Re: [asterisk-users] Polycom BLF - multiple buddies

[Matt Riddell]

We've used the devstate backport with the snom phones for this. The
buttons toggle log in and out with one and pause/unpause with another.
We use the astdb to store current status and add/remove/pause/unpause
queue member functions. Works great

On 9/6/08, James Sneeringer <[EMAIL PROTECTED]> wrote:
> I have not applied the 1.4 backport to my system, so I haven't used
> DEVSTATE, but this page appears to show how to do what you want:
>
> http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
>
> That page also has a link to the backport.
>
> -James
>
>
> On Thu, Sep 4, 2008 at 11:34 PM, Lee, John (Sydney)
> <[EMAIL PROTECTED]> wrote:
>> James, very useful info especially about enable/disable the light next
>> to the speed dial button which is exactly what I am after.  I am
>> currently using 1.4.x and would be interested to know how this can be
>> achieved.
>
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Matt Riddell
Director
VentureVoIP

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Re: [asterisk-users] Transfers on AgentLogin()

[James Sneeringer]

Since AgentLogin() essentially keeps a channel to the agent open all
the time, a normal SIP transfer will do exactly as you say. That is,
it will try to send the agent's login session into queue, which isn't
what you want.

As Matt suggested, you need to pass the "t" option to the Queue()
application. This will let your agents perform a DTMF transfer using
the codes defined in features.conf. The agent basically dials a short
code while talking to the caller. Asterisk intercepts it, and then
prompts the agent for the extension to transfer the call to. Look in
features.conf for more information.

Fair warning, I have never needed to use this feature, so I can't
attest to exactly how it behaves. We use dynamic agent logins, so
we've never had to deal with AgentLogin(). This allows us to do normal
SIP transfers.

Also, you will probably have to do one of two things in your sip.conf.
One, set "canreinvite" to "no" to keep Asterisk in the call path, that
way it can intercept the DTMF tones. Or, two, set "dtmfmode" to
"info", so that DTMF tones are converted to SIP INFO messages, which
Asterisk will see.

At least, that's how I think it works. :)

-James


On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton <[EMAIL PROTECTED]> wrote:
> I've tried the regular, xfer button on xlite, dial 100 (to transfer to the
> queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(),
> so it transfers the full persistent connection to the queue instead of the
> call itself and this causes the transferring agent to logout.
>
> Either that, or I'm doing something wrong. There is no documentation out
> there so I don't know how it would work for AgentLogin().
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
> Sent: August 30, 2008 6:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers on AgentLogin()
>
> What did you try and how did it fail? Are you using the t option in queue?
>

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Re: [asterisk-users] Polycom BLF - multiple buddies

[James Sneeringer]

I have not applied the 1.4 backport to my system, so I haven't used
DEVSTATE, but this page appears to show how to do what you want:

http://www.voip-info.org/wiki/view/Asterisk+func+Devstate

That page also has a link to the backport.

-James


On Thu, Sep 4, 2008 at 11:34 PM, Lee, John (Sydney)
<[EMAIL PROTECTED]> wrote:
> James, very useful info especially about enable/disable the light next
> to the speed dial button which is exactly what I am after.  I am
> currently using 1.4.x and would be interested to know how this can be
> achieved.

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Re: [asterisk-users] dahdi & tdm400p: no luck

[sean darcy]

sean darcy wrote:
> Tzafrir Cohen wrote:
>> On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
>>> Tzafrir Cohen wrote:
 On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
> Tzafrir Cohen wrote:
>> On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
>>> As best i could figure it out, I've installed dahdi and rc4.
>>>
>>> My TDM400P doesn't answer fxo or fxs.
>>>
>>> /etc/dahdi/system.conf:
>>> loadzone   = us
>>> defaultzone=us
>>> fxoks=1,2
>>> fxsks=4
>> echocancel?
>>
> I thought that if you had hardware echocancel ( TDM400P does, doesn't 
> it? ), 
 TDM400P doesn't. Do you mean TDM410P?

> setting the software echocanceller was irrelevant. In any event, 
> isn't mg2 the deefault?
 No. You may have that impression from the configuration generated by
 dahdi_genconf that adds it as a default (that is: generates an explicit
 echocancel line for each channel) due to this limitation. That may
 change in the future if system.conf will grow up its own default echo
 canceller.

>   I'll take the system down and change this, and dahdichan to 1,2 later 
> today, though again that wouldn't explain the lack of call pickup on the 
> _external_ line. show daahdi channels shows _no_ channels.  ( sigh)
 And this still does not explain why you have not posted the output of:

  cat /proc/dahdi/*

 ;-)

> And, I'm using 1.6.0-rc4.
>>> I've got 1.6.0-rc4 up again.
>>>
>>> cat /proc/dahdi/*
>>> Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
>>>
>>>1 WCTDM/4/0 FXOKS
>>>2 WCTDM/4/1 FXOKS
>>>3 WCTDM/4/2
>>>4 WCTDM/4/3 FXSKS
>>>
>>> and dahdi_cfg seems to have worked:
>>>
>>> dahdi_cfg -vv
>>> DAHDI Tools Version - 2.0.0-rc2
>>>
>>> DAHDI Version: 2.0.0-rc3
>>> Echo Canceller(s):
>>> Configuration
>>> ==
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>>>
>>> 3 channels to configure.
>>>
>>> Changing signalling on channel 1 from Unused to FXO Kewlstart
>>> Changing signalling on channel 2 from Unused to FXO Kewlstart
>>> Changing signalling on channel 4 from Unused to FXS Kewlstart
>>>
>>> but still no luck. No dial tone for the internal phones, no answer on pstn.
>>>
>>> *CLI> dahdi show status
>>> Description  Alarms  IRQbpviol CRC4 
>>> Fra Codi Options  LBO
>>> Wildcard TDM400P REV I Board 5   OK  0  0  0 
>>> CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
>>> *CLI> dahdi show channels
>>> Chan Extension  Context Language   MOH Interpret 
>>> BlockedState
>>>   pseudodefaultdefault 
>>> In Service
>>> *CLI> dahdi show channel 1
>>> Unable to find given channel 1
>>> Command 'dahdi show channel 1' failed.
>>>
>>> cat /etc/dahdi/system.conf
>>> # note change in fxo_ks and fx2_ks. 1 & 2 are internal, 4 is extension
>>> fxoks=1,2
>>> fxsks=4
>>>
>>> loadzone= us
>>> defaultzone = us
>>>
>>> BTW, this file is sometimes referred to as dahdi.conf - to keep us on 
>>> our toes.  and what is the comment sign ; or # ?
>>>
>>> cat /etc/asterisk/chan_dahdi.conf
>>>
>>> [trunkgroups]
>>>
>>> [channels]
>>> usecallerid=yes
>>> callwaiting=yes
>>> usecallingpres=yes
>>> callwaitingcallerid=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> canpark=yes
>>> cancallforward=yes
>>> callreturn=yes
>>> echocancel=no
>>> echocancelwhenbridged=no
>>> echotraining=no
>>>
>>> group=1
>>> callgroup=1
>>> pickupgroup=1
>>>
>>> callprogress=yes
>>> progzone=us
>>> tonezone = 0 ; 0 is US
>>> jbenable = yes  ; Enables the use of a jitterbuffer on the 
>>> receiving side of a
>>>; DAHDI channel. Defaults to "no". An 
>>> enabled jitterbuffer will
>>>; be used only if the sending side can 
>>> create and the receiving
>>>; side can not accept jitter. The DAHDI 
>>> channel can't accept jitter,
>>>; thus an enabled jitterbuffer on the 
>>> receive DAHDI side will always
>>>; be used if the sendi
>>>
>>> [home-phones]
>>> context=internal  ; Uses the [internal] context in extensions.conf
>>> signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as 
>>> set in sytem.conf.conf
>>> ;channel => 1  ; Telephone attached to port 1
>>> ;channel => 2  ; Telephone attached to port 2
>>> dahdichan => 1,2
>>>
>>> [pstn]
>>> context=incoming-pstn-line  ; Incoming calls go to [incoming-pstn-line] 
>>> in extensions.conf
>>> signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use 
>>> as set in system.conf
>>> faxdetect=incoming
>>> busydetect=yes
>>> ;channel => 4
>>> 

Re: [asterisk-users] dahdi & tdm400p: no luck

[sean darcy]

