Re: [asterisk-users] Questions on hardware or software-based echo cancellation

[Olivier]

2012/1/12, Kevin P. Fleming :
> On 01/12/2012 06:39 AM, Olivier wrote:
>> Hi,
>>
>> I'm having some questions related to echo cancellation configuration
>> on a Digium board enabled systems (B410P, TE420, TE420B, ) for
>> cases when a hardware ech canceller is present or not.
>>
>> I read in TEXXX manual that when setting echocancel=yes in
>> chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
>> cancellation was enabled.
>>
>> 1. I'm correct thinking that it is then impossible to switch from
>> hardware to software echo can without removing the VPMOCT64 module
>> itself ?
>> 2. Does the same also apply to HA8 and its VPMOCT032 module ?
>
> With DAHDI 2.6 (and possibly 2.5), it is possible to override the
> configuration and apply a software echo canceller to a channel even if
> it has a hardware one. With prior versions, yes, the echo cancellation
> module would have to be physically removed (or disabled using a
> parameter to the kernel module).

Then, maybe a line mentioning that in the next User Manual edition
would be perfect.
Thanks for replying.

>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] FAX Installation in Asterisk

[Ruben Rögels]

Am 12.01.2012 18:50, schrieb mahesh katta:
> I was search for free license but for this Digium require purchase any
> Hardware then they can provide Free License.
> But I have no Digium Device , I am using Grand stream FXO Gateway and
> Asterisk.1.8.XX .
> I was connected like
> PSTN==>FXOGateway==>Asterisk(FXO configure through IP)
> 
> If anything wrong please correct me.

Hi Mahesh,

the FreeFax for asterisk is really free and not bound to digium
hardware, but it is limited to one concurrent fax session. At least you
should be able to try if fax receiving is possible with this setup. As
far as I can see, it should work with your setup.

The URL I posted leads you to the FreeFAX for Asterisk Module.

best regards,
Ruben

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Re: [asterisk-users] FAX Installation in Asterisk

[mahesh katta]

On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels <
ruben.roeg...@jumping-frog.org> wrote:

> Am 12.01.2012 18:50, schrieb mahesh katta:
> > I was search for free license but for this Digium require purchase any
> > Hardware then they can provide Free License.
> > But I have no Digium Device , I am using Grand stream FXO Gateway and
> > Asterisk.1.8.XX .
> > I was connected like
> > PSTN==>FXOGateway==>Asterisk(FXO configure through IP)
> >
> > If anything wrong please correct me.
>
> Hi Mahesh,
>
> the FreeFax for asterisk is really free and not bound to digium
> hardware, but it is limited to one concurrent fax session. At least you
> should be able to try if fax receiving is possible with this setup. As
> far as I can see, it should work with your setup.
>
> The URL I posted leads you to the FreeFAX for Asterisk Module.
>
> Sir,Its done.I receive the FAX.Thank you sir.
One more thing sir if I sent at a time multiple fax to this is it receive.
can you clarify me.
scenario is I have PRI line of 30 channels. one Boarding no.
if I send this is it receive the fax at a time with single free license.


>  best regards,
> Ruben
>
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[asterisk-users] Queue member is permanently BUSY

[Raj Mathur (राज माथुर)]

Hi, we have a queue with some 20 agents.  Agents are defined statically 
in queues.conf (in an include, actually).  One of the agents is showing 
up as "NOT AVAILABLE" in "queue show..." when his client is disconnected 
from Asterisk.  However, the moment his system gets connected, his 
status in "queue show..." changes to "BUSY", even though he's not taking 
any calls.

We've tried both soft and hard phones as clients for this agent, but the 
problem remains the same.  Any help appreciated.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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[asterisk-users] odbc storage for video message

[shalu dhamija]

Hello, 

I am using ODBC storage for the message deposited through voicemail() 
application. Everything works fine for the audio message if the format given in 
voicemail .conf is any audio format . e.g. 

in voicemail.conf give the format as 

format=gsm 



This case works fine. 



But if I give the following format in voicemail.conf 

format=h263|gsm 



And no video codec is sent from the phone, t hen the audio message is not 
stored in the database. I am getting the following warning: 



[Jan 13 15:14:17] VERBOSE[27656] app_voicemail.c: -- Recording was 0 
seconds long but needs to be at least 1 - abandoning 



Although the nessage deposited is of duration of around 10-15 seconds. 



