firm with all lawyers wanting softphones
> (eyebeam) on their laptops.  Centrally pushing out the install executable
> was easy, setting up the parameters for each user was time consuming (i.e.
> expensive).  With hard phones, we setup a TFTP server for each phone to
pull
> config on bootup.  We've even built a couple of tools to build config
files
> (text ini files) dynamically from a database.  This has shaved up to 8
hours
> off a large install.
> 
> I think you're confusing installation with configuration.  Without ascii
> config files (or a tool from the mfg to create binary config files from a
> script), each soft device must manually configured.

Can you name a decent Linux soft phone worth its salt for which you
cannot generate such a provisioning system in 1 hour? It would
probably be custom and site-specific. The generic provisioning systems
used for soft phones require way too much trust on the provisioning
server and are lacking on the security side.

-- 
   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] Softphone that supports central provisioning?

[Salvatore Giudice]

I think you are assuming that the company owns the computer that the
employee runs the soft phone on. It's possible that employees will want to
run them from untrusted computers at home, etc. 


--
Salvatore Giudice
[EMAIL PROTECTED]

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

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, April 20, 2007 9:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Softphone that supports central provisioning?

On Fri, Apr 20, 2007 at 09:21:54PM -0400, Michelle Dupuis wrote:
> Tzafrir,
> 
> I don't know if you do many large deployments, but this would be a
godsend!
> We did an install for a large law firm with all lawyers wanting softphones
> (eyebeam) on their laptops.  Centrally pushing out the install executable
> was easy, setting up the parameters for each user was time consuming (i.e.
> expensive).  With hard phones, we setup a TFTP server for each phone to
pull
> config on bootup.  We've even built a couple of tools to build config
files
> (text ini files) dynamically from a database.  This has shaved up to 8
hours
> off a large install.
> 
> I think you're confusing installation with configuration.  Without ascii
> config files (or a tool from the mfg to create binary config files from a
> script), each soft device must manually configured.

I am not. The soft phone is not the only software on that computer that
needs cetral configuration.

How do you configure the networking on those computers? The mail
clients? How do you deploy updates?

-- 
   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] Softphone that supports central provisioning?

[Salvatore Giudice]

I hope to god you didn't put that TFTP server on the open internet. 

--
Salvatore Giudice
[EMAIL PROTECTED]

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

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Friday, April 20, 2007 9:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Softphone that supports central provisioning?

Tzafrir,

I don't know if you do many large deployments, but this would be a godsend!
We did an install for a large law firm with all lawyers wanting softphones
(eyebeam) on their laptops.  Centrally pushing out the install executable
was easy, setting up the parameters for each user was time consuming (i.e.
expensive).  With hard phones, we setup a TFTP server for each phone to pull
config on bootup.  We've even built a couple of tools to build config files
(text ini files) dynamically from a database.  This has shaved up to 8 hours
off a large install.

I think you're confusing installation with configuration.  Without ascii
config files (or a tool from the mfg to create binary config files from a
script), each soft device must manually configured.

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support.  Visit us at
www.generationd.com
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, April 20, 2007 9:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Softphone that supports central provisioning?

On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
> Has anyone found a softphone that supports pulling it's configuration 
> from a central server via TFTP/FTP/HTTP, much like hard desk phones use?

Why would you want to do that?

There are well-known and established tools to "provision" (centrally
configure) software running on computers in a entwork. Why should the soft
phones be configured any differently?

What OS do you use on the desktops?

-- 
   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] Softphone that supports central provisioning?

[Salvatore Giudice]

A complete provisioning system for soft phones could impart some of the same
authentication models used for popular IM clients. Imagine a large
enterprise who wants to give out several thousand soft phones to employees
in a turnkey fashion requiring the employee's network credentials to
authenticate at the start of each session. Generally, it is not acceptable
to use employee credentials to perform SIP digest authentication. Employee
credentials are meant for employees, not devices or software that sets up a
session on behalf of an employee. 

The solution to this kind of setup is to use a soft phone that can be
downloaded on demand and presents the employee with a simple
username/password/domain login box. In one such system that I worked on, the
client would take the credentials from the employee and authenticate via
HTTPS to a simple CGI script that authenticates the credentials against an
Active Directory setup. Once the employee is authenticated, the CGI script
sets a temporary password in a database that is accessible by a radius
server and sends back all the provisioning information including the
employee's office number and the temporary session password via XML in the
HTTPS POST response. The client then logs into the SIP service using the
session credentials.

The employee is required to re-authenticate at the start of each soft phone
session or after a timed interval when the temporary session password is
expired from radius.

The advantages to this kind of setup are:
1.) you don't have employee credentials stored in soft phones
2.) you avoid locking out employee credentials when policy-based password
changes are required because of rapid authentication failures from a SIP
device with stored credentials
3.) no SIP service credentials are stored in the soft phones
4.) in the event that the temporary session password is stolen from a soft
phone installation, it is only good for a short period of time usually
limited to 12 hours
5.) HTTPS is a significantly better provisioning method than TFTP (cough
Cisco...) because it is encrypted and you have the opportunity to validate a
cert from the provisioning server to ensure that the soft phone client is
talking directly to the provisioning server. Man in the middle attacks suck.
6.) it's a lot easier to change provisioning information for all clients
without requiring employees to download a new soft phone with hardcoded
settings or trying to get employees to implement changes on their phones
manually. For the same reason, it reduces initial setup complexity and also
eliminates the bulk of setup related support calls

We have put together implementations of this kind of system before for
clients. Usually, this kind of scenario is not something we discuss outside
our training classes or at conventions. Generally, this kind of system is
commonly requested by enterprise and government customers when they seek to
extend their phone system to employees for road warrior, pandemic, disaster
recovery, or occasional work at home scenarios.



--
Salvatore Giudice
[EMAIL PROTECTED]

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

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, April 20, 2007 9:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Softphone that supports central provisioning?

On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
> Has anyone found a softphone that supports pulling it's configuration from
a
> central server via TFTP/FTP/HTTP, much like hard desk phones use?

Why would you want to do that?

There are well-known and established tools to "provision" (centrally
configure) software running on computers in a entwork. Why should the
soft phones be configured any differently?

What OS do you use on the desktops?

-- 
   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] Failed to authenticate on INVITE

[dave cantera]

ango, 
I have been playing with connecting two * servers... I had to stop but
I do think I had it working...  even with this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
it wasn't as straight forward as I would have liked...  I used a
register on one box and a conf entry on the other.  then I reversed the
config for the other * box

pbx82 = 10.10.15.82
pbx15 = 10.10.15.15

on pbx15

sip.conf
register => sip_pbx15:[EMAIL PROTECTED]
[sip_to_pbx82]
type=user
username=sip_pbx15
accountcode=sip_from_pbx15
secret=1234
context=sip_from_pbx15
host=10.10.15.82
disallow=all
allow=ulaw
allow=alaw
allow=gsm

extensions.conf
[sip_pbx15_to_pbx82]
; dial a pbx82 extension via SIP with 982XXX where XXX is the extension
exten =>
_982XXX,1,Dial(SIP/sip_pbx15:[EMAIL PROTECTED]/${EXTEN:3},20,r)
;exten => _982XXX,1,Dial(SIP/${EXTEN:3},20,r)
exten => _982XXX,n,Playback(connection-failed)
exten => _982XXX,n,Playback(vm-goodbye)
exten => _982XXX,n,Congestion
exten => _982XXX,n,Hangup

on pbx82

extensions.conf
[sip_from_pbx15]
exten => _XXX,1,Wait(1)
exten => _XXX,n,Answer()
exten => _XXX,n,Dial(SIP/${EXTEN},20,,r)
exten => _XXX,n,VoiceMailMain
exten => _XXX,n,Hangup()

[sip_from_pbx15] must be accessible in your inbound or default
context...
I don't think I made any general section changes...

it has been a few weeks since I played with it and I went only one
way... but if it worked one way it should work the other way too by
reverse duplicating the above config on pbx82 and pbx15 respectively.
let me know how you make out...
daveC


Rilawich Ango wrote:
I use realtime.  Both information and extensions are
stored in DB.  It
  
is just a simple setting of the user with dial plan "Dial([EMAIL PROTECTED])".
  
exten => 9003,1,Dial([EMAIL PROTECTED])
  
What I found is the following.
  
  
9002 ---> S1 ---> S2
  
9002 can make request to S1 and S1 forward the request to S2.
  
9002 ---> S1 <--- S2
  
S2 returns the mentioned error message to S1.  (What I guess is 9002
  
only registers in S1 not in S2, so mentioned error message issued by
  
S2).
  
  
It is what I got from the above case.  Do you have such configuration?
  
I have no idea to solve the problem
  
  
On 4/20/07, dave cantera <[EMAIL PROTECTED]> wrote:
  
  ango,

can you provide some sip.conf and extens.conf info?

daveC


Rilawich Ango wrote:

> hi,

>  I have 2 asterisks with the following configuration.

> asterisk server 1 (S1) has an user 9002

> asterisk server 2 (S2) has an user 9003

> Both users can make call to each other without problem.

> Now I add both users to both servers, i.e.

> asterisk server 1 (S1) has users 9002,9003

> asterisk server 2 (S2) has users 9002,9003

> When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa.  Both
processes

> failed to make call with the following error.

> Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802
handle_response_invite:

> Failed to authenticate on INVITE to '"9002"

> ;tag=as2ff0c493'

> Any solution to let them call each others?

> ango

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Interlocking Business Solutions, LLC

856-380-0894 x5000



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-- 
Building Strong Relationships w/ Intelligent Customer Service
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Interlocking Business Solutions, LLC
856-380-0894 x5000




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Re: [asterisk-users] Developing Marketing materials ...

[dave cantera]

robert,
I might be interested depending on cost, message, and quality...
keep me in the loop.
daveC

Robert Augustyn wrote:

Hi,
I am working on developing a professional Marketing Materials for my 
systems.
I plan on using a very good(expensive) company to do that so splitting 
the costs with several people would be nice.

Let me know if you are interested on taking part in it.
robert
 



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 269.5.1/765 - Release Date: 04/17/2007 05:20 
PM
  


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Re: [asterisk-users] Softphone that supports central provisioning?

[Philipp Kempgen]

Tzafrir Cohen wrote:

> On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
>> Has anyone found a softphone that supports pulling it's configuration from a
>> central server via TFTP/FTP/HTTP, much like hard desk phones use?
> 
> Why would you want to do that?

Because you could provision softphones the way you provision hard
phones. Dynamic configuration through HTTP or even SIP messages.
That would really be great.

I think it's a valid question and I've been searching for such
softphones as myself. They should be usable (so most of them fail)
and should work on a real OS (tm). And no Java please :)


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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Re: [asterisk-users] Softphone that supports central provisioning?

[Tzafrir Cohen]

On Fri, Apr 20, 2007 at 09:21:54PM -0400, Michelle Dupuis wrote:
> Tzafrir,
> 
> I don't know if you do many large deployments, but this would be a godsend!
> We did an install for a large law firm with all lawyers wanting softphones
> (eyebeam) on their laptops.  Centrally pushing out the install executable
> was easy, setting up the parameters for each user was time consuming (i.e.
> expensive).  With hard phones, we setup a TFTP server for each phone to pull
> config on bootup.  We've even built a couple of tools to build config files
> (text ini files) dynamically from a database.  This has shaved up to 8 hours
> off a large install.
> 
> I think you're confusing installation with configuration.  Without ascii
> config files (or a tool from the mfg to create binary config files from a
> script), each soft device must manually configured.

Can you name a decent Linux soft phone worth its salt for which you
cannot generate such a provisioning system in 1 hour? It would
probably be custom and site-specific. The generic provisioning systems
used for soft phones require way too much trust on the provisioning
server and are lacking on the security side.

-- 
   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] Softphone that supports central provisioning?

[Tzafrir Cohen]

On Fri, Apr 20, 2007 at 09:21:54PM -0400, Michelle Dupuis wrote:
> Tzafrir,
> 
> I don't know if you do many large deployments, but this would be a godsend!
> We did an install for a large law firm with all lawyers wanting softphones
> (eyebeam) on their laptops.  Centrally pushing out the install executable
> was easy, setting up the parameters for each user was time consuming (i.e.
> expensive).  With hard phones, we setup a TFTP server for each phone to pull
> config on bootup.  We've even built a couple of tools to build config files
> (text ini files) dynamically from a database.  This has shaved up to 8 hours
> off a large install.
> 
> I think you're confusing installation with configuration.  Without ascii
> config files (or a tool from the mfg to create binary config files from a
> script), each soft device must manually configured.