Tzafrir Cohen wrote:
> On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
>> Tzafrir Cohen wrote:
>>> On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
 Tzafrir Cohen wrote:
> On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
>> As best i could figure it out, I've installed dahdi and rc4.
>>
>> My TDM400P doesn't answer fxo or fxs.
>>
>> /etc/dahdi/system.conf:
>> loadzone   = us
>> defaultzone=us
>> fxoks=1,2
>> fxsks=4
> echocancel?
>
 I thought that if you had hardware echocancel ( TDM400P does, doesn't 
 it? ), 
>>> TDM400P doesn't. Do you mean TDM410P?
>>>
 setting the software echocanceller was irrelevant. In any event, 
 isn't mg2 the deefault?
>>> No. You may have that impression from the configuration generated by
>>> dahdi_genconf that adds it as a default (that is: generates an explicit
>>> echocancel line for each channel) due to this limitation. That may
>>> change in the future if system.conf will grow up its own default echo
>>> canceller.
>>>
   I'll take the system down and change this, and dahdichan to 1,2 later 
 today, though again that wouldn't explain the lack of call pickup on the 
 _external_ line. show daahdi channels shows _no_ channels.  ( sigh)
>>> And this still does not explain why you have not posted the output of:
>>>
>>>  cat /proc/dahdi/*
>>>
>>> ;-)
>>>
 And, I'm using 1.6.0-rc4.
>> I've got 1.6.0-rc4 up again.
>>
>> cat /proc/dahdi/*
>> Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
>>
>> 1 WCTDM/4/0 FXOKS
>> 2 WCTDM/4/1 FXOKS
>> 3 WCTDM/4/2
>> 4 WCTDM/4/3 FXSKS
>>
>> and dahdi_cfg seems to have worked:
>>
>> dahdi_cfg -vv
>> DAHDI Tools Version - 2.0.0-rc2
>>
>> DAHDI Version: 2.0.0-rc3
>> Echo Canceller(s):
>> Configuration
>> ==
>>
>>
>> Channel map:
>>
>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>>
>> 3 channels to configure.
>>
>> Changing signalling on channel 1 from Unused to FXO Kewlstart
>> Changing signalling on channel 2 from Unused to FXO Kewlstart
>> Changing signalling on channel 4 from Unused to FXS Kewlstart
>>
>> but still no luck. No dial tone for the internal phones, no answer on pstn.
>>
>> *CLI> dahdi show status
>> Description  Alarms  IRQbpviol CRC4 
>> Fra Codi Options  LBO
>> Wildcard TDM400P REV I Board 5   OK  0  0  0 
>> CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
>> *CLI> dahdi show channels
>> Chan Extension  Context Language   MOH Interpret 
>> BlockedState
>>   pseudodefaultdefault 
>> In Service
>> *CLI> dahdi show channel 1
>> Unable to find given channel 1
>> Command 'dahdi show channel 1' failed.
>>
>> cat /etc/dahdi/system.conf
>> # note change in fxo_ks and fx2_ks. 1 & 2 are internal, 4 is extension
>> fxoks=1,2
>> fxsks=4
>>
>> loadzone= us
>> defaultzone = us
>>
>> BTW, this file is sometimes referred to as dahdi.conf - to keep us on 
>> our toes.  and what is the comment sign ; or # ?
>>
>> cat /etc/asterisk/chan_dahdi.conf
>>
>> [trunkgroups]
>>
>> [channels]
>> usecallerid=yes
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=no
>> echocancelwhenbridged=no
>> echotraining=no
>>
>> group=1
>> callgroup=1
>> pickupgroup=1
>>
>> callprogress=yes
>> progzone=us
>> tonezone = 0 ; 0 is US
>> jbenable = yes  ; Enables the use of a jitterbuffer on the 
>> receiving side of a
>>; DAHDI channel. Defaults to "no". An 
>> enabled jitterbuffer will
>>; be used only if the sending side can 
>> create and the receiving
>>; side can not accept jitter. The DAHDI 
>> channel can't accept jitter,
>>; thus an enabled jitterbuffer on the 
>> receive DAHDI side will always
>>; be used if the sendi
>>
>> [home-phones]
>> context=internal  ; Uses the [internal] context in extensions.conf
>> signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as 
>> set in sytem.conf.conf
>> ;channel => 1  ; Telephone attached to port 1
>> ;channel => 2  ; Telephone attached to port 2
>> dahdichan => 1,2
>>
>> [pstn]
>> context=incoming-pstn-line  ; Incoming calls go to [incoming-pstn-line] 
>> in extensions.conf
>> signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use 
>> as set in system.conf
>> faxdetect=incoming
>> busydetect=yes
>> ;channel => 4
>> dahdichan => 4  ; PSTN attached to port 4
> 
> Looks OK.
> 
> What messages do you get when you run in the CLI:
> 
>   dahdi restart
> 
dahdi restart
  Destroyi

Re: [asterisk-users] dahdi & tdm400p: no luck

[Tzafrir Cohen]

On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
> Tzafrir Cohen wrote:
> > On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
> >> Tzafrir Cohen wrote:
> >>> On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
>  As best i could figure it out, I've installed dahdi and rc4.
> 
>  My TDM400P doesn't answer fxo or fxs.
> 
>  /etc/dahdi/system.conf:
>  loadzone   = us
>  defaultzone=us
>  fxoks=1,2
>  fxsks=4
> >>> echocancel?
> >>>
> >> I thought that if you had hardware echocancel ( TDM400P does, doesn't 
> >> it? ), 
> > 
> > TDM400P doesn't. Do you mean TDM410P?
> > 
> >> setting the software echocanceller was irrelevant. In any event, 
> >> isn't mg2 the deefault?
> > 
> > No. You may have that impression from the configuration generated by
> > dahdi_genconf that adds it as a default (that is: generates an explicit
> > echocancel line for each channel) due to this limitation. That may
> > change in the future if system.conf will grow up its own default echo
> > canceller.
> > 
> >>   I'll take the system down and change this, and dahdichan to 1,2 later 
> >> today, though again that wouldn't explain the lack of call pickup on the 
> >> _external_ line. show daahdi channels shows _no_ channels.  ( sigh)
> > 
> > And this still does not explain why you have not posted the output of:
> > 
> >  cat /proc/dahdi/*
> > 
> > ;-)
> > 
> >> And, I'm using 1.6.0-rc4.
> > 
> I've got 1.6.0-rc4 up again.
> 
> cat /proc/dahdi/*
> Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
> 
>  1 WCTDM/4/0 FXOKS
>  2 WCTDM/4/1 FXOKS
>  3 WCTDM/4/2
>  4 WCTDM/4/3 FXSKS
> 
> and dahdi_cfg seems to have worked:
> 
> dahdi_cfg -vv
> DAHDI Tools Version - 2.0.0-rc2
> 
> DAHDI Version: 2.0.0-rc3
> Echo Canceller(s):
> Configuration
> ==
> 
> 
> Channel map:
> 
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
> 
> 3 channels to configure.
> 
> Changing signalling on channel 1 from Unused to FXO Kewlstart
> Changing signalling on channel 2 from Unused to FXO Kewlstart
> Changing signalling on channel 4 from Unused to FXS Kewlstart
> 
> but still no luck. No dial tone for the internal phones, no answer on pstn.
> 
> *CLI> dahdi show status
> Description  Alarms  IRQbpviol CRC4 
> Fra Codi Options  LBO
> Wildcard TDM400P REV I Board 5   OK  0  0  0 
> CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
> *CLI> dahdi show channels
> Chan Extension  Context Language   MOH Interpret 
> BlockedState
>   pseudodefaultdefault 
> In Service
> *CLI> dahdi show channel 1
> Unable to find given channel 1
> Command 'dahdi show channel 1' failed.
> 
> cat /etc/dahdi/system.conf
> # note change in fxo_ks and fx2_ks. 1 & 2 are internal, 4 is extension
> fxoks=1,2
> fxsks=4
> 
> loadzone= us
> defaultzone = us
> 
> BTW, this file is sometimes referred to as dahdi.conf - to keep us on 
> our toes.  and what is the comment sign ; or # ?
> 
> cat /etc/asterisk/chan_dahdi.conf
> 
> [trunkgroups]
> 
> [channels]
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=no
> echocancelwhenbridged=no
> echotraining=no
> 
> group=1
> callgroup=1
> pickupgroup=1
> 
> callprogress=yes
> progzone=us
> tonezone = 0 ; 0 is US
> jbenable = yes  ; Enables the use of a jitterbuffer on the 
> receiving side of a
>; DAHDI channel. Defaults to "no". An 
> enabled jitterbuffer will
>; be used only if the sending side can 
> create and the receiving
>; side can not accept jitter. The DAHDI 
> channel can't accept jitter,
>; thus an enabled jitterbuffer on the 
> receive DAHDI side will always
>; be used if the sendi
> 
> [home-phones]
> context=internal  ; Uses the [internal] context in extensions.conf
> signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as 
> set in sytem.conf.conf
> ;channel => 1  ; Telephone attached to port 1
> ;channel => 2  ; Telephone attached to port 2
> dahdichan => 1,2
> 
> [pstn]
> context=incoming-pstn-line  ; Incoming calls go to [incoming-pstn-line] 
> in extensions.conf
> signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use 
> as set in system.conf
> faxdetect=incoming
> busydetect=yes
> ;channel => 4
> dahdichan => 4  ; PSTN attached to port 4

Looks OK.