 And If i send audio and video codecs both then one message gets stored in the 
database but I am not able to retrive that message. 



Please suggest waht changes needs to be done for enabling the storage of video 
messages in database as well. 




Regards, 
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[asterisk-users] dialplan problem : not including context

[Jonas Kellens]

Hello,

I have the following in dialplan :


[TrunkAccounts]

exten => 32380837,1,GoTo(01,32380837,1)
exten => 32380838,1,GoTo(01,32380838,1)
exten => 32380839,1,GoTo(01,32380839,1)

[CheckOnNet]

include => TrunkAccounts



But when a call for 32380837 enters CheckOnNet, it is not found. How come ??



Kind regards,
Jonas.
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[asterisk-users] SIP hardphone with dual gigabit ethernet ports

[Vieri]

Hi,

I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All the 
ones I've seen only have dual 10/100Mbps ethernet ports (eg. Grandstream 
products).

Any suggestions?

Thanks,

Vieri


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Re: [asterisk-users] SIP hardphone with dual gigabit ethernet ports

[isrlgb]

The new snom 7 series and maybe the 8 series have Gig ethernet 
-Original Message-
From: Vieri 
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 13 Jan 2012 04:45:12 
To: 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] SIP hardphone with dual gigabit ethernet ports

Hi,

I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All the 
ones I've seen only have dual 10/100Mbps ethernet ports (eg. Grandstream 
products).

Any suggestions?

Thanks,

Vieri


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Re: [asterisk-users] FAX Installation in Asterisk

[Steve Underwood]

On 01/13/2012 05:17 PM, mahesh katta wrote:




On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels 
> wrote:


Am 12.01.2012 18:50, schrieb mahesh katta:
> I was search for free license but for this Digium require
purchase any
> Hardware then they can provide Free License.
> But I have no Digium Device , I am using Grand stream FXO
Gateway and
> Asterisk.1.8.XX .
> I was connected like
> PSTN==>FXOGateway==>Asterisk(FXO configure through IP)
>
> If anything wrong please correct me.

Hi Mahesh,

the FreeFax for asterisk is really free and not bound to digium
hardware, but it is limited to one concurrent fax session. At
least you
should be able to try if fax receiving is possible with this setup. As
far as I can see, it should work with your setup.

The URL I posted leads you to the FreeFAX for Asterisk Module.

Sir,Its done.I receive the FAX.Thank you sir.
One more thing sir if I sent at a time multiple fax to this is it 
receive. can you clarify me.

scenario is I have PRI line of 30 channels. one Boarding no.
if I send this is it receive the fax at a time with single free license.

best regards,
Ruben

Remove the Digium FAX module and install SpanDSP. Then the number of 
FAXes you can receive at once will only be limited by the speed of your 
hardware.


Steve


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Re: [asterisk-users] dialplan problem : not including context

[Doug Lytle]

Jonas Kellens wrote:

I have the following in dialplan :


[TrunkAccounts]


dialplan show TrunkAccounts

Make sure the sort order is what you're expecting.

Doug


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Re: [asterisk-users] dialplan problem : not including context

[Jonas Kellens]

On 01/13/2012 02:23 PM, Doug Lytle wrote:


Jonas Kellens wrote:

I have the following in dialplan :


[TrunkAccounts]


dialplan show TrunkAccounts

Make sure the sort order is what you're expecting.

Doug


Hello,

The order is correct for as far as I'm sure.

[TrunkAccounts]

exten => 32380837,1,GoTo(01,32380837,1)
exten => 32380838,1,GoTo(01,32380838,1)
exten => 32380839,1,GoTo(01,32380839,1)

[CheckOnNet]

include => TrunkAccounts

exten => _321[0-3],1,GoTo(context1,${EXTEN},1)

exten => 3214,1,GoTo(context2,${EXTEN},1)

exten => _.,1,NoOp()
exten => _.,n,Return()


This is what I see on the CLI :

/[Jan 13 14:30:01] -- Executing [s@macro-uit789:47] 
Gosub("SIP/yoc1-5a3c", "CheckOnNet,//32380837,1") in new stack
[Jan 13 14:30:01] -- Executing [32380837@CheckOnNet:1] 
NoOp("SIP/yoc1-5a3c", "") in new stack
[Jan 13 14:30:01] -- Executing [32380837//@CheckOnNet:2] 
Return("SIP/yoc1-5a3c", "") in new stack/



So the context TrunkAccounts is not included.