I am not. The soft phone is not the only software on that computer that
needs cetral configuration.

How do you configure the networking on those computers? The mail
clients? How do you deploy updates?

-- 
   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] Softphone that supports central provisioning?

[Michelle Dupuis]

Tzafrir,

I don't know if you do many large deployments, but this would be a godsend!
We did an install for a large law firm with all lawyers wanting softphones
(eyebeam) on their laptops.  Centrally pushing out the install executable
was easy, setting up the parameters for each user was time consuming (i.e.
expensive).  With hard phones, we setup a TFTP server for each phone to pull
config on bootup.  We've even built a couple of tools to build config files
(text ini files) dynamically from a database.  This has shaved up to 8 hours
off a large install.

I think you're confusing installation with configuration.  Without ascii
config files (or a tool from the mfg to create binary config files from a
script), each soft device must manually configured.

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support.  Visit us at
www.generationd.com
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, April 20, 2007 9:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Softphone that supports central provisioning?

On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
> Has anyone found a softphone that supports pulling it's configuration 
> from a central server via TFTP/FTP/HTTP, much like hard desk phones use?

Why would you want to do that?

There are well-known and established tools to "provision" (centrally
configure) software running on computers in a entwork. Why should the soft
phones be configured any differently?

What OS do you use on the desktops?

-- 
   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] ztdummy does not load properly at server startup

[Tzafrir Cohen]

Replying to Theo Band's post. Please don't use HTML mail...

Matthew J. Roth wrote:
> Theo,
>
> Unless things have changed significantly in the newer releases, you
> must load zaptel prior to loading ztdummy.  Additionally, the zaptel
> devices are not created instantly, so after you load zaptel you must
> wait a few seconds before loading ztdummy.  You can perform some sort
> of polling if you want to script this, but a less sophisticated method
> is just to sleep for 10 or 15 seconds between the calls to modprobe.
>
> If your goal is to start Asterisk automatically at boot, some init
> scripts for different distributions are available at
> . 
> I'm using Fedora, so I installed 'rc.redhat.asterisk' with chkconfig
> as follows:
>
> # install -m 755 ./rc.redhat.asterisk /etc/rc.d/init.d/asterisk
> # chkconfig --add asterisk
> # chkconfig --list asterisk
> asterisk0:off   1:off   2:on3:on4:on5:on6:off
The make install already did that for me .
>
>
> Note that I made the following customizations to the script prior to
> installing it:
>
> * I don't want to run safe_asterisk, so I comment out all of the lines
> that reference the SAFE_ASTERISK variable.

Not using safe_asterisk is a good idea.

> * I want to load ztdummy and raise the open file limit, so I add the
> following lines to the start() function immediately prior to the
> 'daemon' statement:
> modprobe zaptel > /dev/null 2> /dev/null
> sleep 15
> modprobe ztdummy > /dev/null 2> /dev/null
> ztcfg > /dev/null 2> /dev/null

As you don't need ztcfg for ztdummy, the only line you need from the
above four lines is:

  modprobe ztdummy

Or use the zaptel init script. If it finds no zaptel timing source it
modprobes ztdummy.

> > ulimit -n 65536 > /dev/null 2> /dev/null
> > * And add the following lines to the stop() function, immediately
> > after the 'RETVAL=$?' line:
> > rmmod ztdummy > /dev/null 2> /dev/null
> > rmmod zaptel > /dev/null 2> /dev/null

With the latest zaptel init script, you can simply use it (once the
modules unloading fixes are in place). But then aghain, there is no harm
whatsoever from not unloading ztdummy at system shutdown.

> Yes it works indeed thanks. It turns out that (for me) the "modprobe
> ztdummy" alone works as well. I do get an error message as shown before,

That's because of the bogus setting in /etc/modprobe.conf that runs
ztcfg needlessly on the time of the load of ztdummy, and because uev is
slightly slow in creating the zaptel devices (the init.d script only
runs ztcfg after wait_fot_zapctl).

> but both ztummy and zaptel get loaded and Meetme() works. The 15 seconds
> delay would add to the boot time

You can avoid all of that sleep (or at least most of it) easily.

-- 
   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] ZT_CHANCONFIG failed on channel 1: No suchdeviceor address

[Tzafrir Cohen]

On Sat, Apr 21, 2007 at 04:28:08AM +1200, CSB wrote:
> >
> > lsmod | grep ^zaptel
> >
> lsmod | grep ^zaptel
> zaptel183076  2 zttranscode,wctdm

Did it identify a card?

rmmod wctdm; modprobe wctdm; dmesg | tail

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] Softphone that supports central provisioning?

[Tzafrir Cohen]

On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
> Has anyone found a softphone that supports pulling it's configuration from a
> central server via TFTP/FTP/HTTP, much like hard desk phones use?

Why would you want to do that?

There are well-known and established tools to "provision" (centrally
configure) software running on computers in a entwork. Why should the
soft phones be configured any differently?

What OS do you use on the desktops?

-- 
   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] Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemailto text translation)

[Matthew Rubenstein]

(This subthread is more appropriate to -users than to -dev, so it is
crossposted only to mark its transition. Please reply on the -user list
only.)

What are the cheapest prices for (humans) transcribing voicemail to
text as a service? The absolute cheapest, regardless of (known) quality
- the quality only has to compete with (cheaper) automated
transcription, which is abysmal quality.


On Wed, 2007-04-04 at 09:25 -0700, [EMAIL PROTECTED]
wrote:
> Date: Wed, 4 Apr 2007 09:25:02 -0700
> From: "Mike Taht" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-dev] Voicemail to text translation
> To: [EMAIL PROTECTED],"Asterisk Developers Mailing
> List"
> <[EMAIL PROTECTED]>
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> On 4/4/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]>
> wrote:
> >
> > Is anybody aware of a way to automate the translation or conversion
> of
> > voice mail files into text ?
> 
> 
> Being that understanding random human speech at 8khz
> 
> I had had a different idea. Merely have a voice mail option "press 4
> to
> transcribe this" - which would take the vmail and ship it to a
> transcription
> service like "transcribr.com". There's a couple companies like that
> that out
> there do transcription - quite well, and cheaply.
> 
> Sent via BlackBerry from T-Mobile
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> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> 
> 
> 
> -- 
> Mike Taht
> PostCards From the Bleeding Edge
> http://the-edge.blogspot.com 
-- 

(C) Matthew Rubenstein

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Re: [asterisk-users] why do I get this message

[Bruce Ferrell]

Alex Balashov wrote:

On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:



set_format: Unable to find a codec translation path from ulaw to g729

Both endpoints are PAP2 set to G711 only



  Where precisely are they so set?

--
Alex Balashov <[EMAIL PROTECTED]>


The are locked to G711u at the line... Using the web interface
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Re: [asterisk-users] VPM450: Not Present

[Kevin P. Fleming]

Chris Miller wrote:
> I've got a system with a TE412P installed under Fedora Core 6 and I
> continue to see this message in the logs. The card most certainly does
> have an EC module installed. The system is suffering from echo problems,
> and I suspect this is no coincidence... I've double checked to ensure
> the module has been inserted correctly. I've not seen any other
> complaints on the lists, etc. about this error message, so I'm running
> out of clues. Same problem under Fedora Core 4. How does one
> confirm/troubleshoot EC card detection?

You should call Digium support; if the module is present and installed
properly and the driver does not see it, there is nothing that can be
done other than to get the card and module replaced. This is not a
software problem.
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Re: [asterisk-users] HPEC audio clipping: IMPORTANT DETAIL

[Kevin P. Fleming]

Stephen Bosch wrote:

> I've confirmed this: hpec-9.00.002 has the clipping problem. hpec-8.20
> definitely doesn't. I've implemented and reverted. The clipping makes
> the phones unusable.

This is most definitely helpful information; I can put the 8.20 files
back on the FTP site (or make them available via Support) if needed.

> What's the word on the patch?

The patch is _NOT_ an attempt to cure this problem, nor did I advertise
it as such. It is a patch to allow us to capture audio streams in a way
that will allow us to debug (and presumably fix) the problem.

I don't know yet which versions of Zaptel it will be merged into, but it
is small enough that it will likely go into both 1.2 and 1.4. The patch
is being reviewed by some developers here at Digium and will then be
merged, possibly as early as tomorrow evening. Once that is done we can
work with one or more of you who are experiencing this problem to get us
audio capture files, and we will supply those to the HPEC code creators
to find out what is going wrong.
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Re: [asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup with log

[Leo Ann Boon]

Just curios, does the CS1000 now support RFC2833? Previously, I know the 
NRS can only support SIP-INFO.


Leo

Jerry Geis wrote:
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming 
calls just fine. However, using outgoing call files the CS1000 is 
hanging up after I answer the call.


I dont know why?

Thanks, for any assistance.

Jerry

my sip.conf entry is:
   [Nortel]
   type=friend
   dtmfmode=rfc2833
   username=X
   disallow=all
   allow=ulaw
   allow=alaw
   context=nortel
   host=XXX
   canreinvite=yes
   qualify=yes
usereqphone=yes


-

Use 'exit' when done

Asterisk 1.4.2, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' 
for details.
This is free software, with components licensed under the GNU General 
Public
License version 2 and other licenses; you are welcome to redistribute 
it under

certain conditions. Type 'core show license' for details.
=
 == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing 
'/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.2 currently running on hfemsrv (pid = 
18420)

hfemsrv*CLI> Verbosity is at least 5

hfemsrv*CLI> sip debug
hfemsrv*CLI> SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future 
release. Please use 'sip set debug' instead.


hfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060:
OPTIONS sip:192.168.45.129 SIP/2.0
Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK2508d83c;rport
From: "asterisk" ;tag=as2cc96e52
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Apr 2007 19:25:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
?
hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
SIP/2.0 200 OK
From: "asterisk";tag=as2cc96e52
To: ;tag=812da8c0-13c4-46277c06-279cd106-42ff
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 


Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
Content-Length: 0


<->
?--- (10 headers 0 lines) ---
?
hfemsrv*CLI> Really destroying SIP dialog 
'[EMAIL PROTECTED]' Method: OPTIONS

?
hfemsrv*CLI>-- Attempting call on SIP/QuadNortel/7113 for 
[EMAIL PROTECTED]:1 (Retry 1)

?
hfemsrv*CLI> Audio is at 161.49.142.250 port 1
?
hfemsrv*CLI> Adding codec 0x4 (ulaw) to SDP
?Adding codec 0x8 (alaw) to SDP
?
hfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK11268a7d;rport
From: "Admin System 34" ;tag=as4e5a553d
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Apr 2007 19:25:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 18420 18420 IN IP4 161.49.142.250
s=session
c=IN IP4 161.49.142.250
t=0 0
m=audio 1 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
?
hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
SIP/2.0 100 Trying
From: "Admin System 34";tag=as4e5a553d
To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
Contact: 
Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 


Content-Length: 0


<->
?--- (11 headers 0 lines) ---
?
hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
SIP/2.0 180 Ringing
From: "Admin System 34";tag=as4e5a553d
To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
Contact: 
 

Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 


Content-Length: 0


<->
?--- (11 headers 0 lines) ---
?
hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
SIP/2.0 200 OK
From: "Admin System 34";tag=as4e5a553d
To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
Supported: 100rel,sipvc,re

Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording

[Gavin Henry]

On 21/04/07, Leo Ann Boon <[EMAIL PROTECTED]> wrote:


Gavin Henry wrote:
> Dear All,
>
> Is it possible to install * in front of a Avaya IP 406 system via a T
> connector E1 tap so it's external to the Avaya system?
>
Voicetronix has an open sourced solution using their OpenPRI in Hi-Z mode.

http://www.voicetronix.com/open-source.htm#logger



Thanks.




Leo

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Re: [asterisk-users] Transfer via CTI

[Leo Ann Boon]

Phil Menico wrote:

I used autodial to allow a user to make a call by clicking on a web
directory and placing a call file into the Asterisk "outgoing"
directory. That works perfectly for me.

What if I want to click on the web directory and transfer my existing
call? Is there a comparable interface? 
  