What messages do you get when you run in the CLI:

  dahdi restart

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL P

Re: [asterisk-users] dahdi & tdm400p: no luck

[sean darcy]

Tzafrir Cohen wrote:
> On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
>> Tzafrir Cohen wrote:
>>> On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
 As best i could figure it out, I've installed dahdi and rc4.

 My TDM400P doesn't answer fxo or fxs.

 /etc/dahdi/system.conf:
 loadzone   = us
 defaultzone=us
 fxoks=1,2
 fxsks=4
>>> echocancel?
>>>
>> I thought that if you had hardware echocancel ( TDM400P does, doesn't 
>> it? ), 
> 
> TDM400P doesn't. Do you mean TDM410P?
> 
>> setting the software echocanceller was irrelevant. In any event, 
>> isn't mg2 the deefault?
> 
> No. You may have that impression from the configuration generated by
> dahdi_genconf that adds it as a default (that is: generates an explicit
> echocancel line for each channel) due to this limitation. That may
> change in the future if system.conf will grow up its own default echo
> canceller.
> 
>>   I'll take the system down and change this, and dahdichan to 1,2 later 
>> today, though again that wouldn't explain the lack of call pickup on the 
>> _external_ line. show daahdi channels shows _no_ channels.  ( sigh)
> 
> And this still does not explain why you have not posted the output of:
> 
>  cat /proc/dahdi/*
> 
> ;-)
> 
>> And, I'm using 1.6.0-rc4.
> 
I've got 1.6.0-rc4 up again.

cat /proc/dahdi/*
Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)

   1 WCTDM/4/0 FXOKS
   2 WCTDM/4/1 FXOKS
   3 WCTDM/4/2
   4 WCTDM/4/3 FXSKS

and dahdi_cfg seems to have worked:

dahdi_cfg -vv
DAHDI Tools Version - 2.0.0-rc2

DAHDI Version: 2.0.0-rc3
Echo Canceller(s):
Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

3 channels to configure.

Changing signalling on channel 1 from Unused to FXO Kewlstart
Changing signalling on channel 2 from Unused to FXO Kewlstart
Changing signalling on channel 4 from Unused to FXS Kewlstart

but still no luck. No dial tone for the internal phones, no answer on pstn.

*CLI> dahdi show status
Description  Alarms  IRQbpviol CRC4 
Fra Codi Options  LBO
Wildcard TDM400P REV I Board 5   OK  0  0  0 
CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
*CLI> dahdi show channels
Chan Extension  Context Language   MOH Interpret 
BlockedState
  pseudodefaultdefault 
In Service
*CLI> dahdi show channel 1
Unable to find given channel 1
Command 'dahdi show channel 1' failed.

cat /etc/dahdi/system.conf
# note change in fxo_ks and fx2_ks. 1 & 2 are internal, 4 is extension
fxoks=1,2
fxsks=4

loadzone= us
defaultzone = us

BTW, this file is sometimes referred to as dahdi.conf - to keep us on 
our toes.  and what is the comment sign ; or # ?

cat /etc/asterisk/chan_dahdi.conf

[trunkgroups]

[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
echotraining=no

group=1
callgroup=1
pickupgroup=1

callprogress=yes
progzone=us
tonezone = 0 ; 0 is US
jbenable = yes  ; Enables the use of a jitterbuffer on the 
receiving side of a
   ; DAHDI channel. Defaults to "no". An 
enabled jitterbuffer will
   ; be used only if the sending side can 
create and the receiving
   ; side can not accept jitter. The DAHDI 
channel can't accept jitter,
   ; thus an enabled jitterbuffer on the 
receive DAHDI side will always
   ; be used if the sendi

[home-phones]
context=internal  ; Uses the [internal] context in extensions.conf
signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as 
set in sytem.conf.conf
;channel => 1  ; Telephone attached to port 1
;channel => 2  ; Telephone attached to port 2
dahdichan => 1,2

[pstn]
context=incoming-pstn-line  ; Incoming calls go to [incoming-pstn-line] 
in extensions.conf
signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use 
as set in system.conf
faxdetect=incoming
busydetect=yes
;channel => 4
dahdichan => 4  ; PSTN attached to port 4

Thanks for any help.

sean


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Re: [asterisk-users] (no subject)

[Shariq Khan]

What asterisk cli shows when you soft hangup these channels


Shariq

On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen <[EMAIL PROTECTED]>wrote:

> V 1.4
>
> When I do a "show channels" I get the following.
>
> CLI> show channels
> Channel  Location State   Application(Data)
> SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
> Page(&Local/[EMAIL PROTECTED]&Local/71
> SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
> Page(&Local/[EMAIL PROTECTED]&Local/71
> 2 active channels
> 2 active calls
>
> I need to kill these SIP channels, but the only thing I have found when
> searching
> is the "soft hangup" solution - which simply doesn't do anything to these
> channels.
>
> CLI> soft hangup SIP/7110-b495d3b0
>
> CLI> soft hangup SIP/7110-afd286e0
>
> CLI> show channels
> Channel  Location State   Application(Data)
> SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
> Page(&Local/[EMAIL PROTECTED]&Local/71
> SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
> Page(&Local/[EMAIL PROTECTED]&Local/71
> 2 active channels
> 2 active calls
>
> Can someone suggest a better way of getting rid of these channels?  My
> solution
> so far has been to restart Asterisk... not a good solution.
>
> Thanks
>
> Bill
>
>
>
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Re: [asterisk-users] FAX over T1 Question

[Eric "ManxPower" Wieling]

The thing is, you are doing FAX over PRI, not FAX over T-1 (which to me 
implies Channelized T-1). Seems like most people gave you advice that 
might apply to a Channelized T-1, but would not apply or be practical 
for a PRI.

Amaru Netapshaak wrote:
> Bob,
> 
> I should have added that I have disabled hardware EC on the T1 ports.
> 
> Here is a sample of my zapata.conf -- channels 1-23 are my incoming PRI.
> This PRI handles both Voice AND FAX calls.  Having the hardware EC
> disabled makes for poor voice communications, and im looking for a way to
> enable/disable EC per the call type.   I understand that 
> "echoncancelwhenbridged"
> and Zapata should be telling my A104d to enable/disable HWEC automatically.
> 
> Channel 25 is the first FXS port on my Rhino CB.  It has a FAX directly
> attached to it.
> 
> [channels]
> language=  en
> switchtype  =  national
> signalling  =  pri_cpe
> pridialplan =  national
> prilocaldialplan=  national
> faxdetect   =  both
> echotraining=  no
> echocancel  =  256
> echocancelwhenbridged   =  no
> relaxdtmf   =  yes
> overlapdial =  no
> usecallingpres  =  yes
> amaflags=  default
> context =  default
> group   =  1
> channel => 1-23
> 
> signalling  = fxo_ls
> faxdetect   = both
> echotraining= no
> echocancel  = no
> echocancelwhenbridged   = no
> relaxdtmf   = yes
> context = default
> callerid= "FAX" <111>
> channel => 25
> 
> Thanks for your assistance everyone! 
> 
> 
> 
> --- On Fri, 9/5/08, Bob Pierce <[EMAIL PROTECTED]> wrote:
> From: Bob Pierce <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] FAX over T1 Question
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Date: Friday, September 5, 2008, 4:43 PM
> 
> On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
>> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
>> running 
>> Asterisk 1.4.21.2
> 
> I think you're mostly right on this setup, but I wonder if your A104d is
> doing some hardware echo cancellation on these calls. If I'm not
> mistaken, that can mess up fax machine communications.
> 
> Bob
> 
> 
> 
>   
> 
> 
> 
> 
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-- 
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T-1, PRI, Frame Relay, Linux, and network design.  Based near 
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Re: [asterisk-users] svn branches for dhadi and its tools

[John covici]

OK, thanks.

on Friday 09/05/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
 > On Fri, Sep 05, 2008 at 10:32:30AM -0400, John covici wrote:
 > > Hi.  I want to use the new asterisk 1.4 with dahdi, but I would like
 > > to know the svn branches for the dahdi, so I can use them that way --
 > > much easier to keep up with bug fixes, etc.
 > 
 > trunk, in both cases.
 > 
 > http://svn.digium.com/svn/dahdi/linux/trunk
 > http://svn.digium.com/svn/dahdi/tools/trunk
 > 
 > -- 
 >Tzafrir Cohen
 > icq#16849755  jabber:[EMAIL PROTECTED]
 > +972-50-7952406   mailto:[EMAIL PROTECTED]
 > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
 > 
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How do
you spend it?