Do you know why ?


Jonas.
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Re: [asterisk-users] dialplan problem : not including context

[Andreas Sikkema]

On 1/13/12 2:32 PM, Jonas Kellens wrote:
> So the context TrunkAccounts is not included.
> 
> Do you know why ?

Does reloading the dialplan (dialplan reload) give any useful output
relating to these two contexts?

-- 
Andreas Sikkema

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Re: [asterisk-users] dialplan problem : not including context

[Jonas Kellens]

On 01/13/2012 02:37 PM, Andreas Sikkema wrote:

On 1/13/12 2:32 PM, Jonas Kellens wrote:

So the context TrunkAccounts is not included.

Do you know why ?

Does reloading the dialplan (dialplan reload) give any useful output
relating to these two contexts?


I include this context in 2 other contexts :

[Jan 13 14:19:12] VERBOSE[4220] config.c: [Jan 13 14:19:12]   == Parsing 
'/etc/asterisk/extensions.conf': [Jan 13 14:19:12] VERBOSE[4220] 
config.c: [Jan 13 14:19:12]   == Found

...
[Jan 13 14:19:12] VERBOSE[4220] pbx.c: [Jan 13 14:19:12] -- 
Including context 'TrunkAccounts' in context 'PROVIDERin'

...
[Jan 13 14:19:12] VERBOSE[4220] pbx.c: [Jan 13 14:19:12] -- 
Registered extension context 'CheckOnNet' (0x2aaacc40d260) in local 
table 0xd0ba610; registrar: pbx_config
[Jan 13 14:19:12] VERBOSE[4220] pbx.c: [Jan 13 14:19:12] -- 
Including context 'TrunkAccounts' in context 'CheckOnNet'

...

Nothing special here it seems...


Everything works fine when including context 'TrunkAccounts' in context 
'PROVIDERin'. Here it functions as expected.


But it does not work the same in context 'CheckOnNet'.

Extra question : is there a difference between the "context" PROVIDERin 
and the "procedure" (sub) CheckOnNet ??



Kind regards,

Jonas.


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Re: [asterisk-users] dialplan problem : not including context

[Doug Lytle]

Jonas Kellens wrote:
Everything works fine when including context 'TrunkAccounts' in 
context 'PROVIDERin


dialplan showPROVIDERin

Doug

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Re: [asterisk-users] dialplan problem : not including context

[Jonas Kellens]

On 01/13/2012 02:59 PM, Doug Lytle wrote:


Jonas Kellens wrote:
Everything works fine when including context 'TrunkAccounts' in 
context 'PROVIDERin


dialplan showPROVIDERin

Doug



Meaning ?

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Re: [asterisk-users] dialplan problem : not including context

[Doug Lytle]

Jonas Kellens wrote:
Meaning ? 


Meaning I want to see the dialplan order of that context.  I'm guessing 
that's your inbound context.  With includes that also include sub-contexts.


Usually, there is something ordered differently then expected.

Also, what version of Asterisk?

Doug


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Re: [asterisk-users] dialplan problem : not including context

[Jonas Kellens]

On 01/13/2012 03:07 PM, Doug Lytle wrote:


Jonas Kellens wrote:
Meaning ? 


Meaning I want to see the dialplan order of that context.  I'm 
guessing that's your inbound context.  With includes that also include 
sub-contexts.


Usually, there is something ordered differently then expected.

Also, what version of Asterisk?

Doug


Asterisk 1.6.2.22

It is impossible to post all of this information... However, this is the 
context CheckOnNet :


..
  '_32962' =>  1. GoTo(solutions,${EXTEN},1) [pbx_config]
  '_3295[0-9]' => 1. GoTo(step,${EXTEN},1)   [pbx_config]
  '_.' =>   1. NoOp() 
[pbx_config]
2. Return()   
[pbx_config]
  Include =>'TrunkAccounts'   
[pbx_config]


-= 252 extensions (253 priorities) in 1 context. =-


Does this mean the Return() comes before Asterisk looks into the context 
[TrunkAccounts] ??