Use the manager interface.

Leo

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Re: [asterisk-users] Queue problems

[Tim Verscheure]

I'm not familiar with that...

2007/4/20, Darryl Dunkin <[EMAIL PROTECTED]>:

Not only that, are the phones logged into the agents?

The agents are most likely statically assigned but need to be logged
into. This can be confusing. I use AddQueueMember/RemoveQueueMember for
the phones themselves skipping the agents.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bkruse
Sent: Friday, April 20, 2007 15:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue problems

Are your agents logged into the queue?

-brandon

Tim Verscheure wrote:
> Hi,
>
> I've been configuring AsteriskNOW from the GUI but could it be that
> the GUI isn't working properly? because when I make a queue and add a
> few agents, and when I call the queue none of the phones ring. The
> queue is also configured at "Ringall"
>
> I checked the queues.conf file and the settings matched. I also
> noticed that the agents I made in the GUI, that they were not written
> away in agents.conf file, so I've added them there but still no
> results...
>
> any suggestions?
>
> Tim
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Re: [asterisk-users] Queue problems

[Tim Verscheure]

Yes, that's the thing... at first the GUI wouldn't write the agents
into the agents.conf file. I can logon with my agents but when I do a
"show agents" command in CLI he says that none of my agents are logged
in...

...

2007/4/20, bkruse <[EMAIL PROTECTED]>:

Are your agents logged into the queue?

-brandon

Tim Verscheure wrote:
> Hi,
>
> I've been configuring AsteriskNOW from the GUI but could it be that
> the GUI isn't working properly? because when I make a queue and add a
> few agents, and when I call the queue none of the phones ring. The
> queue is also configured at "Ringall"
>
> I checked the queues.conf file and the settings matched. I also
> noticed that the agents I made in the GUI, that they were not written
> away in agents.conf file, so I've added them there but still no
> results...
>
> any suggestions?
>
> Tim
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RE: [asterisk-users] Softphone that supports central provisioning?

[Salvatore Giudice]

I tried to get the SJPhone folks to implement this two years ago. It's one
of the major features missing from the market. You may want to contact
Bluenote networks.  http://www.bluenotenetworks.com/ They have an IP PBX and
a soft phone client. They only sell their products to the enterprise market.
They can do this for you. Their clients and servers are based on the
Microsoft RTC and Radvision stacks.

 

If you have any problems, tell them I referred you.

 

--
Salvatore Giudice
[EMAIL PROTECTED]

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

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Friday, April 20, 2007 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Softphone that supports central provisioning?

 

Has anyone found a softphone that supports pulling it's configuration from a
central server via TFTP/FTP/HTTP, much like hard desk phones use?

I'm looking for something for a call center that I can provision from a
central location by generating config files.  If the phone has "soft keys"
(yes, I know they're all soft - but you know what I mean; programmable
buttons whose function comes from the provisioning system), even better. 

I know idefisk Biz says they'll do this, but it's not in the release
candidate and will make it's debut in the "final" version, which is a little
too much early adoption for my liking.  Other than that, I'm back at
X-Lite/eyeBeam, which stores it's configs in binary files, preventing me
from   I'm open to SIP/IAX, so long as I don't have to jump through hoops to
get it talking to *. 

Thanks for any experience you can share.

-- 
j. 

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Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording

[Leo Ann Boon]

Gavin Henry wrote:

Dear All,

Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?


Voicetronix has an open sourced solution using their OpenPRI in Hi-Z mode.

http://www.voicetronix.com/open-source.htm#logger

Leo

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Re: [asterisk-users] Queue problems

[bkruse]

Darryl: Correct, and this is a common mis-conception among
people that use queues, especially ring-all.


We plan on having huntgroups in replacement for queues
that dont actually want responsible agents to login and whatnot,
but rather to just be called when one comes in.

-brandon


Darryl Dunkin wrote:

Not only that, are the phones logged into the agents?

The agents are most likely statically assigned but need to be logged
into. This can be confusing. I use AddQueueMember/RemoveQueueMember for
the phones themselves skipping the agents.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bkruse
Sent: Friday, April 20, 2007 15:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue problems

Are your agents logged into the queue?

-brandon

Tim Verscheure wrote:
  

Hi,

I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at "Ringall"

I checked the queues.conf file and the settings matched. I also
noticed that the agents I made in the GUI, that they were not written
away in agents.conf file, so I've added them there but still no
results...

any suggestions?

Tim
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RE: [asterisk-users] Queue problems

[Darryl Dunkin]

Not only that, are the phones logged into the agents?

The agents are most likely statically assigned but need to be logged
into. This can be confusing. I use AddQueueMember/RemoveQueueMember for
the phones themselves skipping the agents.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bkruse
Sent: Friday, April 20, 2007 15:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue problems

Are your agents logged into the queue?

-brandon

Tim Verscheure wrote:
> Hi,
>
> I've been configuring AsteriskNOW from the GUI but could it be that
> the GUI isn't working properly? because when I make a queue and add a
> few agents, and when I call the queue none of the phones ring. The
> queue is also configured at "Ringall"
>
> I checked the queues.conf file and the settings matched. I also
> noticed that the agents I made in the GUI, that they were not written
> away in agents.conf file, so I've added them there but still no
> results...
>
> any suggestions?
>
> Tim
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RE: [asterisk-users] [OT] OMG Verizon is terrible

[Chris Bagnall]

> Ah, back in the old days our government privatized the state monopoly
> (BT) intact (attitudes and all).

For all their customer service failings at the call centre end, once you do get 
an engineer, it's very rare to get a bad one.

Many of them have been in the job for decades and, at a guess, probably have 
civil service pensions to look forward to from the pre-privatisation days, so 
they're far more willing to go the extra mile to be helpful than the 
paid-by-the-hour folks with very little training some companies send out.

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] why do I get this message

[bkruse]

Sip Debug,

But I can tell you now that one of them is requesting g729, or, asterisk
has g729 set for one of its codecs in sip.conf and needs to translate it.

grep -r "g729" /etc/asterisk/*

Alex Balashov wrote:

On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:



set_format: Unable to find a codec translation path from ulaw to g729

Both endpoints are PAP2 set to G711 only


  Where precisely are they so set?

--
Alex Balashov <[EMAIL PROTECTED]>
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Re: [asterisk-users] Queue problems

[bkruse]

Are your agents logged into the queue?

-brandon

Tim Verscheure wrote:

Hi,

I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at "Ringall"

I checked the queues.conf file and the settings matched. I also
noticed that the agents I made in the GUI, that they were not written
away in agents.conf file, so I've added them there but still no
results...

any suggestions?

Tim
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Re: [asterisk-users] why do I get this message

[Alex Balashov]

On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:



set_format: Unable to find a codec translation path from ulaw to g729

Both endpoints are PAP2 set to G711 only


  Where precisely are they so set?

--
Alex Balashov <[EMAIL PROTECTED]>
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Re: [asterisk-users] 3rd T1 of quad card won't change signaling

[Jay Wilton]

--- Barton Fisher <[EMAIL PROTECTED]> wrote:

> Looks like:
> 
> amaflags=billing
> switchtype=national
> 
> is being carry-over from prior PRI.. (All PRI stuff) Try
> moving below 
> before the first PRI?


Thanks all, I tried the:
T1 as port 1 and then the PRI as ports 2 and 3 but zap
dumped again.  I tried to blank the  switchtype=  , but zap
didn't like that.

span=3,0,0,d4,ami did not work with the original setup.

ztcfg -vv gives no errors at all and shows correct
signaling per port.  I think this only reads the
zaptel.conf and the error is occuring while parsing the
zapata.conf.

After every change, I remove/readd all zap modules and then
ztcfg -vv.  I will try the ztcfg -vvf tonight.

The customer asked the faxman and said a PRI would be
better  anyway.  Thanks again, I'll let you know.

JJ

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[asterisk-users] why do I get this message

[Bruce Ferrell]

set_format: Unable to find a codec translation path from ulaw to g729

Both endpoints are PAP2 set to G711 only

I have 1.2.17 on Suse 10.1
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Re: [asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...

[Arturo Ochoa]

Noah Miller wrote:

> I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
> and it also has the echo canceller...
> I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
> 2.6.9-34.0.2.EL
> I'm using Polycom's  501 with the SIP 1.6.2.0041
>
> The problem is when someone dials to or from the PSTN through the
> TDM2400, the voice quality is crappy...Instead of hearing:
>
> Hello, this is John
>
> You hear..
>
> He  o, th  s   J hn
>
> I already tried with the fxotune utility, also using G711 or G729,
> dealing with the gains... but I can't see the light...

This is a bug in the 9.00-002 HPEC echo canceller.

I have no idea when a patch will be available.


I don't think this is the HPEC issue.  I don't think Zaptel 1.2.12
supported HPEC.  This must be the hardware echo can on the TDM2400.

Arturo, can you post your zapata.conf and sip.conf?

Also, I don't think this is your problem, but you may want to consider
upgrading to Asterisk 1.2.17 and Zaptel 1.2.16.  There have been many
bug and security fixes since Asterisk 1.2.13.


- Noah
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While trying to fix the problem, yesterday I spoke to Digium, they 
logged to the computer and they upgraded Zaptel to 1.2.16 with the HPEC 
support.
They enabled the MARK2 software echo cancellation by default. Then we 
made a few tests but nothing changed.
Then I enabled the aggressive mode of this echo canceller, then I ran 
the fxotune -i, and last I ran the fxotune -s to load the parameters...


With this changes, the voice quality seems to be better, at least you 
can hear almost all the words, but still have low quality...


This is the zapata.conf

[channels]

language=en
context=from-zaptel
signalling=fxs_ks
;rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
;usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
;echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

group=1


--
Ing. Arturo Ochoa N
Network Administrator
Electrosystems,

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RE: [asterisk-users] Asterisk & PiX devices

[Don E. Wisdom]

Sorry I should clarify.  I need to pass sip traffic thru the pix to the
asterisk server.  (from sip phones at my house and wherever else I might
be) The pix has 7.2.2 os
--Don




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Mendoza
Sent: Friday, April 20, 2007 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk & PiX devices

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

Don E. Wisdom wrote:
>
> Hi All,
>
> Im just getting started in the asterisk world and im wondering if 
> anyone can point me in the right direction towards getting asterisk 
> working from my house to my asterisk server in my colocation facility.
>
> Thanks
>
> --Don
>
>  
>
>
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!SIG:4629331f169582021920165!

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[asterisk-users] Developing Marketing materials ...

[Robert Augustyn]

Hi,
I am working on developing a professional Marketing Materials for my
systems.
I plan on using a very good(expensive) company to do that so splitting the
costs with several people would be nice.
Let me know if you are interested on taking part in it.
robert
 
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Re: [asterisk-users] Big trouble with zap lines

[Bruce Reeves]

Do you have fxs modules or fxo modules? PSTN connects to fxo, but the
signaling is fxs like you have.


On 4/20/07, Ricardo Melendez <[EMAIL PROTECTED]> wrote:


Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap
channels like this

In zaptel.conf
fxsks=1-4
fxsks=5-8
fxsks=9-12
loadzone=us
defaultzone=us

in zapata.conf
in channels section
context=incoming
signalling=fxs_ks
channel => 1-4
channel => 5-8
channel => 9-12

when i run ztcfg -vv show 12 channels correctly configured
whe i run "zap show channels" in asterisk console this show 12 channels
correctly configured
when i call to a zap channel like this "Dial(zap/1/somenumber,15,r)" in
the console appear that asterisk is dialing trought this channel to this
somenumber but in the line the call
never go out nor in, the same happens when dial from outside, the line is
ringing until the normal timeout.

the PSTN lines used work normally whit normal hardphones (PSTN)

zaptel, asterisk, zttool and ztcfg all never send any error message.

What  could be the problem??

Could be a damaged wildcard
My card is wctdm2400p with 12 fxs ports in 3 modules

thanks in advance


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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] Big trouble with zap lines

[Bruce Reeves]

Do you have fxs modules or fxo modules? PSTN connects to fxo, but the
signaling is fxs like you have.