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 [EMAIL PROTECTED]

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[asterisk-users] soft hangup (was: Re: (no subject))

[Philipp Kempgen]

Bill Andersen schrieb:
> V 1.4
> 
> When I do a "show channels" I get the following.
> 
> CLI> show channels
> Channel  Location State   Application(Data)
> SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
> Page(&Local/[EMAIL PROTECTED]&Local/71
> SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
> Page(&Local/[EMAIL PROTECTED]&Local/71
> 2 active channels
> 2 active calls
> 
> I need to kill these SIP channels, but the only thing I have found when
> searching
> is the "soft hangup" solution - which simply doesn't do anything to these
> channels.

> Can someone suggest a better way of getting rid of these channels?  My
> solution
> so far has been to restart Asterisk... not a good solution.

I'd try if the "/n" flag help when dialing to Local channels,
e.g. Local/7114/n instead of Local/7114

afaicr there is a problem with the channel masquerading code.

Of course that doesn't help you to get rid of the channels
without a restart.

   Philipp Kempgen

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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Steve Repo wrote:
> 
> > AFAIK, Zaptel is no longer supported. I'd recommend dahdi if it's a new 
> > setup.
> 
> Zaptel is still 'supported', in that we'll help you analyze problems,
> fix your configuration, etc. The only area where Zaptel is not
> 'supported' is that we won't be making any more regular releases, except
> for fixing critical bugs in the 1.4.12 release (of which I've seen one
> or two reports so far).
> 
> If you are comfortable with Zaptel, and plan to use only hardware in the
> system that is currently supported with Zaptel, then stick with it for
> now. DAHDI is still in 'release candidate' state anyway, so only people
> willing to test it out should be using it, it is not recommended for
> production systems yet.

Cool, thanks. So it seems the current recommendation is to use DAHDI 2.0
with Asterisk 1.6, and Zaptel 1.4 with Asterisk 1.4.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] FAX over T1 Question

[Amaru Netapshaak]

Bob,

I should have added that I have disabled hardware EC on the T1 ports.

Here is a sample of my zapata.conf -- channels 1-23 are my incoming PRI.
This PRI handles both Voice AND FAX calls.  Having the hardware EC
disabled makes for poor voice communications, and im looking for a way to
enable/disable EC per the call type.   I understand that 
"echoncancelwhenbridged"
and Zapata should be telling my A104d to enable/disable HWEC automatically.

Channel 25 is the first FXS port on my Rhino CB.  It has a FAX directly
attached to it.

[channels]
language    =  en
switchtype  =  national
signalling  =  pri_cpe
pridialplan =  national
prilocaldialplan    =  national
faxdetect   =  both
echotraining    =  no
echocancel  =  256
echocancelwhenbridged   =  no
relaxdtmf   =  yes
overlapdial =  no
usecallingpres  =  yes
amaflags    =  default
context =  default
group   =  1
channel => 1-23

signalling  = fxo_ls
faxdetect   = both
echotraining    = no
echocancel  = no
echocancelwhenbridged   = no
relaxdtmf   = yes
context = default
callerid    = "FAX" <111>
channel => 25

Thanks for your assistance everyone! 



--- On Fri, 9/5/08, Bob Pierce <[EMAIL PROTECTED]> wrote:
From: Bob Pierce <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] FAX over T1 Question
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Friday, September 5, 2008, 4:43 PM

On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
> running 
> Asterisk 1.4.21.2

I think you're mostly right on this setup, but I wonder if your A104d is
doing some hardware echo cancellation on these calls. If I'm not
mistaken, that can mess up fax machine communications.

Bob



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[asterisk-users] (no subject)

[Bill Andersen]

V 1.4

When I do a "show channels" I get the following.

CLI> show channels
Channel  Location State   Application(Data)
SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
Page(&Local/[EMAIL PROTECTED]&Local/71
SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
Page(&Local/[EMAIL PROTECTED]&Local/71
2 active channels
2 active calls

I need to kill these SIP channels, but the only thing I have found when
searching
is the "soft hangup" solution - which simply doesn't do anything to these
channels.

CLI> soft hangup SIP/7110-b495d3b0

CLI> soft hangup SIP/7110-afd286e0

CLI> show channels
Channel  Location State   Application(Data)
SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
Page(&Local/[EMAIL PROTECTED]&Local/71
SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
Page(&Local/[EMAIL PROTECTED]&Local/71
2 active channels
2 active calls

Can someone suggest a better way of getting rid of these channels?  My
solution
so far has been to restart Asterisk... not a good solution.

Thanks

Bill



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Re: [asterisk-users] svn branches for dhadi and its tools

[Tzafrir Cohen]

On Fri, Sep 05, 2008 at 10:32:30AM -0400, John covici wrote:
> Hi.  I want to use the new asterisk 1.4 with dahdi, but I would like
> to know the svn branches for the dahdi, so I can use them that way --
> much easier to keep up with bug fixes, etc.

trunk, in both cases.

http://svn.digium.com/svn/dahdi/linux/trunk
http://svn.digium.com/svn/dahdi/tools/trunk

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

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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI

[Tzafrir Cohen]

On Fri, Sep 05, 2008 at 02:46:46PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Russell Bryant <[EMAIL PROTECTED]> wrote:
> > 
> > On Sep 3, 2008, at 8:32 PM, sean darcy wrote:
> > 
> > > Great.
> > >
> > > But I'm still a little confused.
> > >
> > > Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
> > 
> > No.  Asterisk 1.6.0 now _only_ supports DAHDI.
> > 
> > > It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We
> > > can go back to this release of zaptel if we have problems with dahdi.
> > >
> > > Or if we go back to zaptel, do we go back to 1.6.0-beta9 also?
> > 
> > You can upgrade directly to DAHDI.  However, if you have trouble with  
> > DAHDI and need to go back to Zaptel, then I would go back to Asterisk  
> > 1.4 instead of using an old beta of 1.6.0.  Many things have been  
> > fixed since 1.6.0-beta9.
> 
> If I'm installing a new system based on the latest Asterisk 1.4, should
> I use zaptel or dahdi with it? Which version?

DAHDI is only supported as of 1.4.22 (currently only availalbe as an
RC).

every Asterisk 1.4 version should work with Zaptel . Though 1.6.0 and on
will no longer work with Zaptel. If you install 1.4.22, consider DAHDI.

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

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Re: [asterisk-users] FAX over T1 Question

[Vinícius Fontes]

I'm having problems exactly with that tone detection. I even submitted a bug 
report (http://bugs.digium.com/view.php?id=13286) but it still has not been 
viewed yet, I guess.




Atenciosamente,

Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
 
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000

- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> escreveu:

> If I am not mistaken every single echo canceler out there will disable
> 
> itself if it detects a fax tone.
> 
> Echo Cancelers do not screw up faxes, people screw up faxes. 8-)
> 
> Bob Pierce wrote:
> > On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
> >> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
> >> running 
> >> Asterisk 1.4.21.2
> > 
> > I think you're mostly right on this setup, but I wonder if your
> A104d is
> > doing some hardware echo cancellation on these calls. If I'm not
> > mistaken, that can mess up fax machine communications.
> > 
> > Bob
> > 
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> 
> -- 
> Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN,
> QoS, 
> T-1, PRI, Frame Relay, Linux, and network design.  Based near 
> Birmingham, AL.  Now accepting clients worldwide.
> 
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Re: [asterisk-users] FAX over T1 Question

[Alex Balashov]

Eric "ManxPower" Wieling wrote:

> If I am not mistaken every single echo canceler out there will disable 
> itself if it detects a fax tone.
> 
> Echo Cancelers do not screw up faxes, people screw up faxes. 8-)

Never underestimate how ghetto an echo canceller can be.  :-)

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Re: [asterisk-users] FAX over T1 Question

[Steve Totaro]

I would not bother with fax detection with fax DIDs and on T1s/PRIs.
The fewer the modules that you need to rely on and load, the better.