Kind regards,
Jonas.

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Re: [asterisk-users] dialplan problem : not including context

[Doug Lytle]

Jonas Kellens wrote:
Does this mean the Return() comes before Asterisk looks into the 
context [TrunkAccounts] ??


No, I believe the includes are read first, but the order in important.

Since you may be matching against another context that may cause 
failure.  For example, I have the following in a internal context:


  Include =>'analog-extensions'   
[pbx_config]
  Include =>'sip-utilities'   
[pbx_config]
  Include =>'internal-extensions' 
[pbx_config]
  Include =>'dial-local'  
[pbx_config]
  Include =>'dial-ld' 
[pbx_config]
  Include =>'incoming'
[pbx_config]
  Include =>'fall-through'
[pbx_config]


If I had fall-through first, it'd cause lots of issues.

Doug


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Re: [asterisk-users] odbc storage for video message

[Matthew Jordan]

- Original Message - 

> From: "shalu dhamija" 
> To: "asterisk-dev" , "asterisk-users"
> 
> Sent: Friday, January 13, 2012 3:56:45 AM
> Subject: [asterisk-users] odbc storage for video message

> Hello,
> I am using ODBC storage for the message deposited through voicemail()
> application. Everything works fine for the audio message if the
> format given in voicemail .conf is any audio format. e.g.
> in voicemail.conf give the format as
> format=gsm

> This case works fine.

> But if I give the following format in voicemail.conf
> format=h263|gsm

The first format specified in the format list is the 'primary' format the 
voicemails are stored in.  Some preference is given to the first format for 
certain operations.  With IMAP and ODBC backends, *only* the first format 
specified is used.

> And no video codec is sent from the phone, then the audio message is
> not stored in the database. I am getting the following warning:

> [Jan 13 15:14:17] VERBOSE[27656] app_voicemail.c: -- Recording was 0
> seconds long but needs to be at least 1 - abandoning

> Although the nessage deposited is of duration of around 10-15
> seconds.

Since no video was negotiated, nothing is recorded in the h263 file that 
voicemail attempts to create.  Hence the reason why the recording was dropped.  
If you specify gsm as your first codec, you would (at least on the file system) 
record the voicemails.  However, you won't store the h263 file on the ODBC 
backend - the gsm file will be stored in the database.

As it is, storing video messages for either IMAP / ODBC backends in 
app_voicemail is going to run into similar problems, as the video / audio are 
stored in separate files.  This is a limitation of app_voicemail that has not 
been addressed - as it is, storing video related messages is best supported 
using the file system.

> And If i send audio and video codecs both then one message gets
> stored in the database but I am not able to retrive that message.

> Please suggest waht changes needs to be done for enabling the storage
> of video messages in database as well.

> Regards,
> Shalu

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Re: [asterisk-users] dialplan problem : not including context

[Jonas Kellens]

On 01/13/2012 04:22 PM, Doug Lytle wrote:


Jonas Kellens wrote:
Does this mean the Return() comes before Asterisk looks into the 
context [TrunkAccounts] ??


No, I believe the includes are read first, but the order in important.

Since you may be matching against another context that may cause 
failure.  For example, I have the following in a internal context:


  Include =>'analog-extensions'   
[pbx_config]
  Include =>'sip-utilities'   
[pbx_config]
  Include =>'internal-extensions' 
[pbx_config]
  Include =>'dial-local'  
[pbx_config]
  Include =>'dial-ld' 
[pbx_config]
  Include =>'incoming'
[pbx_config]
  Include =>'fall-through'
[pbx_config]


If I had fall-through first, it'd cause lots of issues.

Doug


Hello,

there is only one include-statement...


Jonas.