On 4/20/07, Ricardo Melendez <[EMAIL PROTECTED]> wrote:


Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap
channels like this

In zaptel.conf
fxsks=1-4
fxsks=5-8
fxsks=9-12
loadzone=us
defaultzone=us

in zapata.conf
in channels section
context=incoming
signalling=fxs_ks
channel => 1-4
channel => 5-8
channel => 9-12

when i run ztcfg -vv show 12 channels correctly configured
whe i run "zap show channels" in asterisk console this show 12 channels
correctly configured
when i call to a zap channel like this "Dial(zap/1/somenumber,15,r)" in
the console appear that asterisk is dialing trought this channel to this
somenumber but in the line the call
never go out nor in, the same happens when dial from outside, the line is
ringing until the normal timeout.

the PSTN lines used work normally whit normal hardphones (PSTN)

zaptel, asterisk, zttool and ztcfg all never send any error message.

What  could be the problem??

Could be a damaged wildcard
My card is wctdm2400p with 12 fxs ports in 3 modules

thanks in advance


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--
Bruce Reeves
Nortex Networks
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[asterisk-users] Queue problems

[Tim Verscheure]

Hi,

I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at "Ringall"

I checked the queues.conf file and the settings matched. I also
noticed that the agents I made in the GUI, that they were not written
away in agents.conf file, so I've added them there but still no
results...

any suggestions?

Tim
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Re: [asterisk-users] Asterisk & PiX devices

[Jorge Mendoza]

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

Don E. Wisdom wrote:


Hi All,

Im just getting started in the asterisk world and im wondering if 
anyone can point me in the right direction towards getting asterisk 
working from my house to my asterisk server in my colocation facility.


Thanks

--Don

 




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RE: [asterisk-users] CallerID Auth

[Yuan LIU]

From: "Arun Kumar" <[EMAIL PROTECTED]>
Date: Fri, 20 Apr 2007 17:58:10 +0400

Hi,

in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.


Just detect that a call is international, then branch out.  e.g., if 011 is 
the prefix required for international,


[outbound]
exten => _011.,1,Dial(Local/${EXTEN}/international)
exten => _X.,1,Dial(ZAP/g1/${EXTEN})

[international]
exten => _X.,1,GotoIf(${DBEXISTS(international/${CALLERID(NUMBER)})}?:deny)
exten => _X.,n,Dial(ZAP/g1/${EXTEN})
exten => _X.,n,Hangup; just in case
exten => _X.,n(deny),Playback(not-a-valid-number&try-again)
exten => _X.,n,DISA(nopassword,outbound)

This is assuming AstDB contains a family international that includes 
extensions/ID's allowed.  Hope this helps.


Yuan Liu


thanks

arun



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[asterisk-users] Big trouble with zap lines

[Ricardo Melendez]

Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels 
like this

In zaptel.conf
fxsks=1-4
fxsks=5-8
fxsks=9-12
loadzone=us
defaultzone=us

in zapata.conf
in channels section
context=incoming
signalling=fxs_ks
channel => 1-4
channel => 5-8
channel => 9-12

when i run ztcfg -vv show 12 channels correctly configured
whe i run "zap show channels" in asterisk console this show 12 channels 
correctly configured
when i call to a zap channel like this "Dial(zap/1/somenumber,15,r)" in the 
console appear that asterisk is dialing trought this channel to this somenumber 
but in the line the call
never go out nor in, the same happens when dial from outside, the line is 
ringing until the normal timeout.

the PSTN lines used work normally whit normal hardphones (PSTN)

zaptel, asterisk, zttool and ztcfg all never send any error message.

What  could be the problem??

Could be a damaged wildcard
My card is wctdm2400p with 12 fxs ports in 3 modules

thanks in advance


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[asterisk-users] FW: Asterisk & PiX devices

[Don E. Wisdom]

I forgot to add the hardware.   Im using Gentoo Linux & a Pix 515 

Thanks

--Don

 

 



From: Don E. Wisdom 
Sent: Friday, April 20, 2007 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Asterisk & PiX devices

 

Hi All,

Im just getting started in the asterisk world and im wondering if anyone
can point me in the right direction towards getting asterisk working
from my house to my asterisk server in my colocation facility.

Thanks

--Don

 

 

 

 

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[asterisk-users] Asterisk & PiX devices

[Don E. Wisdom]

Hi All,

Im just getting started in the asterisk world and im wondering if anyone
can point me in the right direction towards getting asterisk working
from my house to my asterisk server in my colocation facility.

Thanks

--Don

 

 

 

 

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Re: [asterisk-users] Softphone that supports central provisioning?

[Yuan LIU]

From: "Steve Davies" <[EMAIL PROTECTED]>
Date: Fri, 20 Apr 2007 18:26:57 +0100

On 4/20/07, James FitzGibbon <[EMAIL PROTECTED]> wrote:

On 4/20/07, Olivier <[EMAIL PROTECTED]> wrote:

> Are you sure eyeBeam config are binary ?
> I thought it was just the case for XLite.

Having looked into it further, you're right.  For some inexplicable reason
it's not putting the files where the manual says they should be - instead 
of
a directory called "eyeBeam n.n" they're in a folder called 'RegNow 
Basic',
but the .CPS files there are indeed in XML rather than binary format.  
When

I last looked, I suspect I assumed that eyeBeam stored it's configs in the
X-Lite directory and was thus looking at the configs for the free version
that were no longer being accessed.


I went around this loop with CounterPath a couple of months back. It
seems that their idea of provisioning revolves around customising the
software before selling it, so that it is locking the end-user into
using "your" (the seller's) SIP server.

They had trouble understanding that the user just paid money for this
software, which they want to be provisioned by a server on their own
network, and they do not support this. I gave up at this stage, but


That's because mainstream service providers only want a branded client that 
indeed locks users in.  Unless a reasonably powerful commercial entity (or 
even freelance org) exerts pressure, individual users and small companies 
can't do much.


Does a Web deployed client such as JAIN SIP applet count?

Yuan Liu


perhaps if more people apply pressure, it will become possible to
extend their current (quite useable) provisioning interface, but have
a user-configurable setting to determine where the configuration is
fetched from. At present the configuration server setting is fixed at
compile-time by CounterPath.

Regards,
Steve



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[asterisk-users] C7960 TFTP [Slightly off-topic]

[Steve Finkelstein]

Hi all,

This is slightly off-topic, but I was hoping to be able to receive some
insight as I'm sure plenty of experts with c7960's exist on this mailing
list.

I'm attempting to upgrade from SIP 8.3 -> 8.6 on a C7960G that I
inherited. I have my TFTP setup and unfiltered. The phone is doing TFTP
over the internet as I'm telecommuting today, but I've placed it on the
dmz to avoid any firewall headaches.

Here's what a packet capture looks like:

tcpdump -vv -Xnni eth1 -s 1000 port 69
tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size
1000 bytes



16:30:25.649429 IP (tos 0x0, ttl  51, id 1612, offset 0, flags [none],
proto: UDP (17), length: 56) 1.2.3.4.50770 > 64.90.184.96.69: [no cksum]
 28 RRQ "SIP00187330C526.cnf" octet
0x:  4500 0038 064c  3311 b2a8 44c1 9145  E..8.L..3...D..E
0x0010:  405a b860 c652 0045 0024  0001 5349  @Z.`.R.E.$SI
0x0020:  5030 3031 3837  3043 3532 362e 636e  P00187330C526.cn
0x0030:  6600 6f63 7465 7400  f.octet.
16:30:41.667380 IP (tos 0x0, ttl  51, id 1617, offset 0, flags [none],
proto: UDP (17), length: 56) 1.2.3.4.50771 > 1.2.3.5.69: [no cksum]  28
RRQ "SIP00187330C526.cnf" octet
0x:  4500 0038 0651  3311 b2a3 44c1 9145  E..8.Q..3...D..E
0x0010:  405a b860 c653 0045 0024  0001 5349  @Z.`.S.E.$SI
0x0020:  5030 3031 3837  3043 3532 362e 636e  P00187330C526.cn
0x0030:  6600 6f63 7465 7400  f.octet.
16:31:15.596217 IP (tos 0x0, ttl  51, id 1606, offset 0, flags [none],
proto: UDP (17), length: 51) 1.2.3.4.50757 > 1.2.3.5.69: [no cksum]  23
RRQ "SIPDefault.cnf" octet
0x:  4500 0033 0646  3311 b2b3 44c1 9145  E..3.F..3...D..E
0x0010:  405a b860 c645 0045 001f  0001 5349  @Z.`.E.E..SI
0x0020:  5044 6566 6175 6c74 2e63 6e66 006f 6374  PDefault.cnf.oct
0x0030:  6574 00  et.
16:31:31.621286 IP (tos 0x0, ttl  51, id 1611, offset 0, flags [none],
proto: UDP (17), length: 58) 1.2.3.4.50758 > 1.2.3.5.69: [no cksum]  30
RRQ "./SIP00187330C526.cnf" octet
0x:  4500 003a 064b  3311 b2a7 44c1 9145  E..:.K..3...D..E
0x0010:  405a b860 c646 0045 0026  0001 2e2f  @Z.`.F.E.&./
0x0020:  5349 5030 3031 3837  3043 3532 362e  SIP00187330C526.
0x0030:  636e 6600 6f63 7465 7400 cnf.octet.
16:31:47.652531 IP (tos 0x0, ttl  51, id 1617, offset 0, flags [none],
proto: UDP (17), length: 58) 1.2.3.4.50759 > 1.2.3.5.69: [no cksum]  30
RRQ "./SIP00187330C526.cnf" octet
0x:  4500 003a 0651  3311 b2a1 44c1 9145  E..:.Q..3...D..E
0x0010:  405a b860 c647 0045 0026  0001 2e2f  @Z.`.G.E.&./
0x0020:  5349 5030 3031 3837  3043 3532 362e  SIP00187330C526.
0x0030:  636e 6600 6f63 7465 7400 cnf.octet.
16:32:03.679382 IP (tos 0x0, ttl  51, id 1623, offset 0, flags [none],
proto: UDP (17), length: 58) 1.2.3.4.50760 > 1.2.3.5: [no cksum]  30 RRQ
"./SIP00187330C526.cnf" octet
0x:  4500 003a 0657  3311 b29b 44c1 9145  E..:.W..3...D..E
0x0010:  405a b860 c648 0045 0026  0001 2e2f  @Z.`.H.E.&./
0x0020:  5349 5030 3031 3837  3043 3532 362e  SIP00187330C526.
0x0030:  636e 6600 6f63 7465 7400 cnf.octet.

---

The phone also displays TFTP SIP00187330C526.cnf on its LCD, however it
does not appear to be retrieving binaries from the tftpserver.

tftproot # cat OS79XX.TXT
P003-08-6-00

-rwxr-xr-x 1 root   root 129824 Dec 12 16:54 P003-08-6-00.bin
-rwxr-xr-x 1 root   root 130228 Dec 12 17:21 P003-08-6-00.sbn
-rwxr-xr-x 1 root   root459 Dec 12 17:40 P0S3-08-6-00.loads
-rwxr-xr-x 1 root   root 753560 Dec 12 17:20 P0S3-08-6-00.sb2
-rw-r--r-- 1 root   root 681556 Jan 10 14:55 P0S3-08-6-00.zip
-rw-rw-rw- 1 root   root779 Apr 20 14:21 SIP00187330C526.cnf
-rw-rw-rw- 1 root   root   4658 Apr 20 16:29 SIPDefault.cnf
-rw-rw-rw- 1 nobody nobody 11675652 Mar 28 16:18 c2600-entbase-mz.123-22.bin
-rw-rw-r-- 1 nobody nobody  7735532 Jan 16 15:54 c2600-i-mz.123-21.bin
-rw-rw-rw- 1 nobody nobody 11100664 Jan 16 16:24 c2600-ik9s-mz.122-27.bin
-rw-rw-r-- 1 nobody nobody  8450865 Feb 27 23:35
c3560-advipservicesk9-mz.122-35.SE1.bin
-rw-rw-rw- 1 nobody nobody 2726 Jan 16 16:23 cmeinternetlink-confg
-rw-r--r-- 1 root   root223 Apr 20 14:13 dialplan.xml
-rw-rw-rw- 1 nobody nobody 1413 Jan 16 16:10 helfant-confg
-rw-r--r-- 1 root   root779 Apr 20 14:11 xmlDefault.CNF.XML

Thanks for any suggestions all,

- sf
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Re: [asterisk-users] Polycom SIP Phones On LAN can't register without WAN (Internet) Access

[Noah Miller]

Hi Shawn -


We have several Polycom 500/501/601's on both a LAN and at employee homes.
The problem we are having is if our internet connection goes down the Local
LAN phones loose their connection to the Asterisk Server.
I've checked everything I can think of but can't figure out why its
happening.
I believe since the Asterisk Box is on the LAN and the phones are on the
same LAN then it shouldn't need internet to function.

