Re: [asterisk-users] Skype for Asterisk, Skype For SIP

2010-07-19 Thread Brad Finberg 
I have been using SiSky Enterprise Edition to integrate Skype with asterisk. 
You can even call saved skype users from your asterisk system, by creating 
speed dials in SiSky. Unfortunately it is not a free product but it is very 
reasonable.


Thank you,
Brad Finberg


- Original Message -
From: Alejandro Imass a...@p2ee.org
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc:
Date: Sunday, July 18 2010 8:57 AM
Subject: Re: [asterisk-users] Skype for Asterisk, Skype For SIP
On Sun, Jul 18, 2010 at 7:48 AM, Vieri rentor...@yahoo.com wrote:
 Hi,

 I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 
 things:

 1) allow any Asterisk SIP extension to call any Skype user. I do not need 
 to call landlines via Skype.


I think this is _explicitly_ not supported in the Skype for SIP docs.

 2) allow Internet Skype users to call my Asterisk PBX Skype user and 
 route the call to a specific Asterisk SIP extension.


Here is how it goes from my experience with Skype: each SIP channel
will cost you about $5 a month, regardless if you have a landline
number with them or not. Your account will be assigned a special Skype
number 99x . With that number a Skype user can call you
and it will be free. You _cannot_ call Skype users from your PBX, as I
stated above, this is an explicit no-no in the docs. If you want to
make calls from your PBX to landlines you have to buy Skype credit
just like you would with a regular skype client. If you want
land-lines to call your PBX you need to purchase a skype number which
about $60 a year.


 At first, I thought it would be simple and free. However, correct me if I'm 
 wrong but the Skype user I can use within the Asterisk PBX cannot be the 
 standard type (used by eg. desktop Skype applications) but needs to be 
 created by the Skype User Manager for Business Solutions. I believe this has 
 a price although Skype For SIP Open Beta seems to be free until Q4 2010.

I think you can associate existing skype users to your Business
Solutions manager but I still don't understand exactly how or why this
is useful, and I don't think it has to do with you being able to call
any of them from your PBX. Then again I haven't paid much attention to
that and perhaps you have more insight into this.

 Has anyone found a way to make pure Internet user-to-user Skype/SIP calls 
 via Asterisk (no PSTN involved) for free?

As I said above, once you have purchased your SIP channel you can make
free calls to your PBX using the special number but it's only INBOUND
AFAIK.

Best,
Alejandro Imass



 Thanks,

 Vieri





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[asterisk-users] DAHDI TDM440E still has echo on bridged connections

2009-09-18 Thread Brad Finberg 

Hello,

Strangely i purchase a TDM440E with the echo canceller onboard and I still 
receive a horrible echo and i'm only
using bridged connections between DAHDI/4 and DAHDI/1.  I turned of echo 
cancellation on bridged connections which seemed to help alittle bit.  I ran 
fxotune -i5 and setup fxotune -s to apply settings on startup, which has 
helpped but there is still echo on the begining of each call. Any idea's as to 
why there would be an echo at the beginning of a bridged conversation with echo 
cancellation turned off?

Also this only happens on incomming calls not outgoing calls


Thank you,
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Re: [asterisk-users] call transfer using DTMF

2009-07-14 Thread Brad Finberg 

Yes,
In the features.conf under featuremap you need the blindtransfer un-commented
blindxfer = ## 
Then in your extensions.conf you need to have at least a capital T
exten = example,1,Dial(ZAP/4/12345,,T)
Then during the call you can press ## and asterisk will say transfer.
Then dial in the extension you want to transfer too.

Thank you,
Brad Finberg


- Original Message -
From: Michael as...@nettrust.co.nz
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc:
Date: Tuesday, July 14 2009 11:07 PM
Subject: [asterisk-users] call transfer using DTMF
Is there a way to transfer a call, while in the middle of the call, using 
DTMF?

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Re: [asterisk-users] notifyringing=no does not work

2009-04-21 Thread Brad Finberg 

Hello,

If anybody has any idea's to where I should start looking to fix the below 
subscription problem.  If there is another mailing list I should post this to 
please let me know.

Thank you,
Brad Finberg


- Original Message -
From: Brad Finberg b...@finberghouse.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc:
Date: Thursday, April 9 2009 9:42 AM
Subject: notifyringing=no does not work
Hello,

I have been trying to get my Grandstream GXP2000 phones to stop showing ringing 
state on monitored extensions. But no matter where I put notifyringing=no 
asterisk still sends the ringing state to the phones. Is this a bug I should 
report or is there another way around it.

Here is how i have my subscriptions setup:

extensions.conf
[demo]
exten = 6100,hint,SIP/100
exten = 6101,hint,SIP/101
exten = 6102,hint,SIP/102
exten = 6103,hint,SIP/103
exten = 6104,hint,SIP/104
exten = 6105,hint,SIP/105
exten = _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten = _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten = _1XX,3,VoiceMail(${ext...@default,u)
exten = _1XX,104,VoiceMail(${ext...@default,b)

sip.conf
[general]
allowsubscribe=yes 
;subscribecontext = default 
notifyringing=no 
notifyhold=yes 
;limitonpeers=yes

[100]
type=peer
context=demo
callerid=Back Office 100
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1

Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103

Thank you,
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Re: [asterisk-users] notifyringing=no does not work

2009-04-16 Thread Brad Finberg 

Anybody have any idea's

Thank you,
Brad Finberg


- Original Message -
From: Brad Finberg b...@finberghouse.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc:
Date: Thursday, April 9 2009 9:45 AM
Subject: [asterisk-users] notifyringing=no does not work
Hello,

I have been trying to get my Grandstream GXP2000 phones to stop showing ringing 
state on monitored extensions. But no matter where I put notifyringing=no 
asterisk still sends the ringing state to the phones. Is this a bug I should 
report or is there another way around it.

Here is how i have my subscriptions setup:

extensions.conf
[demo]
exten = 6100,hint,SIP/100
exten = 6101,hint,SIP/101
exten = 6102,hint,SIP/102
exten = 6103,hint,SIP/103
exten = 6104,hint,SIP/104
exten = 6105,hint,SIP/105
exten = _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten = _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten = _1XX,3,VoiceMail(${ext...@default,u)
exten = _1XX,104,VoiceMail(${ext...@default,b)

sip.conf
[general]
allowsubscribe=yes 
;subscribecontext = default 
notifyringing=no 
notifyhold=yes 
;limitonpeers=yes

[100]
type=peer
context=demo
callerid=Back Office 100
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1

Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103

Thank you,
Brad Finberg

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[asterisk-users] notifyringing=no does not work

2009-04-09 Thread Brad Finberg 

Hello,

I have been trying to get my Grandstream GXP2000 phones to stop showing ringing 
state on monitored extensions.  But no matter where I put notifyringing=no 
asterisk still sends the ringing state to the phones.  Is this a bug I should 
report or is there another way around it.

Here is how i have my subscriptions setup:

extensions.conf
[demo]
exten = 6100,hint,SIP/100
exten = 6101,hint,SIP/101
exten = 6102,hint,SIP/102
exten = 6103,hint,SIP/103
exten = 6104,hint,SIP/104
exten = 6105,hint,SIP/105
exten = _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten = _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten = _1XX,3,VoiceMail(${ext...@default,u)
exten = _1XX,104,VoiceMail(${ext...@default,b)

sip.conf
[general]
allowsubscribe=yes  
;subscribecontext = default
notifyringing=no   
notifyhold=yes
;limitonpeers=yes

[100]
type=peer
context=demo
callerid=Back Office 100
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1

Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103

Thank you,
Brad Finberg___
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