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[asterisk-users] Stuck DAHDI Channels

[Antonio Modesto]

Good afternoon,

My current Asterisk is a FreeBSD 8.2-STABLE machine with
asterisk18-1.8.7.1 installed from ports collection and . My old asterisk
was a pre-configured system (Brazilian Disc-OS), and on both systems i
have the same problem, sometimes some of my DAHDI channels get stuck,
here is my chan_dahdi.conf


#cat chan_dahdi.conf
[trunkgroups]
; define any trunk groups

[channels]
; hardware channels
; default
language=pt_BR
hanguponswitchpolarity=yes
;usecallerid=yes
;callerid=asreceived
;cidsignalling=dtmf
;cidstart=dtmf
hidecallerid=no
callwaiting=yes
;usecallingpres=yes
;callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=yes
;useincomingcalleridonzaptransfer=yes ( quando transferir a ligacao, o
numero original sera enviado )
;relaxdtmf=yes

;canpark=yes
;cancallforward=yes
;callreturn=yes
;echocancel=yes
;echocancelwhenbridged=yes
;group=1
;callgroup=1
;pickupgroup=1
context=from_clients2
signalling=fxs_ks

; define channels
channel => 1
channel => 2
channel => 3
channel => 4
channel => 5

context=from_celular
signalling=fxs_ks
channel => 9
channel => 10
channel => 11
channel => 12
channel => 13
channel => 14
channel => 15
channel => 16

;context=default
;switchtype=national
;signalling=fxo_ls
;hidecallerid=no
;echocancel=yes
;echocancelwhenbridged=yes
;rxgain=0.0
;txgain=0.0
;immediate=no

The first five channels work fine but some of the channels that are in
the "from_celular" context which are reserved for callback (these
channels are plugged in a celular interface, the callback is configured
in the own interface) get stuck sometimes, the strange is that "core
show channels" shows that one line is calling another, but it happens
without explication. You can think that we are calling the callback
number by another channel and starting a loop, but unfortunately it's
not what is happening =).

There is something that I think that may be causing this problem, we use
these lines in Brazil, where the signaling is DTMF, and my lines are not
configured to use it, they are using the asterisk default, which is FSK,
i don't know if it could cause it, it's just an option.


I know that it's a difficult problem to solve with only the information
above, but if somebody has an idea of what may be happening i would be
grateful.


Regards.
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Re: [asterisk-users] SIP hardphone with dual gigabit ethernet ports

[Carlos Alvarez]

On Fri, Jan 13, 2012 at 5:45 AM, Vieri  wrote:

> Hi,
>
> I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All
> the ones I've seen only have dual 10/100Mbps ethernet ports (eg.
> Grandstream products).
>
> Any suggestions?
>

Cisco, Polycom, Yealink, and MANY others make them.

http://lmgtfy.com/?q=sip+phone+gigabit

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Questions on hardware or software-based echo cancellation

[Kevin P. Fleming]

On 01/13/2012 02:12 AM, Olivier wrote:

2012/1/12, Kevin P. Fleming:

On 01/12/2012 06:39 AM, Olivier wrote:

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?


With DAHDI 2.6 (and possibly 2.5), it is possible to override the
configuration and apply a software echo canceller to a channel even if
it has a hardware one. With prior versions, yes, the echo cancellation
module would have to be physically removed (or disabled using a
parameter to the kernel module).


Then, maybe a line mentioning that in the next User Manual edition
would be perfect.


Sure, but you have to understand that the user manuals for our board 
products are typically only updated when the board itself gets changed; 
we don't actually deliver DAHDI with the boards, and 
installation/configuration instructions for DAHDI are primarily included 
in the manual for user convenience. Users should be aware that they 
could easily be out of date, and not include all options that are 
currently offered (although if the user manual's instructions become 
incorrect, we'll update the manual). Regardless, I'll mention this to 
the people who manage those products. Thanks!


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] dialplan problem : not including context

[Doug Lytle]

Jonas Kellens wrote:

there is only one include-statement


Then I don't know.  I am still on 1.4.x and my PRI context contains all 
that I'm matching against (No sub contexts).


Doug


--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] dialplan problem : not including context

[Doug Lytle]

Doug Lytle wrote:



Then I don't know.  I am still on 1.4.x and my PRI context contains 
all that I'm matching against (No sub contexts).




One thing does come to mind;  the inbound call is coming into your s 
extension and then your doing a gosub, in which case, you might be 
matching against s.


Put in a few NoOP statements to find out what EXTEN or ARG1 is.

Doug


--

Ben Franklin quote:

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deserve neither Liberty nor Safety."


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Re: [asterisk-users] dialplan problem : not including context

[Administrator TOOTAI]

Le 13/01/2012 14:32, Jonas Kellens a écrit :

On 01/13/2012 02:23 PM, Doug Lytle wrote:


Jonas Kellens wrote:

I have the following in dialplan :


[TrunkAccounts]


dialplan show TrunkAccounts

Make sure the sort order is what you're expecting.