The closest I have narrowed this down is to the DNS area. I found that if I
block access to our ISP's DNS that the phones aren't able to register with
asterisk.

This baffles me because the phone has the LAN address for the Asterisk
server so I don't know why it's doing DNS lookups.


Hmm.  Well, you've got me.  I don't know why it would be doing that,
it certainly shouldn't be.  You might try a newer version of the SIP
firmware or the 3.2.2 bootrom.

If it still happens with the latest bootrom/firmware, you could do a
packet trace on the phone.  Is it doing DNS queries?  If so, I'd call
your Polycom reseller and have them take this up with Polycom (support
requests are supposed to go through the reseller).  Actually, in any
case, I'd take it up with your Polycom reseller.


- Noah
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[asterisk-users] 7921G running linux

[Zachary Whitley]

I was just watching the informational video on cisco's web site about
the 7921G and they guy mentions that the phone is running Linux. Anyone
know if they've released the source code?


This page confirms that the phone is running Linux

http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0900aecd80601788.shtml

The phone doesn't support sipyet ;)

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Re: [asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...

[Noah Miller]

> I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
> and it also has the echo canceller...
> I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
> 2.6.9-34.0.2.EL
> I'm using Polycom's  501 with the SIP 1.6.2.0041
>
> The problem is when someone dials to or from the PSTN through the
> TDM2400, the voice quality is crappy...Instead of hearing:
>
> Hello, this is John
>
> You hear..
>
> He  o, th  s   J hn
>
> I already tried with the fxotune utility, also using G711 or G729,
> dealing with the gains... but I can't see the light...

This is a bug in the 9.00-002 HPEC echo canceller.

I have no idea when a patch will be available.


I don't think this is the HPEC issue.  I don't think Zaptel 1.2.12
supported HPEC.  This must be the hardware echo can on the TDM2400.

Arturo, can you post your zapata.conf and sip.conf?

Also, I don't think this is your problem, but you may want to consider
upgrading to Asterisk 1.2.17 and Zaptel 1.2.16.  There have been many
bug and security fixes since Asterisk 1.2.13.


- Noah
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Re: [asterisk-users] Polycom Phones

[Noah Miller]

Hi Wiley -


Can anyone tell me which config file tells the phone what file to load as 
bootrom.ld?

Or is this hardcoded in the phone?


Yup, it's hardcoded.  I believe this is the way it works: If there's a
bootrom.ld on your configuration server, and it is newer than the one
on the phone, the phone will load it.  Otherwise, it will use what is
already on the phone.  There's no option to change the name or
anything.



I just got a IP501 but I have a bunch of IP500s…

Will the bootrom (2.6.2) work OK with both the IP500 and 501?


You bet.


- Noah
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[asterisk-users] STUN

[kodorn]

Hi guys,

I'm trying to implement STUN support in *, is there anyone here which 
have any experience in something like that?

I've got the STUND and I'll try to buld a patch or something for sip.

Any ideas or existing implementation would be nice. I know openpbx have it.
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[asterisk-users] VPM450: Not Present

[Chris Miller]

I've got a system with a TE412P installed under Fedora Core 6 and I continue to 
see this message in the logs. The card most certainly does have an EC module 
installed. The system is suffering from echo problems, and I suspect this is no 
coincidence... I've double checked to ensure the module has been inserted 
correctly. I've not seen any other complaints on the lists, etc. about this 
error message, so I'm running out of clues. Same problem under Fedora Core 4. 
How does one confirm/troubleshoot EC card detection?

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Re: [asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...

[Stephen Bosch]

Arturo Ochoa wrote:
> Hi List...
> 
> I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
> and it also has the echo canceller...
> I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
> 2.6.9-34.0.2.EL
> I'm using Polycom's  501 with the SIP 1.6.2.0041
> 
> The problem is when someone dials to or from the PSTN through the
> TDM2400, the voice quality is crappy...Instead of hearing:
> 
> Hello, this is John
> 
> You hear..
> 
> He  o, th  s   J hn
> 
> I already tried with the fxotune utility, also using G711 or G729,
> dealing with the gains... but I can't see the light...

This is a bug in the 9.00-002 HPEC echo canceller.

I have no idea when a patch will be available.

-Stephen-

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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Stephen Bosch]

Tim Panton wrote:
> Ah, back in the old days our government privatized the state monopoly
> (BT) intact (attitudes and all).
> As one of the conditions they had to deliver within 6 weeks of order.
> 
> So I ordered  a data line to my house (ok a bit obscure in those days, but
> I needed it). 6 weeks roll past, nothing happens. I call BT and ask why
> they haven't installed my line within 6 weeks of order. The guy gently
> explains
> that it is 6 weeks from order being accepted, and they haven't accepted
> mine yet!
> When are you going to accept it ? - About 5 weeks from when we plan to
> fit it!
> 
> Hey, at least he was an honest bloke in a twisted system.

I find the best approach is to ignore what's in the publicly distributed
marketing material. If you have a doubt about something that a company
representative is telling you, ask for it to be confirmed in writing. If
they waffle, you know they're having you on.

Next, assume you are in the jungle, and there is no civilisation.
Recruit everybody in the chain as a friend and accomplice. If you have
to deal with the provider often, this is worth the initial effort.

People tend to like food :)

-Stephen-
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Re: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article

[Stephen Bosch]

Per Jessen wrote:
> Remco Post wrote:
> 
>> Hans Witvliet wrote:
>>
>>> The only obstacles currently, are the ISP's.
>> Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well
>> as an ipv4 address.
> 
> Not around here (Zurich, Switzerland) they won't.  I think there is one
> single provider with IPV6 as an option.  And the other ones are
> perfectly decent providers too.  Like I said, when the low-cost DSL
> routers/modems do not yet support IPV6, why should the provider?

Not here, either. The best you can do is a tunnel host.

-Stephen-
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Re: [asterisk-users] HPEC audio clipping: IMPORTANT DETAIL

[Stephen Bosch]

Hi, everybody:

Stephen Bosch wrote:
> Kevin P. Fleming wrote:
>> Eric "ManxPower" Wieling wrote:
>>
>>> Any updates on this?
>> The code is done and initially tested; it is being reviewed internally
>> and should be available on Friday or Monday.
> 
> Under what circumstances would this clipping be present? Is this patch
> going to be recommended for anybody using HPEC?

Guess what *I* noticed today?

We've been using the HPEC for about a month now and hadn't had this
clipping problem. Today I added two licences. In the process, I noticed
that the HPEC version had been updated to 9.x. We'd been using 8.2.

Since I'm going through the process of adding the licence, I thought I'd
try updating the HPEC. The moral of that story: if it ain't broke, don't
fix it.

I've confirmed this: hpec-9.00.002 has the clipping problem. hpec-8.20
definitely doesn't. I've implemented and reverted. The clipping makes
the phones unusable.

I just count my lucky stars that I kept the old archive, or I'd be up
the creek right now.

What's the word on the patch?

-Stephen-

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Re: [asterisk-users] How can I improve call quality?

[Tim Panton]

Try turning the jitterbuffer off, I found that often the endpoints can
do better on their own.


On 20 Apr 2007, at 19:01, Adrian Marsh wrote:


Hi All,

I've a single 1.2.17 Asterisk system.  Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have  
ever
used 50% max bandwidth).  We've no E1/T1 links, everything is IP  
based.


My boss complains that many of the calls he holds with others has a  
bad

quality.  He also says that its not just him.

My iax.conf file has:

disallow=all
allow=ulaw
allow=alaw
bandwidth=high
jitterbuffer=yes
dropcount=2
maxjitterbuffer=1000
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
tos=lowdelay
autokill=yes

He complains of broken audio, muffled audio, and says compared to  
Skype

its very poor, particularly during conference calls (zaptel meetme).
Most of these would be SIP based within our server though, rather than
IAX/PSTN based (X-lite/SJphone).


Obviously I can't do much about the far end IP connections/Mobiles  
etc,

but what can I do to tweak/improve the call quality on the A*k box
itself?

The CPU stays at a constant 10% usage, mainly due to a few other
monitoring apps on there (with these turned off, its < 2%, but  
still the

same issues).


Also - are there any useful stats/logs that I can examine to "see" the
quality of calls?

Thanks,

Adrian
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Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Stephen Bosch]

Andrew Latham wrote:
> I make a habit of just buying hamburgers and stopping by the CO or hut
> where I see the vans.  I tell them that I am buying favors with food
> and they like it..   Its a lot of work but it helps...

This is by far the most effective way of getting something done with a
telco. And as for it being a lot of work? No more work than wasting
hours, days, or weeks waiting for a problem to get fixed properly or a

Remember -- in a lot of ILECs, the technicians are still union. More
often than not, they are none to happy with their corporate overlords
either. It is sensible and not too difficult to turn them into friends.

Cheers,

-Stephen-
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Re: [asterisk-users] How can I improve call quality?

[Ed W]

Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth).  


Remember in computer terms this means that you used 100% of the 
connection, 50% of the time  Your voice will loose out against the 
big data packets and spoil the voice quality big time


Ed W
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Re: [asterisk-users] How can I improve call quality?

[Ed W]

Hi


Also - are there any useful stats/logs that I can examine to "see" the
quality of calls?
  


You didn't mention that you have any QOS on your router, so we can 
basically guarantee that your problem is the internet connection.


Remember that all the research on networking has been how to saturate a 
single connection and download as fast as possible, so when some spod 
hits a website and reads a web page then he grabs basically the whole 
connection for a short space of time.  During that time your voip 
packets tend to loose out and get delayed - the jitter buffer does some 
stuff to try and compensate, but ultimately it will loose


Add some kind of priorisation to the T1 line and your quality should go 
up dramatically


Check first using something like testmyvoip.com to get an idea of your 
situation (stress the internet by opening up lots of simultaneous 
downloads during the test)


Cheap fix is to get a separate DSL line and run the voice over that...

Ed W


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[asterisk-users] How can I improve call quality?

[Adrian Marsh]

Hi All,

I've a single 1.2.17 Asterisk system.  Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth).  We've no E1/T1 links, everything is IP based.

My boss complains that many of the calls he holds with others has a bad
quality.  He also says that its not just him.

My iax.conf file has:  

disallow=all
allow=ulaw
allow=alaw
bandwidth=high
jitterbuffer=yes
dropcount=2
maxjitterbuffer=1000
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
tos=lowdelay
autokill=yes

He complains of broken audio, muffled audio, and says compared to Skype
its very poor, particularly during conference calls (zaptel meetme).
Most of these would be SIP based within our server though, rather than
IAX/PSTN based (X-lite/SJphone).


Obviously I can't do much about the far end IP connections/Mobiles etc,
but what can I do to tweak/improve the call quality on the A*k box
itself?

The CPU stays at a constant 10% usage, mainly due to a few other
monitoring apps on there (with these turned off, its < 2%, but still the
same issues).


Also - are there any useful stats/logs that I can examine to "see" the
quality of calls?

Thanks,

Adrian
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Re: [asterisk-users] G.729 & Voicemail

[Robert Lister]

> transcode out of .gsm?) I am not sure what parts of the system are 
> enabled/disabled without the licence.

This mentions voicemail g729 in pass-thru mode. I'm not sure if it works as 
I've never tried it, but it may be worth a try...

http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru


-- 
Robert Lister   -   London Internet Exchange-  http://www.linx.net/
[EMAIL PROTECTED]   -   tel: +44 (0)20 7645 3510-  RL786-RIPE
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Re: [asterisk-users] G.729 & Voicemail

[Robert Lister]

On Fri, Apr 20, 2007 at 02:59:28PM +0200, Michael Landin Hostbaek wrote:
> List, 
> 
> I have some cisco phones (7940) and asterisk 1.4 running nicely.. 
> Communication
> between the phones is G.729, and my sip.conf looks like this:
> 
> disallow=all; First disallow all codecs
> allow=g729  ; 
> allow=gsm
> allow=ulaw
> allow=alaw
> 
> However, I cannot call voicemail - I get the following error:
> [Apr 20 14:58:31] WARNING[87184]: channel.c:2816 set_format: Unable to
> find a codec translation path from g729 to gsm
> 
> Shouldn't it switch to gsm automatically?