Thanks,
Steve Totaro
1.888.777.1888

On Fri, Sep 5, 2008 at 1:58 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]> 
wrote:
> If I am not mistaken every single echo canceler out there will disable
> itself if it detects a fax tone.
>
> Echo Cancelers do not screw up faxes, people screw up faxes. 8-)
>
> Bob Pierce wrote:
>> On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
>>> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
>>> running
>>> Asterisk 1.4.21.2
>>
>> I think you're mostly right on this setup, but I wonder if your A104d is
>> doing some hardware echo cancellation on these calls. If I'm not
>> mistaken, that can mess up fax machine communications.
>>
>> Bob
>>

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Re: [asterisk-users] FAX over T1 Question

[Eric "ManxPower" Wieling]

If I am not mistaken every single echo canceler out there will disable 
itself if it detects a fax tone.

Echo Cancelers do not screw up faxes, people screw up faxes. 8-)

Bob Pierce wrote:
> On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
>> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
>> running 
>> Asterisk 1.4.21.2
> 
> I think you're mostly right on this setup, but I wonder if your A104d is
> doing some hardware echo cancellation on these calls. If I'm not
> mistaken, that can mess up fax machine communications.
> 
> Bob
> 
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> 

-- 
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T-1, PRI, Frame Relay, Linux, and network design.  Based near 
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Re: [asterisk-users] Call monitor/barge/train

[Steve Totaro]

On Fri, Aug 29, 2008 at 2:14 PM, Mark Hamilton <[EMAIL PROTECTED]> wrote:
> Hi,
>
>
>
> I'm planning on migrating someone who uses a very mature system. They would
> be logging in either as AgentLogin() or AQM. The main requirement however,
> is:
>
> The supervisor will have a control panel, where he will see how many of his
> agents are on call. If they are, he can "right-click" on the agent and get
> the options Call Monitor (where the super just listens in on the call, or
> new reps can listenin), Call Train (where the super and agent can talk to
> each other for training, but the customer doesn't hear them, or older reps
> can train newer reps), Call Barge (where everyone can hear everyone else,
> super agent and caller)
>
>
>
> How or what can facilitate this?
>
>
>
> Thanks!
>

You might want to check out QueueMetrics.  It is free for two agents,
so I don't feel it out of place to post here.
http://queuemetrics.com/

It may not fit the bill for what you want entirely but it is very
feature rich and may solve other requirements you may have now or in
the future.

Thanks,
Steve Totaro
1.888.777.1888

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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI

[Kevin P. Fleming]

Steve Repo wrote:

> AFAIK, Zaptel is no longer supported. I'd recommend dahdi if it's a new setup.

Zaptel is still 'supported', in that we'll help you analyze problems,
fix your configuration, etc. The only area where Zaptel is not
'supported' is that we won't be making any more regular releases, except
for fixing critical bugs in the 1.4.12 release (of which I've seen one
or two reports so far).

If you are comfortable with Zaptel, and plan to use only hardware in the
system that is currently supported with Zaptel, then stick with it for
now. DAHDI is still in 'release candidate' state anyway, so only people
willing to test it out should be using it, it is not recommended for
production systems yet.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Call monitor/barge/train

[David Backeberg]

> The supervisor will have a control panel, where he will see how many
> of his agents are on call. If they are, he can "right-click" on the
> agent and get the options Call Monitor (where the super just listens
> in on the call, or new reps can listenin), Call Train (where the super
> and agent can talk to each other for training, but the customer
> doesn't hear them, or older reps can train newer reps), Call Barge
> (where everyone can hear everyone else, super agent and caller)

We built our own solution that does everything you said, except I
didn't bother with the AJAX, right-click handling, etc. I don't know
why "right-click" has to be a particular part of the feature.

Anyway, our agents dial into a special number, authenticate, get
prompted by an IVR menu for what they want to do (take calls, dial
out, train) and then await calls. Other agents can train by shadowing
other agent calls using ChanSpy(). People with web access can control
who can listen to whom, or if at all. To actively monitor somebody, a
person with web access can configure their own account to have listen
privileges against the chosen agent. We configure all of this using
custom-built web interface with PHP, MySQL backend, and Perl agi-bin
scripts. We built the current level of features over a span of about a
year, and at each stage of development, would add a few features, test
drive on a development system, then put it live, repeat. I don't see
how you're going to get from zero to 100mph without some custom
development work.

The least important part of what you want to do is the interface, but
that's what you led with. If you're really going through with this,
don't prioritize the interface. Instead prioritize the phones
features, and improve the interface once the phones features are rock
solid. I'm not aware of anything ready-built that does what you want.
FreePBX might provide the web-interface part that would at least let
you know what agents are currently logged in, and you might be then
able to extend FreePBX. I've never tried to do anything like this with
FreePBX. Very briefly, if you do the custom thing, design a good data
model that handles your workflow. Make agi-bin scripts and web scripts
that update that data model when things change in the system, then
hook them up to the web interface and asterisk.

I don't know what you're using for agent phones. Before I built our
particular web-based Asterisk system, our proprietary PBX phones had a
literal double-jack feature, where the call could be monitored in
person, while sitting next to the agent. There were also messier ways
to do that with feature codes, using that proprietary PBX, but
ultimately it was too hard to explain, and this system is easier to
use for the end-users.

-David

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Re: [asterisk-users] FAX over T1 Question

[Tim Nelson]

Yep... your A104d has HWEC onboard (as signified by the 'd' on your model). It 
is necessary to set echocancel=no and probably echocancelwhenbridged=no in your 
zapata.conf to get reliable faxing to work. 

Tim Nelson
Systems/Network Engineer
Rockbochs Inc.
(218)727-4332 x105

- "Bob Pierce" <[EMAIL PROTECTED]> wrote:

> On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
> > I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
> > running 
> > Asterisk 1.4.21.2
> 
> I think you're mostly right on this setup, but I wonder if your A104d
> is
> doing some hardware echo cancellation on these calls. If I'm not
> mistaken, that can mess up fax machine communications.
> 
> Bob
> 
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Re: [asterisk-users] FAX over T1 Question

[Steve Totaro]

On Fri, Sep 5, 2008 at 12:43 PM, Bob Pierce <[EMAIL PROTECTED]> wrote:
>
> On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
>> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
>> running
>> Asterisk 1.4.21.2
>
> I think you're mostly right on this setup, but I wonder if your A104d is
> doing some hardware echo cancellation on these calls. If I'm not
> mistaken, that can mess up fax machine communications.
>
> Bob
>

Try echocancelwhenbridged=no

BTW, you should most likely be able to get this working very well, if
not on par with POTS lines.

Thanks,
Steve Totaro

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Re: [asterisk-users] FAX over T1 Question

[Bob Pierce]

On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
> I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
> running 
> Asterisk 1.4.21.2

I think you're mostly right on this setup, but I wonder if your A104d is
doing some hardware echo cancellation on these calls. If I'm not
mistaken, that can mess up fax machine communications.

Bob

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[asterisk-users] FAX over T1 Question

[Amaru Netapshaak]

Hello,

I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am running 
Asterisk 1.4.21.2

FAXing works, but not so reliably. I'm wondering what I might have to do in
 order to make this work. I have my FAX machines connected to individial 
FXS ports on my Rhino FXS channel bank, and that Channel Bank is 
connected to port #2 on my Sangoma A104d.

All my incoming calls come across a PRI which is connected to 
port #1 on that Sangoma Card.

Port #2 is configured to get its timing from port #1.

Right now, when a FAX comes in, it hits the corresponding extension in 
my extensions.conf, and then I have it Dial the appropriate Zap channel. eg:

exten => 111,1,dial(Zap/23,20)

111 being the FAX number in question. (obviously).

Is this the right thing to do?=A0 I constantly see in my CLI

"Fax detected on Zap/XX, but not FAX extension"

Should I be transferring these calls to the right Zap channel instead of
Dialing? Or should I be using that new fangled "ChannelRedirect" ?

Im pretty much lost on this.. any direction/advice is appreciated!!

Thank you!


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Re: [asterisk-users] PRI Splitter

[Craig Guy]

I had a look at mine and it has only relays for pins 1,2,4,5 - the other relay 
positions are on the PCB are not populated.  Maybe it has changed recently.

Craig

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Igor Hernandez
Sent: Thursday, 4 September 2008 7:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Splitter

Hy Craig,

Can you elaborate on that? In our setup we have it doing just that and
it works without a glitch.

Regards,

Igor H.