Doug


Hello,

The order is correct for as far as I'm sure.

[TrunkAccounts]

exten => 32380837,1,GoTo(01,32380837,1)
exten => 32380838,1,GoTo(01,32380838,1)
exten => 32380839,1,GoTo(01,32380839,1)

[CheckOnNet]

include => TrunkAccounts

exten => _321[0-3],1,GoTo(context1,${EXTEN},1)

exten => 3214,1,GoTo(context2, ${EXTEN} ,1)

exten => _.,1,NoOp()
exten => _.,n,Return()


Are you sure about your _. exten? Typo in the mail? It means 9 
and more digits but your extensions are 8 digits ...


Include are always treated *after* context command. If _. is 
right, something is wrong with Asterisk as it should treat 
TrunkAccounts. If _XXX. (8 digits or more) is what you have in 
yourdialplan, than the behavior of Asterisk is OK


Try

[TrunkAccounts]

exten => 32380837,1,GoTo(01,32380837,1)
exten => 32380838,1,GoTo(01,32380838,1)
exten => 32380839,1,GoTo(01,32380839,1)

[TrunkNotTreated]

exten => _.,1,NoOp()
exten => _.,n,Return()

[CheckOnNet]

include => TrunkAccounts
include => TrunkNotTreated

exten => _321[0-3],1,GoTo(context1,${EXTEN},1)
exten => 3214,1,GoTo(context2, ${EXTEN} ,1)

[...]

--
Daniel

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Re: [asterisk-users] FAX Installation in Asterisk

[mahesh katta]

Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
222, Arunvihar,Sector-28, Noida 201301
GSM +91.3 45699 | Phone +91.12.0431.0581
Web http://www.buzzworks.com



On Fri, Jan 13, 2012 at 6:27 PM, Steve Underwood  wrote:

> On 01/13/2012 05:17 PM, mahesh katta wrote:
>
>
>>
>>
>> On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels <
>> ruben.roegels@jumping-frog.**org > ruben.roegels@jumping-**frog.org >>
>> wrote:
>>
>>Am 12.01.2012 18:50, schrieb mahesh katta:
>>> I was search for free license but for this Digium require
>>purchase any
>>> Hardware then they can provide Free License.
>>> But I have no Digium Device , I am using Grand stream FXO
>>Gateway and
>>> Asterisk.1.8.XX .
>>> I was connected like
>>> PSTN==>FXOGateway==>Asterisk(**FXO configure through IP)
>>>
>>> If anything wrong please correct me.
>>
>>Hi Mahesh,
>>
>>the FreeFax for asterisk is really free and not bound to digium
>>hardware, but it is limited to one concurrent fax session. At
>>least you
>>should be able to try if fax receiving is possible with this setup. As
>>far as I can see, it should work with your setup.
>>
>>The URL I posted leads you to the FreeFAX for Asterisk Module.
>>
>> Sir,Its done.I receive the FAX.Thank you sir.
>> One more thing sir if I sent at a time multiple fax to this is it
>> receive. can you clarify me.
>> scenario is I have PRI line of 30 channels. one Boarding no.
>> if I send this is it receive the fax at a time with single free license.
>>
>>best regards,
>>Ruben
>>
>>  Remove the Digium FAX module and install SpanDSP. Then the number of
> FAXes you can receive at once will only be limited by the speed of your
> hardware.
>
>which spandsp version is compatible with asterisk-1.4.27 or 1.8.XX.
   And which application Receive the fax .I mean "RecieveFax" in Dialplan.
   above Dial plan is it work for this.

Steve
>
>
>
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[asterisk-users] Preços por serviços e equipamento

[Cláudio Duarte]

Senhores alguém pode me passar os valores aproximados para realizar essas
configurações,
uma amigo não sabe como e quanto cobrar para fazer as configurações para um
cliente e fica me perguntando, eu cobro por hora, e o senhores ?