Cisco 79XX phones only support ulaw, alaw or g729, not gsm.

Asterisk only supports g.729 protocol in passthrough mode without the 
licence (i.e. It can set up a session between two licenced g.729 endpoints 
to talk to each other, but cannot get into the media path itself.)

The voicemail system is presumably trying to transcode from g.729 to gsm and 
you haven't got the licence for that. (Maybe you can get hold of/convert the 
sounds in the g729 format for the voicemail system, then it may not have to 
transcode out of .gsm?) I am not sure what parts of the system are 
enabled/disabled without the licence.

http://www.voip-info.org/wiki-Asterisk+G.729+Licensing

> I cannot purchase g729 licenses, as FreeBSD is not yet supported (with
> asterisk 1.4)

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Re: [asterisk-users] Softphone that supports central provisioning?

[Steve Davies]

On 4/20/07, James FitzGibbon <[EMAIL PROTECTED]> wrote:

On 4/20/07, Olivier <[EMAIL PROTECTED]> wrote:

>
>
> Are you sure eyeBeam config are binary ?
> I thought it was just the case for XLite.

Having looked into it further, you're right.  For some inexplicable reason
it's not putting the files where the manual says they should be - instead of
a directory called "eyeBeam n.n" they're in a folder called 'RegNow Basic',
but the .CPS files there are indeed in XML rather than binary format.  When
I last looked, I suspect I assumed that eyeBeam stored it's configs in the
X-Lite directory and was thus looking at the configs for the free version
that were no longer being accessed.



I went around this loop with CounterPath a couple of months back. It
seems that their idea of provisioning revolves around customising the
software before selling it, so that it is locking the end-user into
using "your" (the seller's) SIP server.

They had trouble understanding that the user just paid money for this
software, which they want to be provisioned by a server on their own
network, and they do not support this. I gave up at this stage, but
perhaps if more people apply pressure, it will become possible to
extend their current (quite useable) provisioning interface, but have
a user-configurable setting to determine where the configuration is
fetched from. At present the configuration server setting is fixed at
compile-time by CounterPath.

Regards,
Steve
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[asterisk-users] Polycom Phones

[Wiley Siler]

 

Can anyone tell me which config file tells the phone what file to load
as bootrom.ld?

Or is this hardcoded in the phone?  I just got a IP501 but I have a
bunch of IP500s...

Will the bootrom (2.6.2) work OK with both the IP500 and 501?

 

Thanks!

 

Wiley E. Siler
Director of Information Technology
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
mailto:[EMAIL PROTECTED]
www.education2020.com   

 

 

 

Helping students on a mission. Graduation and beyond.

 

<>
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Tim Panton]

On 20 Apr 2007, at 17:10, Kenneth Padgett wrote:

I once had to oversee Verizon install a PRI line in Manhattan.  I  
live

2.5 hours away, but we made the appointment, and I was there, but the
Verizon tech never showed.  I made another appointment, and it
happened again, and again, and again.  I don't even remember how many
times it finally took, but it was ridiculous.  The techs were even
lying and saying they came and there was no one there to let them in.
They seem to have gotten better in recent years, but they own all the
physical lines, and they know it.


Back in the old days, our government de-monoplized monopolys. What
happened to the good old days when we could just split apart big
companies for fun and better competition? Blasted FTC for approving
all these telcom mergers.


Ah, back in the old days our government privatized the state monopoly
(BT) intact (attitudes and all).
As one of the conditions they had to deliver within 6 weeks of order.

So I ordered  a data line to my house (ok a bit obscure in those  
days, but

I needed it). 6 weeks roll past, nothing happens. I call BT and ask why
they haven't installed my line within 6 weeks of order. The guy  
gently explains
that it is 6 weeks from order being accepted, and they haven't  
accepted mine yet!
When are you going to accept it ? - About 5 weeks from when we plan  
to fit it!


Hey, at least he was an honest bloke in a twisted system.
Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Softphone that supports central provisioning?

[James FitzGibbon]

On 4/20/07, Olivier <[EMAIL PROTECTED]> wrote:


Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.



Having looked into it further, you're right.  For some inexplicable reason
it's not putting the files where the manual says they should be - instead of
a directory called "eyeBeam n.n" they're in a folder called 'RegNow Basic',
but the .CPS files there are indeed in XML rather than binary format.  When
I last looked, I suspect I assumed that eyeBeam stored it's configs in the
X-Lite directory and was thus looking at the configs for the free version
that were no longer being accessed.

That does help a little for provisioning, as I can at least generate the
configs and then place them someone central, but actually getting them to
the phone is still kludgey.  Since they are in the Local Settings folder,
they can't be made part of a roaming profile.  I've tried moving the
CounterPath directory from "\Documents and Settings\username\Local
Settings\Application Data" to "\Documents and Settings\username\Application
Data", but the phone never references the configs held there.  Right now
(with X-Lite) i'm configuring each phone manually, then zipping up the
configs and storing them in a location named for the windows username.  On
login, the zipfile is fetched and unzipped to the right location.  Inelegant
to be sure, but it works.  XML just saves me having to do the configuration
manually.

In any case, this is now going down an OT path - I'll take it up with
CounterPath on their forums.  Thanks for the pointer.
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Re: [asterisk-users] Why duoble digits must be so fast to activate features?

[Leonardo Kamache (Gmail)]

Hi Mauro;

Try to add featuredigittimeout => 1500 at features.conf in the [global] section.






On 4/20/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:

On Fri, 20 Apr 2007, Mauro Zanin wrote:

> Hi everybody,
>
> I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm
> using Trixbox..).
> I must be as fast as a flash to press *2 and do an attended transfer. If I
> wait only a tenth of a second nothing happens.
> I think it is an issue. I have seen the source code and found nothing bad.
> Is this a known issue?

Change it in features.conf.

Gordon
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Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No suchdeviceor address

[CSB]

 lsmod | grep ^zaptel


lsmod | grep ^zaptel
zaptel183076  2 zttranscode,wctdm

Cameron
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Re: [asterisk-users] Softphone that supports central provisioning?

[Olivier]

2007/4/20, James FitzGibbon <[EMAIL PROTECTED]>:


Other than that, I'm back at X-Lite/eyeBeam, which stores it's configs in
binary files, preventing me  ...

--
j.



James,

Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.

Regards
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Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording

[Gavin Henry]

On 20/04/07, David Gomillion <[EMAIL PROTECTED]> wrote:

I did this with a Nortel MICS a few years ago. No problem.

The dialplan was something like:

[incoming]
exten => _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten => _X.,2,Monitor(filename)
exten => _X.,3,Dial(Zap/G2/${EXTEN})

[outgoing]
exten => _X.,1,setvar(filename) ;  If you want to record outgoing calls
exten => _X.,2,Monitor(filename); use these two lines, otherwise, just skip
them
exten => _X.,3,Dial(Zap/G1/${EXTEN})

Obviously, this isn't production code, but you should get the idea. If
you're in a 2-party area, you probably need to make your employees sign a
disclosure, and play a sound file to your callers to warn them that the call
is/may be recorded. While it will waste space, I recommend starting the
recording before the file is played. That way, if you're ever challenged,
you'd have something to back up your position that the caller knew. Add the
signed disclosure, and you may be OK.

Of course, I am no lawyer. And you probably ought to talk to one before you
do this. We did, and he had some helpful pointers on what to include in the
disclosure.

There are some areas that will require you to play an annoying beep to
callers. We didn't have to do that, so I'm not sure of the best way to go
about it.


Thanks for this. Given me some ideas. I think our solution has to be
non-evasive, i.e. in case the recording box goes down, the main pbx
works :-)



Good luck,
David

On 4/20/07, Gavin Henry <[EMAIL PROTECTED]> wrote:
>
> Dear All,
>
> Is it possible to install * in front of a Avaya IP 406 system via a T
> connector E1 tap so it's external to the Avaya system?
>
> We would like to record upto 60 channels (2 * ISDN30e). This may increase
> later.
>
> Also, could the calls go into the cdr for retrieval/browsing later?
>
> What hardware/server would you recommend?
>
> Thanks.
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Kenneth Padgett]

I once had to oversee Verizon install a PRI line in Manhattan.  I live
2.5 hours away, but we made the appointment, and I was there, but the
Verizon tech never showed.  I made another appointment, and it
happened again, and again, and again.  I don't even remember how many
times it finally took, but it was ridiculous.  The techs were even
lying and saying they came and there was no one there to let them in.
They seem to have gotten better in recent years, but they own all the
physical lines, and they know it.


Back in the old days, our government de-monoplized monopolys. What
happened to the good old days when we could just split apart big
companies for fun and better competition? Blasted FTC for approving
all these telcom mergers.

Comcast is no better than Verizon. Took them three visits to get my
new home serviced, and I'm still limping along at 3mb / 256k (they
can't tell me why, I'm provisioned for 6mb). Now they're charging me
for DVR service I didn't subscribe too. They're just idiots too.

The only successful way I've found to deal with large unmanaged
companies is to keep calling in trouble tickets, get new techs, and
you'll eventually get one that knows what they're doing.

I personally hate Verizon so much that I've elimitated all my lines
with them, from cell, to home, to business. I refuse to deal with them
and recommend alternative (and less expensive) solutions to all my
customers. It seems to work great, and the only one loosing is
Verizon. They won't have any money left to throw their weight around
with if all their customers leave.

-Kenneth
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Lee Jenkins]

Alan Bunch wrote:
Remember the big lie with Verizon is not "The tech will be there at 
noon".  It is "Your FOC (Firm Order Commit) date is xxx"  The dates they 
give are neither Firm or Committed.  Just ask them.


As long as you remember, they are the Phone Company and you are just the 
customer. no body will be disapointed.


Do I sound a lil grumppy. 5 and half week for  fiber buildout quote.  
Yup, Im grumppy.


alabun

Lee Jenkins wrote:


Steve Totaro wrote:


They are all terrible in their own way.  Don't you have someone you can
delegate the Verizon babysitting responsibility to?  I would consider
sales calls a little more important than being a babysitter.

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


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Wednesday, April 18, 2007 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] [OT] OMG Verizon is terrible


Had an appointment for these schmoes to come out and install another
line.  Was supposed to be 8-12.  Its now 6PM and not even call.


Missed


3 sales calls waiting on these jerks.

No wonder customers were jumping ship to Vonage.

--




Not on that day, I was the only one available unfortunately.



I just got off the phone with Verizon and had them turn off all POTS 
lines except one.  Between the humming noise they seem to not be able to 
fix and CID problems with additional lines we just ordered, I think I've 
had enough of verizon over the last 3 days.


I'll keep the one line for when voip lines are down and shed the 
frustrations...


It'd be nice if T1's were more reasonably priced or FIOS was here.

--

Warm Regards,

Lee



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Re: [asterisk-users] ztdummy does not load properly at server startup

[Matthew J. Roth]

Theo,

I'm glad my reply was helpful to you.  The responses pointed out that 
it's time for me to update my procedures and documentation, so I'm 
benefiting as well.  My thanks go out to everyone who participated in 
this thread.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Call queue problem

[Tim Verscheure]

Hi,

I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at "Ringall"

I checked the queues.conf file and the settings matched. I also
noticed that the agents I made in the GUI, that they were not written
away in agents.conf file, so I've added them there but still no
results...

any suggestions?


Tim
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Re: [asterisk-users] adding second TDM400P card causes echo cancellation to fail for all Zap channels

[Stephen Bosch]

Stephen Bosch wrote:
> Hi, folks:
> 
> Yesterday I added a second TDM400P card to a working, echo-free server
> running HPEC.
> 
> Today, I'm getting these messages:
> 
>> Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to 
>> enable echo cancellation on channel 3
> 
> along with complaints of severe echo. The channels have all been tuned
> using 'fxotune -s' and the echo numbers are all under 2%. I guess that
> Asterisk can't do echo cancellation on these channels.
> 
> Why would I be getting this?

Maybe I would be getting this because I didn't have the zaphpec_enable
in my system startup :)

So -- for future reference, in case anybody else makes this mistake:

zaphpec_enable has to run at boot time, after the zap modules are loaded.