Craig Guy wrote:
> The FSV-4PFS as shipped will not switch Ethernet – it switches pins 1,2,4,5.
> 
>  
> 
> Craig
> 
>  
> 
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of *FailSafe
> Inc.
> *Sent:* Tuesday, 2 September 2008 11:27 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] PRI Splitter
> 
>  
> 
> Although the original topic of this thread has changed quite a bit, I
> wanted to point out that the "SPF" Product that you are discussing is
> quite similar to our product, the FSV-4PFS.  Ours is a 4 port device
> which can switch 4 T1/E1/J1/Ethernet or as many as 16 analog lines from
> a primary to a backup server.  It uses similar logic (power outage =
> failover server, loss of hearbeat = failover server) and also has a
> physical mechanical switch on the front of it which allows manual
> override switching to main or secondary server.
> 
>  
> 
> We also have addressed the 'clean startup' that was discussed a few
> posts back.  The switch will start and remain in 'failover mode' until
> such time as it receives a hearbeat or the physical switch is moved to
> the "main' position.  A failed main server can be restarted/repowered
> without bothering the backup server operation one bit - until you are
> ready to switch back to the main server.
> 
>  
> 
> http://www.failsafevoip.com/index.php?main_page=product_info&products_id=1
> 
> 
>  
> 
> 
> -- 
> FailSafeVOIP, Inc.
> "Safe is always better than failed"
> http://www.failsafevoip.com
> [EMAIL PROTECTED] 
> 
>  
> 
> On Tue, 2 Sep 2008 00:22:45 +0200, "Christian Victor" said:
> 
>> that when both servers power fail you have a problem no matter if the
> 
>> failover switch ist still working or not.
> 
>  
> 
> You've got that right my friend! :-)
> 
>  
> 
> On Tue, 2 Sep 2008 00:22:45 +0200, "Christian Victor" said:
> 
>> http://store.variantdistribution.com/category-s/49.htmVariant - one of
> 
>> Rhinos distributors and the only source I was able to find
> 
>> - quotes the card for US$ 700.
> 
>  
> 
> Strange.  I've seen this happen before where retailers will list
> 
> outrageously high prices for soon-to-be-released products.   For example
> 
> the SNOM KlarVoice handset.  MSRP is $32, but I've seen it advertised
> for $200!
> 
>  
> 
> http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice
> 
>  
> 
> I can say with confidence that the LIST price is US $350.  The street
> price will be considerably lower.  Frankly, if I were Snom or Rhino I'd
> be pretty cheezed off about this phenomenon.  After hearing the 'buzz'
> 
> about a new product such as this, I'd hate for customers to *decide*
> against it mistkenly believing this incorrect price.  I'd turn my nose
> at either of these two products for the incorrect prices I've seen
> advertised.
> 
>  
> 
> We're pretty stoked to have stumbled onto this product because it's
> brand new, and we've been looking for something like it for some time.
> 
>  
> 
> -Karl
> 
> 
> 
> 
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Re: [asterisk-users] Call-leg stays on MusicOnHold forever

[Jorge Mendoza]

Andreas,

We can't help, but just to say that after 2 weeks of debugging, we have
found yesterday that the one way audio experienced by the agents some
times, is related to hold function.

Jorge Mendoza

Andreas Brodmann wrote:
> Hi
>
> I have a strange behaviour; perhaps someone who had a similar issue
> can help.
>
> I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco
> Call-Manager 6.1 cluster.
> Two phones/users from the Cisco environment call extensions on the
> Asterisk.
>
> Phone 1 / Call 1 is parked on the asterisk using:
> exten => xyz,1,Answer()
> exten => xyz,n,Set(PARKEXTENSION=555)
> exten => xyz,n,Park()
>
> Phone 2 / Call 2 is picking it up:
> exten => xyz,1,Answer()
> exten => xyz,n,ParkedCall(555)
>
> so far so good, they can talk to each other.
>
> Now if one of them presses Hold, Asterisk will:
>
> [Sep  5 14:16:05] VERBOSE[6351] logger.c: -- Started music on
> hold, class 'default', on SIP/10.16.17.162-081bb720
> [Sep  5 14:16:05] VERBOSE[6351] logger.c: -- Stopped music on hold
> on SIP/10.16.17.162-081bb720
> [Sep  5 14:16:05] VERBOSE[6351] logger.c: -- Started music on
> hold, class 'default', on SIP/10.16.17.162-081bb720
>
> start - stop - start
> strange, but it works ...
>
> If the same user/phone now presses hold/resume so that they could
> talk to each other again Asterisk does:
>
> [Sep  5 14:16:07] VERBOSE[6351] logger.c: -- Stopped music on hold
> on SIP/10.16.17.162-081bb720
> [Sep  5 14:16:07] VERBOSE[6351] logger.c: -- Started music on
> hold, class 'default', on SIP/10.16.17.162-081bb720
>
> stops the music and starts it again ...
>
> now the guy who pressed hold at first can hear the other party, but
> the other party only hears music from Asterisk.
>
> Has anyone had a similar phenomenon?
>
> Regards,
>
> Andreas
>
>
> 
>
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Re: [asterisk-users] Transfers on AgentLogin()

[Mark Hamilton]

So, nobody?
How is Asterisk vying to become a bigtime key player in PBX systems when
some things are not documented, and one cannot get help on a mailing list or
irc (maybe because people don't know themselves)?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 31, 2008 4:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Transfers on AgentLogin()

I've tried the regular, xfer button on xlite, dial 100 (to transfer to the
queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(),
so it transfers the full persistent connection to the queue instead of the
call itself and this causes the transferring agent to logout.

Either that, or I'm doing something wrong. There is no documentation out
there so I don't know how it would work for AgentLogin(). 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: August 30, 2008 6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()

What did you try and how did it fail? Are you using the t option in queue?

On 8/30/08, Mark Hamilton <[EMAIL PROTECTED]> wrote:
> So, no answers or is this thread going to remain unanswered too?
>
>
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mark
Hamilton
> Sent: August 28, 2008 6:15 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Transfers on AgentLogin()
>
>
>
> Oh, by the way, the agent who will be doing the assisted transfer will be
> using eyebeam.
>
>
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mark
Hamilton
> Sent: August 28, 2008 5:54 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Transfers on AgentLogin()
>
>
>
> Hi,
>
>
>
> I have the same question as:
>
> http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html
>
> ..which like all important things was never answered.
>
>
>
> How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's
> just pure SIP/VoIP.
>
>
>
> Help please.
>
> Thanks,
>
> Mark.
>
>
>
>

-- 
Sent from Gmail for mobile | mobile.google.com

Matt Riddell
Director
VentureVoIP

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Re: [asterisk-users] Polycom BLF - multiple buddies

[Robert McNaught]

Seems that this got it working as suggested in the thread - thank you
all for replies.




I took out the attendant.uri option as you dont need it.  It seems to
be that you can set up a buddy watch for one endpoint using this
option - dont know exactly what this option is supposed to be.  I was
reading the Polycom SIP 2.2.2 admin guide, which suggested that the
above was only for Microsoft Live Communications Server, so I ignored
it...and used the attendant.uri option.

Robert



On Thu, Sep 4, 2008 at 9:34 PM, Lee, John (Sydney)
<[EMAIL PROTECTED]> wrote:
>> Sorry, needed to add one more note. To clarify, my agent phones have a
>> speed dial assigned for their login, and another to pause/unpause. I
>> could then use DEVSTATE to enable or disable the light next to their
>> speed dial button based on their status. I can't use it to update
>> anything on the LCD screen.
>
> James, very useful info especially about enable/disable the light next
> to the speed dial button which is exactly what I am after.  I am
> currently using 1.4.x and would be interested to know how this can be
> achieved.
>
>
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Re: [asterisk-users] Bridge 2 incoming calls

[Tim Panton]

On 5 Sep 2008, at 15:50, Steve Murphy wrote:

> On Fri, 2008-09-05 at 12:27 +0100, Tim Panton wrote:
>> I think I've forgotten something obvious
>>
>> I've got 2 incoming calls, I want to bridge them - how can I do  
>> this ?
>>
>> (assume I somehow know which calls should be paired up...)
>>
>> I could dump them both in a meetme - but that seems wasteful
>> as i _know_ there will only ever be 2 parties. (And I need DTMF
>> to flow through). I may want to record the bridged call, but that  
>> isn't
>> vital.
>>
>> I'm thinking of dialing chan_local with a call-id but I'm sure I
>> am missing something simpler.
>>
>>
>
> Not in 1.4, but in trunk,(and 1.6.x) there is a the Bridge manager
> command you can call via the manager interface, which takes two
> required args, the names of the two channels to bridge, and an
> optional arg, that will send a tone to the second channel.
>
> see main/features.c
>
> murf

Thats good to know.
Will the xml-over http manager interface be able to do it too? (pretty  
please?)