O serviços são:

- Configuração do PABX digital
- Configuração de URA
- Configuração de redirecionamento para celular por operadora

ATT



-- 


Cláudio Duarte
(61) 3037 - 2015
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[asterisk-users] Queue option 'R'

[georg]

Hi all,

I've got a queue with two agents and everything works great so far.
However, I would like to use the option 'R', which "stops moh and rings
once an agent is ringing (Asterisk Trunk)" [1] according to voip-info.org

However, using this option changes nothing. The moh is played until an
agent picks up the phone.

I'm using the Asterisk package provided by the current Debian stable:
Asterisk 1.6.2.9-2+squeeze4

Any hints?

Thanks,
Georg

[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue


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[asterisk-users] Sporadic one way audio problem

[georg]

Hi all again,

I've got a problem with sporadic one way audio calls, which means
sometimes I can't hear the calling party (call is established, but audio
is missing). Today I received ~90 calls, one of them got this problem.

I've got two networks involved, without NAT:

- 192.168.1.X, in there one nic of my server and all the phones
- a private net to my provider, in there a nic of my server and the voice
switch of my provider

My sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
language=de
allowguest=no
;echocancel=yes
;echotraining=yes
alwaysauthreject=yes
disallow=all
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=XXX.XXX.X.X/29
permit=192.168.1.0/24
;jbenable=yes
;jbforce=yes
;jbmaxsize=20
;jbresyncthreshold=1000
tos=0x10
directmedia=no
nat=no
directrtpsetup=no

[provider]
type=peer
host=XXX.XXX.X.X
insecure=port,invite
context=X
dtmfmode=rfc2833
directmedia=no
nat=no
directrtpsetup=no
;qualify=300

[one-phone]
[10]
type=peer
context=X
secret=XX
host=dynamic
;qualify=300
directmedia=no
nat=no
directrtpsetup=no
dtmfmode=inband

Any help greatly appreciated!

Thanks,
Georg


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Re: [asterisk-users] Queue option 'R'

[bakko]

Hello,

I think this option work only with Asterisk 1.8.X

On the Asterisk 1.8.X CHANGES files:

* Added 'R' option to app_queue.  This option stops moh and indicates 
ringing
  to the caller when an Agent's phone is ringing.  This can be used to 
indicate
  to the caller that their call is about to be picked up, which is nice 
when

  one has been on hold for an extened period of time.

Regards 



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Re: [asterisk-users] Queue option 'R'

[georg]

Hi,

> I think this option work only with Asterisk 1.8.X

Ah yes, I see. Now "Asterisk Trunk" in the description makes sense...

Thanks,
Georg


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[asterisk-users] Asterisk as UAC: How to put call OnHold

[Johannes Zweng]

Hi!

Maybe I am missing something or am a little blind at the moment, but I
didn't find out how asterisk can place a call on hold when acting as user
agent client to another SIP server.

Scenario:
--
Asterisk registers to another SIP server (provider) as user agent.
An inbound call from this other SIP server comes in and arrives at asterisk.
Asterisk performs some actions in the dialplan and should place the call on
hold after some time, so that the caller only hears the on hold music from
my provider (not streamed by my Asterisk).

Technically speaking I want asterisk to send a re-INVITE message containing
an updated SDP body with the attribute "a=sendonly" or "a=inactive" added
so that the SIP server of my provider (where Asterisk is registered to as
user) will recognize that the call should be placed on hold.


A good example of what I want to achieve is presented in Section 2.1 of RFC
5359 (Session Initiation Protocol Service Examples) (
http://tools.ietf.org/html/rfc5359#section-2.1) where "Bob" would be my
Asterisk (as UAC), "Alice" is the external caller and "Proxy" is the
provider's SIP server.


Question:
--
Is there any way to perform this from the dialplan or by means of the
manager API? Is there an application like "Hold"?


Kind regards and greetings from Austria,
John :-)
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[asterisk-users] asterisk problem sip

[Carlos Rojas]

Hi everybody

I have been presenting a periodic problem, do not know if anyone listed has
happened something similar,I'm using the asterisk, asterisk-1.6.2.13, in
different locations works well, but every so often fails, hangs on Asterisk
server or simply asterisk, SIP requirements do not answer, apparently
unaware of the dialplan as well as voicemails, etc, but keeps the
registry, apparently the service is up but theusers, call and doesn't make
calls.


Asterisk is restarted, and solve the problem, I think it may be a bug in
asterisk, someone has had a similar problem?


Regards

Carlos
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