I'll go back to my corner now.

-Stephen-
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Re: [asterisk-users] pci 2.2 - pci-e x16

[Carlos Chavez]

On Fri, 2007-04-20 at 15:37 +0300, asterisk wrote:
> Hi,
> 
> Does anyone know if it is possible to plug a tdm400p pci digium card 
> into an pci-e 16x slot ?
> Is there a possibility to work?
> I have a sun fire x2100 which doesn't have pci slots.
> Does Digium make pci-e cards?
> 
Can you insert a square peg in a round hole?

The only company that has announced PCI-Express products at this moment
is Sangoma.  Rhino has also stated that they will deliver all their
products in a PCI-Ex option.

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] Softphone that supports central provisioning?

[James FitzGibbon]

Has anyone found a softphone that supports pulling it's configuration from a
central server via TFTP/FTP/HTTP, much like hard desk phones use?

I'm looking for something for a call center that I can provision from a
central location by generating config files.  If the phone has "soft keys"
(yes, I know they're all soft - but you know what I mean; programmable
buttons whose function comes from the provisioning system), even better.

I know idefisk Biz says they'll do this, but it's not in the release
candidate and will make it's debut in the "final" version, which is a little
too much early adoption for my liking.  Other than that, I'm back at
X-Lite/eyeBeam, which stores it's configs in binary files, preventing me
from   I'm open to SIP/IAX, so long as I don't have to jump through hoops to
get it talking to *.

Thanks for any experience you can share.

--
j.
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Bruce Reeves]

This could go on forever, I mean take your pick Verizon, At&t, Bell South
any of them. Same story "We are the phone company, who else can you call?".
We have time and again seen it take weeks to get the order documents
created, not the actual order, just the paperwork to create the order. I
personally take great joy in finding anyway not to deal with them. The only
way I see them changing their ways is by losing enough customers. I think
Verizon is learning that lesson, but their response is not to compete and
satisfy the customers, but to put the competition in a strangle hold with
patents that are vague and broad. Ultimately I think Verizon will suffer
from the court decisions more then anyone else, the true nature of their
leadership is not to satisfy the customer.

Bruce Reeves
Nortex Networks
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Re: [asterisk-users] gxp2000 expansion module blf leds not working

[James FitzGibbon]

On 4/19/07, Zoilo Gomez <[EMAIL PROTECTED]> wrote:


Am I the only one using the GXP2000 expansion module?



I'm using one but I'm not terribly happy with it.  With firmware 1.1.3.1 the
phone wouldn't boot, and with 1.1.3.2 having the buttons configured for BLF
caused complete lockups on the phone requiring a power cycle.  It doesn't
reset mind you - the display looks fine until you realize that you haven't
received a call for 10 minutes.

GS tech support suggested disabling BLFs, so until an firmware fix is
available it's a sidecar of 56 speed dials to me.  Granted, it doesn't lock
up anymore either.  I'd love to downgrade to 1.1.2.x, but GS's firmware
isn't capable of doing that and the phone shipped with 1.1.3.1, so I'm kind
of stuck where I am right now.

As to your specific question, I am not sure what would cause this behaviour,
especially if on the phone that the sidecar is attached to the BLFs work.  I
assume that 'show hints' shows state for all your monitored extensions and
'sip show subscriptions' shows that the phone is actually subscribing to the
extensions you have assigned to each multifunction key.

If your problem isn't there, my gut says bad hardware.

I am also looking at the Aastra 57i with the 560 module, but I haven't
gotten one in to play with yet.  I've got a few 480i CTs and they are
performing well, both from a provisioning and usability perspective.  The
ability to do all changes via provisioning is nice, as right now I can
auto-provision changes to the GXP sidecar, but I still have to print up a
replacement button label and walk it over the the phone.  One concern is
whether I have 20 or less most frequently called numbers; if you have more
than 20 your operator will be page-shifting a lot (the 560 has 20 soft keys
and 3 fixed "change page" buttons).

Hope that helps.

--
j.
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Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording

[David Gomillion]

I did this with a Nortel MICS a few years ago. No problem.

The dialplan was something like:

[incoming]
exten => _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten => _X.,2,Monitor(filename)
exten => _X.,3,Dial(Zap/G2/${EXTEN})

[outgoing]
exten => _X.,1,setvar(filename) ;  If you want to record outgoing calls
exten => _X.,2,Monitor(filename); use these two lines, otherwise, just skip
them
exten => _X.,3,Dial(Zap/G1/${EXTEN})

Obviously, this isn't production code, but you should get the idea. If
you're in a 2-party area, you probably need to make your employees sign a
disclosure, and play a sound file to your callers to warn them that the call
is/may be recorded. While it will waste space, I recommend starting the
recording before the file is played. That way, if you're ever challenged,
you'd have something to back up your position that the caller knew. Add the
signed disclosure, and you may be OK.

Of course, I am no lawyer. And you probably ought to talk to one before you
do this. We did, and he had some helpful pointers on what to include in the
disclosure.

There are some areas that will require you to play an annoying beep to
callers. We didn't have to do that, so I'm not sure of the best way to go
about it.

Good luck,
David

On 4/20/07, Gavin Henry <[EMAIL PROTECTED]> wrote:


Dear All,

Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?

We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.

Also, could the calls go into the cdr for retrieval/browsing later?

What hardware/server would you recommend?

Thanks.
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Re: [asterisk-users] Transfer via CTI

[Time Bandit]

Any ideas on this?

Closest thing that comes to mind is FOP : http://www.asternic.org/

hth
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[asterisk-users] adding second TDM400P card causes echo cancellation to fail for all Zap channels

[Stephen Bosch]

Hi, folks:

Yesterday I added a second TDM400P card to a working, echo-free server
running HPEC.

Today, I'm getting these messages:

> Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to enable 
> echo cancellation on channel 3

along with complaints of severe echo. The channels have all been tuned
using 'fxotune -s' and the echo numbers are all under 2%. I guess that
Asterisk can't do echo cancellation on these channels.

Why would I be getting this?

Thanks,

-Stephen-
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[asterisk-users] iaxComm problems

[José Hugo Pérez Casanova]

Hi Folks.

I have installed two sip phones and two PCs in a network. The later with 
iaxComm. Calls are made between the sip phones and between a sip phone and a 
PC.

When calling from one PC to the other the iaxComm shows ??? in the status 
column and the call can't be answered.

The same goes if anyone calls from the outside and tries to reach a PC.

Any ideas?

Regards.


-- 
IEE José Hugo Pérez Casanova
Profesor Investigador

Departamento de Ingeniería Electrónica
Instituto Tecnológico de Veracruz
M.A. de Quevedo #2779, colonia Formando Hogar
Veracruz, Ver. Mexico
Tel/Fax: (52) 229-938-8104
http://electronica.itver.edu.mx
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Re: [asterisk-users] Why duoble digits must be so fast to activate features?

[Gordon Henderson]

On Fri, 20 Apr 2007, Mauro Zanin wrote:


Hi everybody,

I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm 
using Trixbox..).
I must be as fast as a flash to press *2 and do an attended transfer. If I 
wait only a tenth of a second nothing happens.

I think it is an issue. I have seen the source code and found nothing bad.
Is this a known issue?


Change it in features.conf.

Gordon
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[asterisk-users] Pika boards - anyone are using it?

[Peter Aterisk]

Hi,

I`m looking for boards to use with Asterisk and as I already was used Pika
boards few years ago (in a Windows IVR application), I found that they have
new options to Asterisk. It will be nice if I can see some opinions from
here before go ahead on it.

Thanks in advance!


Peter

(*) I`d put this message in biz list but I realize that it seems to be more
related with user list then I resent it here. Sorry about inconvenience.
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Lee Jenkins]

Steve Totaro wrote:

They are all terrible in their own way.  Don't you have someone you can
delegate the Verizon babysitting responsibility to?  I would consider
sales calls a little more important than being a babysitter.

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


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Wednesday, April 18, 2007 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] [OT] OMG Verizon is terrible


Had an appointment for these schmoes to come out and install another
line.  Was supposed to be 8-12.  Its now 6PM and not even call.

Missed

3 sales calls waiting on these jerks.

No wonder customers were jumping ship to Vonage.

--


Not on that day, I was the only one available unfortunately.


--

Warm Regards,

Lee



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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Andrew Latham]

I make a habit of just buying hamburgers and stopping by the CO or hut
where I see the vans.  I tell them that I am buying favors with food
and they like it..   Its a lot of work but it helps


On 4/20/07, Steve Totaro <[EMAIL PROTECTED]> wrote:

Noah Miller wrote:
>> > Had an appointment for these schmoes to come out and install another
>> > line.  Was supposed to be 8-12.  Its now 6PM and not even call.
>> Missed
>> > 3 sales calls waiting on these jerks.
>> >
>> > No wonder customers were jumping ship to Vonage.
>
> I once had to oversee Verizon install a PRI line in Manhattan.  I live
> 2.5 hours away, but we made the appointment, and I was there, but the
> Verizon tech never showed.  I made another appointment, and it
> happened again, and again, and again.  I don't even remember how many
> times it finally took, but it was ridiculous.  The techs were even
> lying and saying they came and there was no one there to let them in.
> They seem to have gotten better in recent years, but they own all the
> physical lines, and they know it.
>
>
> - Noah
>
>
> On 4/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> They are all terrible in their own way.  Don't you have someone you can
>> delegate the Verizon babysitting responsibility to?  I would consider
>> sales calls a little more important than being a babysitter.
>>
>> Thanks,
>> Steve Totaro
>> http://www.asteriskhelpdesk.com
>> KB3OPB
>>
>>
>> > -Original Message-
>> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> > [EMAIL PROTECTED] On Behalf Of Lee Jenkins
>> > Sent: Wednesday, April 18, 2007 6:07 PM
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > Subject: [asterisk-users] [OT] OMG Verizon is terrible
>> >
>> >
>> > Had an appointment for these schmoes to come out and install another
>> > line.  Was supposed to be 8-12.  Its now 6PM and not even call.
>> Missed
>> > 3 sales calls waiting on these jerks.
>> >
>> > No wonder customers were jumping ship to Vonage.
>> >
>> > --
>> >
>> > Warm Regards,
>> >
>> > Lee
>>


Thats funny.  My last turn up with Verizon was a Multilink with three
T1s.  Three DIFFERENT techs showed up at about the same time, each with
a different work order.

Thanks,
Steve
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Steve Totaro]

Noah Miller wrote:

> Had an appointment for these schmoes to come out and install another
> line.  Was supposed to be 8-12.  Its now 6PM and not even call.
Missed
> 3 sales calls waiting on these jerks.
>
> No wonder customers were jumping ship to Vonage.


I once had to oversee Verizon install a PRI line in Manhattan.  I live
2.5 hours away, but we made the appointment, and I was there, but the
Verizon tech never showed.  I made another appointment, and it
happened again, and again, and again.  I don't even remember how many
times it finally took, but it was ridiculous.  The techs were even
lying and saying they came and there was no one there to let them in.
They seem to have gotten better in recent years, but they own all the
physical lines, and they know it.


- Noah


On 4/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:

They are all terrible in their own way.  Don't you have someone you can
delegate the Verizon babysitting responsibility to?  I would consider
sales calls a little more important than being a babysitter.

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


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Lee Jenkins
> Sent: Wednesday, April 18, 2007 6:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] [OT] OMG Verizon is terrible
>
>
> Had an appointment for these schmoes to come out and install another
> line.  Was supposed to be 8-12.  Its now 6PM and not even call.
Missed
> 3 sales calls waiting on these jerks.
>
> No wonder customers were jumping ship to Vonage.
>
> --
>
> Warm Regards,
>
> Lee




Thats funny.  My last turn up with Verizon was a Multilink with three 
T1s.  Three DIFFERENT techs showed up at about the same time, each with 
a different work order.


Thanks,
Steve
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[asterisk-users] Polycom not picking up phone transferred phone call.

[Lee Jenkins]

Hi all,

I'm having a problem with a polycom 301 not picking up a ZAP call. 
Below is the CLI output of the call.  I have:


TDM400 with 2 FXO lines
Asterisk 1.2.14
Polycom 301

When I dial the first ZAP line, I choose an extension that rings the 
polycom, polycom rings and I can pick it up and the call is bridged.