Tim.

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Re: [asterisk-users] Bridge 2 incoming calls

[Tim Panton]

I knew I'd forgotten something.
Doh!

On 5 Sep 2008, at 14:57, Andreas Brodmann wrote:


Tim,

you may want to try:

1) Park call 1
2) Pickup call 1 with call 2 (using ParkedCall)

Regards,

Andreas

2008/9/5 Tim Panton <[EMAIL PROTECTED]>
I think I've forgotten something obvious

I've got 2 incoming calls, I want to bridge them - how can I do this ?

(assume I somehow know which calls should be paired up...)

I could dump them both in a meetme - but that seems wasteful
as i _know_ there will only ever be 2 parties. (And I need DTMF
to flow through). I may want to record the bridged call, but that  
isn't

vital.

I'm thinking of dialing chan_local with a call-id but I'm sure I
am missing something simpler.



Tim.

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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI

[Steve Repo]

On Fri, Sep 5, 2008 at 8:16 PM, Tony Mountifield
<[EMAIL PROTECTED]> wrote:
> In article <[EMAIL PROTECTED]>,
> Russell Bryant <[EMAIL PROTECTED]> wrote:
>>
>> On Sep 3, 2008, at 8:32 PM, sean darcy wrote:
>>
>> > Great.
>> >
>> > But I'm still a little confused.
>> >
>> > Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
>>
>> No.  Asterisk 1.6.0 now _only_ supports DAHDI.
>>
>> > It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We
>> > can go back to this release of zaptel if we have problems with dahdi.
>> >
>> > Or if we go back to zaptel, do we go back to 1.6.0-beta9 also?
>>
>> You can upgrade directly to DAHDI.  However, if you have trouble with
>> DAHDI and need to go back to Zaptel, then I would go back to Asterisk
>> 1.4 instead of using an old beta of 1.6.0.  Many things have been
>> fixed since 1.6.0-beta9.
>
> If I'm installing a new system based on the latest Asterisk 1.4, should
> I use zaptel or dahdi with it? Which version?
>

AFAIK, Zaptel is no longer supported. I'd recommend dahdi if it's a new setup.

Steve

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Re: [asterisk-users] Bridge 2 incoming calls

[Steve Murphy]

On Fri, 2008-09-05 at 12:27 +0100, Tim Panton wrote:
> I think I've forgotten something obvious
> 
> I've got 2 incoming calls, I want to bridge them - how can I do this ?
> 
> (assume I somehow know which calls should be paired up...)
> 
> I could dump them both in a meetme - but that seems wasteful
> as i _know_ there will only ever be 2 parties. (And I need DTMF
> to flow through). I may want to record the bridged call, but that isn't
> vital.
> 
> I'm thinking of dialing chan_local with a call-id but I'm sure I
> am missing something simpler.
> 
> 

Not in 1.4, but in trunk,(and 1.6.x) there is a the Bridge manager 
command you can call via the manager interface, which takes two 
required args, the names of the two channels to bridge, and an 
optional arg, that will send a tone to the second channel.

see main/features.c

murf

> 
> Tim.
> 
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-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Russell Bryant <[EMAIL PROTECTED]> wrote:
> 
> On Sep 3, 2008, at 8:32 PM, sean darcy wrote:
> 
> > Great.
> >
> > But I'm still a little confused.
> >
> > Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
> 
> No.  Asterisk 1.6.0 now _only_ supports DAHDI.
> 
> > It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We
> > can go back to this release of zaptel if we have problems with dahdi.
> >
> > Or if we go back to zaptel, do we go back to 1.6.0-beta9 also?
> 
> You can upgrade directly to DAHDI.  However, if you have trouble with  
> DAHDI and need to go back to Zaptel, then I would go back to Asterisk  
> 1.4 instead of using an old beta of 1.6.0.  Many things have been  
> fixed since 1.6.0-beta9.

If I'm installing a new system based on the latest Asterisk 1.4, should
I use zaptel or dahdi with it? Which version?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Asterisk Crash

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Josiah Bryan <[EMAIL PROTECTED]> wrote:
> Well, at the time I wrote the AGI, fewestcalls wasn't an option (or at 
> least, I couldn't find it through googling or on the voip-info wiki).
> 
> Since then, the script has been in production use for 3+ years and I 
> havn't bothered to go back rework the dialplan. Sorry for the trouble 
> though.
> 
> However, it still begs the question, why does Dial seem to "fall 
> through" like that after the operator transfers the call? Is that 
> expected/designed behavior? If yes, Has that changed since the 1.0 days 
> of asterisk? If yes, Is there a switch that can turn that off?
> 
> Thanks for your patience with all these questions.

You might want to look at the j option to Dial.

In 1.0 days, a failed Dial would jump to priority n+101. Before 1.2 this
got changed so it didn't. Instead you were supposed to check ${DIALSTATUS}.
However, for compatibility, the old jump behaviour can be reinstated with
the j option.

The above applies when Dial is called from the dialplan. I have no idea
whether it applied when Dial was called from EXEC in AGI, but it's worth
a try.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Dear asterisk-users@lists.digium.com 79% OFF on Pfizer

[VIAGRA �]

Dear asterisk-users@lists.digium.com, Best Price Only Today.
http://kcq.diplike.com?mve


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[asterisk-users] svn branches for dhadi and its tools

[John covici]

Hi.  I want to use the new asterisk 1.4 with dahdi, but I would like
to know the svn branches for the dahdi, so I can use them that way --
much easier to keep up with bug fixes, etc.

Thanks.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] dahdi & tdm400p: no luck

[Tzafrir Cohen]

On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
> Tzafrir Cohen wrote:
> > On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
> >> As best i could figure it out, I've installed dahdi and rc4.
> >>
> >> My TDM400P doesn't answer fxo or fxs.
> >>
> >> /etc/dahdi/system.conf:
> >> loadzone   = us
> >> defaultzone=us
> >> fxoks=1,2
> >> fxsks=4
> > 
> > echocancel?
> > 
> 
> I thought that if you had hardware echocancel ( TDM400P does, doesn't 
> it? ), 

TDM400P doesn't. Do you mean TDM410P?

> setting the software echocanceller was irrelevant. In any event, 
> isn't mg2 the deefault?

No. You may have that impression from the configuration generated by
dahdi_genconf that adds it as a default (that is: generates an explicit
echocancel line for each channel) due to this limitation. That may
change in the future if system.conf will grow up its own default echo
canceller.

> 
>   I'll take the system down and change this, and dahdichan to 1,2 later 
> today, though again that wouldn't explain the lack of call pickup on the 
> _external_ line. show daahdi channels shows _no_ channels.  ( sigh)

And this still does not explain why you have not posted the output of:

 cat /proc/dahdi/*

;-)

> 
> And, I'm using 1.6.0-rc4.

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

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Re: [asterisk-users] Bridge 2 incoming calls

[Andreas Brodmann]

Tim,

you may want to try:

1) Park call 1
2) Pickup call 1 with call 2 (using ParkedCall)

Regards,

Andreas

2008/9/5 Tim Panton <[EMAIL PROTECTED]>

> I think I've forgotten something obvious
>
> I've got 2 incoming calls, I want to bridge them - how can I do this ?
>
> (assume I somehow know which calls should be paired up...)
>
> I could dump them both in a meetme - but that seems wasteful
> as i _know_ there will only ever be 2 parties. (And I need DTMF
> to flow through). I may want to record the bridged call, but that isn't
> vital.
>
> I'm thinking of dialing chan_local with a call-id but I'm sure I
> am missing something simpler.
>
>
>
> Tim.
>
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Re: [asterisk-users] G722 and Asterisk 1.6

[Olivier]

Beside Polycom, which hardphone vendor uses G.722.1 ?
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Re: [asterisk-users] dahdi & tdm400p: no luck

[sean darcy]

Tzafrir Cohen wrote:
> On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
>> As best i could figure it out, I've installed dahdi and rc4.
>>
>> My TDM400P doesn't answer fxo or fxs.
>>
>> /etc/dahdi/system.conf:
>> loadzone   = us
>> defaultzone=us
>> fxoks=1,2
>> fxsks=4
> 
> echocancel?
> 

I thought that if you had hardware echocancel ( TDM400P does, doesn't 
it? ), setting the software echocanceller was irrelevant. In any event, 
isn't mg2 the deefault?

  I'll take the system down and change this, and dahdichan to 1,2 later 
today, though again that wouldn't explain the lack of call pickup on the 
_external_ line. show daahdi channels shows _no_ channels.  ( sigh)

And, I'm using 1.6.0-rc4.

Thanks for the quick response.

sean


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[asterisk-users] FW: Vivox SLim

[Dean Collins]

I thought this blog post might interest some people on this list as
well.

http://deancollinsblog.blogspot.com/2008/09/vivox-slim.html 

 

Regards,

Dean Collins
[EMAIL PROTECTED] 

+1-212-203-4357 (New York) 
+61-2-9016-5642 (Sydney)
http://www.Cognation.net  



 

There has been some press in the voip space over the past few days about
the new Vivox SLim
  application.

Linden Labs and Vivox   partnered to create SLim
which is a discrete VOIP client that is meant to run alongside the
Second Life viewer allowing it to communicate voice calls with others
not necessarily running the Second Life application.


Vivox isn't really that big a deal Mexuar implemented a 168k java applet
using the open source Asterisk platform into Second Life about 18 months
ago.

It enabled voice calls in both a many to many 'open voice' conference
room but also the ability to make 'real world calls' onto the pstn
network either outbound or even inbound using the Asterisk servers PRI's
or Voip channels.

At one stage there was a mockup of a virtual cell phone you could use in
Second Life to make calls or link inbound calls to your real world cell
phone as well.


Cheers,
Dean Collins
www.Cognation.net   

--
http://deancollinsblog.blogspot.com/2008/09/vivox-slim.html 

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[asterisk-users] Call-leg stays on MusicOnHold forever

[Andreas Brodmann]

Hi

I have a strange behaviour; perhaps someone who had a similar issue
can help.

I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco Call-Manager
6.1 cluster.
Two phones/users from the Cisco environment call extensions on the Asterisk.

Phone 1 / Call 1 is parked on the asterisk using:
exten => xyz,1,Answer()
exten => xyz,n,Set(PARKEXTENSION=555)
exten => xyz,n,Park()

Phone 2 / Call 2 is picking it up:
exten => xyz,1,Answer()
exten => xyz,n,ParkedCall(555)

so far so good, they can talk to each other.

Now if one of them presses Hold, Asterisk will:

[Sep  5 14:16:05] VERBOSE[6351] logger.c: -- Started music on hold,
class 'default', on SIP/10.16.17.162-081bb720
[Sep  5 14:16:05] VERBOSE[6351] logger.c: -- Stopped music on hold on
SIP/10.16.17.162-081bb720
[Sep  5 14:16:05] VERBOSE[6351] logger.c: -- Started music on hold,
class 'default', on SIP/10.16.17.162-081bb720

start - stop - start
strange, but it works ...

If the same user/phone now presses hold/resume so that they could
talk to each other again Asterisk does:

[Sep  5 14:16:07] VERBOSE[6351] logger.c: -- Stopped music on hold on
SIP/10.16.17.162-081bb720
[Sep  5 14:16:07] VERBOSE[6351] logger.c: -- Started music on hold,
class 'default', on SIP/10.16.17.162-081bb720

stops the music and starts it again ...

now the guy who pressed hold at first can hear the other party, but the
other party only hears music from Asterisk.

Has anyone had a similar phenomenon?

Regards,

Andreas
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[asterisk-users] Grandstream Video Phones & Asterisk..

[Gordon Henderson]

Well, the recent talk about video phones and a project I've had lurking 
which has come to the top of the pile recently made me go out and buy a 
pair of Grandstream video phones.

And stack-me-sideways they "just work".

Amazing little boxes too (actually not that little with a 5.6" screen!)

They even have a web browser built-in with lots of "stuff" in the config 
screens I've not even looked at yet, but their actual configuration is 
easy if you've ever configured up another type of Grandstream in the past.

Oooh - built in digital photo frame too... (I like my new toys!) (2 x USB 
sockets on the back too)

Now I'll have to go and lookup what the pro's and con's of the various 
video codes it supports. The phones seem to be using H.263 by default 
which with the default settings of 15fps, etc. seems to use about 
128Kb/sec of bandwidth most of the time.

Incidentally, I'm using asterisk 1.2.30.

Trying to trunk a vido call over IAX doesn't seem to work... Maybe I 
shouldn't expect it to with 1.2.30 - what's worse though is that calling a 
non video phone over an IAX trunk results in horrible noise on the line - 
unless I disabled the video phone on the GXV3000 first. (Same goes for 
calling into a GXV3000 over an IAX trunk, so it suggests to me that I keep 
the video blocked all the time unless I really want to make a video call!)


Only using it over a LAN right now - but will wander down to a friends 
shortly. (As a remote friend seems to be struggling to make X-Lite work - 
audio is fine, but no video )-: He knows the camera, mic & speakers work 
as he uses it with Skype... X-Lite is such a PITA to configure though.

So all I can say right now is; Woo Hoo ;-)

Gordon
(trying to remember to stop picking his nose when on a call now)

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[asterisk-users] Bridge 2 incoming calls

[Tim Panton]

I think I've forgotten something obvious

I've got 2 incoming calls, I want to bridge them - how can I do this ?

(assume I somehow know which calls should be paired up...)

I could dump them both in a meetme - but that seems wasteful
as i _know_ there will only ever be 2 parties. (And I need DTMF
to flow through). I may want to record the bridged call, but that isn't
vital.

I'm thinking of dialing chan_local with a call-id but I'm sure I
am missing something simpler.



Tim.

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Re: [asterisk-users] Gateway errors

[voip crazy]

Thank you Hatem, I will try it now

Thanks

VoipCrazy

2008/9/2 hatem moiz <[EMAIL PROTECTED]>:
> you can do the following in sip .conf file
>
> register => username:[EMAIL PROTECTED]
>
> and after that write the configuration for the user:
>
> [ user ]
> username =
> host =
> qualify =
> secret =
>
> and so on, do this in the first of sip.conf file
>
> Best Regards
>
> On Mon, Sep 1, 2008 at 11:32 AM, voip crazy <[EMAIL PROTECTED]> wrote:
>>
>> Hatem,
>>
>> I cannot understan exactly what you told me.
>> Could you try to explain that in other words. Better if you could post
>> an example of this SIP trunk.
>>
>> thanks in advance.
>>
>> Voip Crazy
>>
>>
>>
>> 2008/9/1 hatem moiz <[EMAIL PROTECTED]>:
>> > Asterisk is looking for a SIP trunk if you have recorded the usage of
>> > SIP
>> > trunks all it need is to find 1 SIP trunk,
>> >
>> > To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1
>> > and
>> > make sure that it is the first one in sip.conf file. OR you can make a
>> > sip
>> >
>> > trunk to ATA in the same lan and also be sure that it is the first trunk
>> > in
>> > sip.conf .
>> >
>> > On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez <[EMAIL PROTECTED]> wrote:
>> >>
>> >> Thats strange, have you checked that you're not having issues with your
>> >> router? Can you reach all the boxes in your lan while you are
>> >> experiencing this downtime?
>> >>
>> >> voip crazy wrote:
>> >> > When I say extensions, I say extensions in the lan not in wan
>> >> >
>> >> > Thanks.
>> >> >
>> >> > VoipCrazy.
>> >> >
>> >> > 2008/9/1 Igor Hernandez <[EMAIL PROTECTED]>:
>> >> >> Hello,
>> >> >>
>> >> >> By people do you mean people in the lan or external users?
>> >> >>
>> >> >> Regards,
>> >> >>
>> >> >> --
>> >> >> Igor Hernandez
>> >> >> Escape Communications
>> >> >> http://www.escapetel.com
>> >> >>
>> >> >>
>> >> >> voip crazy wrote:
>> >> >>> Hello list,
>> >> >>>
>> >> >>> I have an asterisk instalation with a bad internet connection cause
>> >> >>> this connection is down sometimes.
>> >> >>> When the connection is down and asterisk cannot get internet
>> >> >>> connection. All the extensions log out from the asterisk machine,
>> >> >>> and
>> >> >>> nobody can make any call.
>> >> >>>
>> >> >>> ¿Why if internet connection is down asterisk stops working
>> >> >>> correctly?
>> >> >>> ¿How could I solve that?
>> >> >>>
>> >> >>> Thansk.
>> >> >>>
>> >> >>> VoipCrazy
>> >> >>> ___
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>> >> >>> --
>> >> >>>
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>> >> >>
>> >> >>
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>> >>
>> >>
>> >>
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