When I call my second zap line, the polycom rings, but I cannot pickup 
the call either by hitting the "Answer" button or by picking up the 
handset.


The only difference that I can see is that the second line is not 
sending CID information and I get the 2 NOTICE lines outputted in the 
CLI output below.


Thanks for any help,

Lee


-- Starting simple switch on 'Zap/2-1'
Apr 20 10:22:03 NOTICE[5258]: chan_zap.c:6072 ss_thread: Got event 18 
(Ring Begin)...
Apr 20 10:22:05 NOTICE[5258]: chan_zap.c:6072 ss_thread: Got event 2 
(Ring/Answered)...

-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Ringing("Zap/2-1", "") in new stack
-- Executing SetMusicOnHold("Zap/2-1", "default") in new stack
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing Goto("Zap/2-1", "check_time|s|1") in new stack
-- Goto (check_time,s,1)
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Set("Zap/2-1", "LAST_MENU_REACHED=check_time") in new 
stack
-- Executing Set("Zap/2-1", 
"FAIL_MENU=error_invalid|TIMEOUT_MENU=error_timeout") in new stack
-- Executing GotoIfTime("Zap/2-1", 
"09:00-17:00|mon-fri|*|*|?main_menu|s|1") in new stack

-- Goto (main_menu,s,1)
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Set("Zap/2-1", "LAST_MENU_REACHED=main_menu") in new stack
-- Executing Macro("Zap/2-1", "ProcessCaller") in new stack
-- Executing Set("Zap/2-1", "DTCONN=callers") in new stack
-- Executing AGI("Zap/2-1", "dtfb|GET|CallerLevel|"SELECT 
caller_level FROM callers WHERE caller_phone = ''"") in new stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/dtfb
-- AGI Script dtfb completed, returning 0
-- Executing AGI("Zap/2-1", "dtfb|GET|CallerName|"SELECT 
caller_name FROM callers WHERE caller_phone = ''"") in new stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/dtfb
-- AGI Script dtfb completed, returning 0
-- Executing NoOp("Zap/2-1", "Caller Level is ") in new stack
-- Executing NoOp("Zap/2-1", "Caller Name is ") in new stack
-- Executing GotoIf("Zap/2-1", "0?lee_voice_followme|s|1") in new stack
-- Executing BackGround("Zap/2-1", "custom/attendant") in new stack
-- Playing 'custom/attendant' (language 'en')
  == CDR updated on Zap/2-1
-- Executing Dial("Zap/2-1", "SIP/111|29|tm") in new stack
-- Called 111
-- Started music on hold, class 'default', on channel 'Zap/2-1'
-- SIP/111-083b2bf8 is ringing
-- Stopped music on hold on Zap/2-1
  == Spawn extension (main_menu, 111, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-- Got SIP response 400 "Bad Request" back from 192.168.1.105
--

Warm Regards,

Lee



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RE: [asterisk-users] Passive E1 Pri Tap for Voice Recording

[John Treble]

Gavin,

Call Endance and ask them about their "Lawful Call Intercept" solution(s)
using their DAG TDM E1 cards on Linux (Endance.com).  

Cheers. 


John Treble
Ottawa, Ontario, Canada


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Gavin Henry
> Sent: April 20, 2007 9:07 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Passive E1 Pri Tap for Voice Recording
> 
> Dear All,
> 
> Is it possible to install * in front of a Avaya IP 406 system via a T
> connector E1 tap so it's external to the Avaya system?
> 
> We would like to record upto 60 channels (2 * ISDN30e). This may increase
> later.
> 
> Also, could the calls go into the cdr for retrieval/browsing later?
> 
> What hardware/server would you recommend?
> 
> Thanks.
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Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording

[Gavin Henry]

On 20/04/07, John Treble <[EMAIL PROTECTED]> wrote:



Gavin,

Call Endance and ask them about their "Lawful Call Intercept" solution(s)
using their DAG TDM E1 cards on Linux (Endance.com).


Thanks, will have a look.



Cheers.


John Treble
Ottawa, Ontario, Canada


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Gavin Henry
> Sent: April 20, 2007 9:07 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Passive E1 Pri Tap for Voice Recording
>
> Dear All,
>
> Is it possible to install * in front of a Avaya IP 406 system via a T
> connector E1 tap so it's external to the Avaya system?
>
> We would like to record upto 60 channels (2 * ISDN30e). This may increase
> later.
>
> Also, could the calls go into the cdr for retrieval/browsing later?
>
> What hardware/server would you recommend?
>
> Thanks.
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[asterisk-users] Why duoble digits must be so fast to activate features?

[Mauro Zanin]

Hi everybody,

I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm 
using Trixbox..).
I must be as fast as a flash to press *2 and do an attended transfer. If I 
wait only a tenth of a second nothing happens.

I think it is an issue. I have seen the source code and found nothing bad.
Is this a known issue?

Many thanks
Best regards

Mauro

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.com/


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[asterisk-users] Agents.conf feature replication using addqueuemember

[Jordan Novak]

I (have to) would like to move my agents out of agents.conf in
preparation for the deprecation of agentcallback login. Everyone I have
spoken to is upset about this but the functionality can be accomplished
in the dialplan and that is fine by me. I do have an issue with losing
the features contained in the agents.conf though. I have to have the
ackcall-yes working. All of my agents login on home phones and
cellphones. When a call get presented to them there is a possibilty that
their voicemail will answer before the queue timeout is reached. I fear
that may connect callers to the voicemail when it answers. We currently
get around this using the ackcall-yes which will wait for the caller to
press pound, which a voicemail will not do and therefore the call will
be put back in the queue. Other features that are important are the
recording(which can be done on the dialplan side) and the update cdr.
Both are important as the Monitor/mixmonitor will not name the file to
be assoicated with the agent and only agent related calls. IE: if you
have to look up the recording based on extension you will get personal
calls as well as queue calls. Updatecdr is invalueable for the same
reasons in the call detail records. 
Is anyone aware of a way to accomplish these things? Are any efforts
being made to replicate the missing features? I can deal with the cdr
and recordings, but the ackcall=yes is a show stopper!!!
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Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

[Moises Silva]

Thanks a lot for the fix Humberto.

On 4/18/07, Humberto Figuera <[EMAIL PROTECTED]> wrote:

Hi Moises,

the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:

"This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered."

here the patch for chan_unicall.c ;p

--- chan_unicall.c.orig 2007-04-18 03:32:17.0 -0400
+++ chan_unicall.c  2007-04-18 03:32:26.0 -0400
@@ -2485,7 +2485,7 @@
 }
 while (x < 3);

-if ( ( tmp = ast_channel_alloc(0, state, 0, 0, chan_name) ) == NULL)
+if ( ( tmp = ast_channel_alloc(0, state, 0, 0, i->accountcode,
i->exten, i->context, i->amaflags, chan_name) ) == NULL)
 {
 ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
 return  NULL;

--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA  37AD 3364 01D1 74CA 0603
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--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Alan Bunch]

Remember the big lie with Verizon is not "The tech will be there at 
noon".  It is "Your FOC (Firm Order Commit) date is xxx"  The dates they 
give are neither Firm or Committed.  Just ask them.


As long as you remember, they are the Phone Company and you are just the 
customer. no body will be disapointed.


Do I sound a lil grumppy. 5 and half week for  fiber buildout quote.  
Yup, Im grumppy.


alabun

Lee Jenkins wrote:


Steve Totaro wrote:


They are all terrible in their own way.  Don't you have someone you can
delegate the Verizon babysitting responsibility to?  I would consider
sales calls a little more important than being a babysitter.

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


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Wednesday, April 18, 2007 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] [OT] OMG Verizon is terrible


Had an appointment for these schmoes to come out and install another
line.  Was supposed to be 8-12.  Its now 6PM and not even call.


Missed


3 sales calls waiting on these jerks.

No wonder customers were jumping ship to Vonage.

--




Not on that day, I was the only one available unfortunately.




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Re: [asterisk-users] BSNL caller ID (India)

[Sanjay Rajdev]

Yes,

As I have mentioned below I tried the link
http://bugs.digium.com/view.php?id=6683&nbn=24
but was not able to make it work.

Regards,
Sanjay Rajdev


- Original Message -
From: "Steve Murphy" <[EMAIL PROTECTED]>
To: "sanjay rajdev" <[EMAIL PROTECTED]>
Sent: Thursday, April 19, 2007 4:37:39 AM (GMT+0530) Asia/Calcutta
Subject: Re: [asterisk-users] BSNL caller ID (India)

Sanjay--

Did you look at bug 6683? Sorry, I haven't reviewed all the messages on
this thread.

murf


On Wed, 2007-04-18 at 01:01 +0530, Sanjay Rajdev wrote:
> Tzafrir,
> 
> Can you Please let me know if the zapata.conf below is correct, or do I have 
> to change something.
> 
> Regards,
> Sanjay Rajdev
> 
> - Original Message -
> From: "Sanjay Rajdev" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Cc: "tzafrir cohen" <[EMAIL PROTECTED]>
> Sent: Tuesday, April 17, 2007 3:38:16 AM (GMT+0530) Asia/Calcutta
> Subject: Re: [asterisk-users] BSNL caller ID (India)
> 
> Yes below is the zapata.conf
> 
> [trunkgroups]
> 
> [channels]
> context=incoming
> usecallerid=yes
> cidsignalling=dtmf
> cidstart=ring
> hidecallerid=no
> callerid=asreceived
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> 
> ;Sangoma A200 [slot:2 bus:4 span:1]
> group=0
> signalling = fxs_ks
> channel => 1
> 
> group=0
> signalling = fxs_ks
> channel => 2
> 
> group=0
> signalling = fxs_ks
> channel => 3
> 
> group=0
> signalling = fxs_ks
> channel => 4
> 
> 
> Regards,
> Sanjay Rajdev
> 
> 
> 
> - Original Message -
> From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta
> Subject: Re: [asterisk-users] BSNL caller ID (India)
> 
> On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote:
> > Has anyone figured out the way of getting the caller id for BSNL on 
> > Asterisk 1.4.2
> > I have tried following link
> > http://bugs.digium.com/view.php?id=6683&nbn=24
> > but was not able to get it, although did not ge any error too.
> > 
> > I always get the caller id as asterisk.
> 
> Hmmm... are you sure you have configured your system to get callerid
> from the PSTN?
> 
> callerid=asrecieved
> 
> in zapata.conf.
> 
-- 
Steve Murphy <[EMAIL PROTECTED]>
Digium


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[asterisk-users] CallerID Auth

[Arun Kumar]

Hi,

in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.


thanks

arun
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[asterisk-users] Asterisk stops responding to SIP/ZAP

[Ken Williams]

About once a week or so my Asterisk box stops responding to all phones.
I can pull up the console, do whatever I want at the CLI but the only
way to get things working again is to restart Asterisk altogether.
 
I finally cranked verbose & debugging way up (and watched my log files
go from 1mb/day to 100mb/day), but below I believe contains my problem.
The next line is 1.5 minutes later where I restart Asterisk.
 
SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in
place here).  Zap/3-1 is a Digium TDM400.  
 
I can't quite figure out where my problem is, is it the initial
exception, is it not getting hung up completely, does it have to do with
the call limit on the SIP channel, perhaps 'no provider found'
statements?
 
Any help would be appreciated, I have a relatively simple dial-plan, I
can send over relevant bits of it if necessary.
 
Thanks,
Ken
 
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on
channel 3 (index 0)
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on
channel 3
[Apr 19 13:51:13] DEBUG[27722] channel.c: Didn't get a frame from
channel: Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging channels
SIP/701-08ee6120 and Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'Zap/3-1'
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1)
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0,
normal = 12, callwait = -1, thirdcall = -1
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on
channel 3
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3,
with 0 conference users
[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Hungup 'Zap/3-1'
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension
(from-internal,201,2) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
(from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing
[EMAIL PROTECTED]:1] Hangup("SIP/701-08ee6120", "") in new stack
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension
(from-internal,h,1) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
(from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel
'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call SIP/701-08ee6120,
SIP callid [EMAIL PROTECTED])
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for
incoming call
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701' removed
from call limit 6
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel SIP/701
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel SIP/701-08ee6120
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for Zap - 3
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for Zap/3 -
state 0 (Unknown)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701
- state 1 (Not in use)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701
- state 1 (Not in use)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
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