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dean collins

2005-Feb-23 14:50 UTC

[Asterisk-Users] Vonage <---> Asterisk Working Config!

Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk@home sourceforge forum, you'll probably be able to work out
how to set it up from there.


Cheers,
Dean


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hello All,

I know this question has already been answered but for some reason I am
failing to receive inbound calls.

I got my Asterisk@home 0.6 setup for inbound and outbound and I can make
successful outbound calls. My Asterisk is registered with Vonage
successfully also. 

The only problem I am facing is inbound calls. I enabled the "sip
debug"
and
I can see inbound calls coming in but then asterisk destroys it for some
unknown reason. 

Does anyone have a working config for inbound call?

Thanks Jay for all your help

Nitesh


_______________________________________________
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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Nitesh Divecha

2005-Feb-23 16:04 UTC

head link

[Asterisk-Users] Vonage <---> Asterisk Working Config!

Hey Dean,

I followed your instructions and moved forward abit... Before it used to say
SIP 2.0 404 Not Found. But now with your settings I am getting SIP/2.0 407
Proxy Authentication Required.

And yea for some unknown reason the TRUNK name and USER CONTEXT has to be
same don't know why... 

But still no success for inbound calls... 

This is my sip.conf

[vonage]
username=<<VonageDID>>
type=friend
secret=blah
port=5061
nat=yes
insecure=very
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=inband
context=from-pstn
canreinvite=no
auth=md5

[vonage-out]
username=<<VonageDID>>
type=friend
secret=blah
port=5061
nat=yes
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
auth=md5

Any help would be appreciated.... Thanks Dean!

Nitesh



-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of dean collins
Sent: Wednesday, February 23, 2005 1:50 PM
To: nitesh@vipernetworks.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk@home sourceforge forum, you'll probably be able to work out
how to set it up from there.


Cheers,
Dean


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hello All,

I know this question has already been answered but for some reason I am
failing to receive inbound calls.

I got my Asterisk@home 0.6 setup for inbound and outbound and I can make
successful outbound calls. My Asterisk is registered with Vonage
successfully also. 

The only problem I am facing is inbound calls. I enabled the "sip
debug"
and
I can see inbound calls coming in but then asterisk destroys it for some
unknown reason. 

Does anyone have a working config for inbound call?

Thanks Jay for all your help

Nitesh


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Spam detection software, running on the system "zeus.avanzada7.com",
has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  Hi Nitesh, check out my config that I have for the 
  Faktortel config in the asterisk@home sourceforge forum, you'll 
  probably be able to work out how to set it up from there. Cheers, Dean 

Content analysis details:   (1.1 points, 5.0 required)

 pts rule name              description
---- ----------------------
--------------------------------------------------
 0.1 FORGED_RCVD_HELO       Received: contains a forged HELO
 1.0 RCVD_IN_NJABL_PROXY    RBL: NJABL: sender is an open proxy
                            [68.174.172.170 listed in combined.njabl.org]

dean collins

2005-Feb-23 16:21 UTC

head link

[Asterisk-Users] Vonage <---> Asterisk Working Config!

Nitesh do you have the register trunk portion set up as well?

For faktortel is was 

xxusernamexx:xxpasswordxx@iax.faktortel.com.au



-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 6:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hey Dean,

I followed your instructions and moved forward abit... Before it used to
say
SIP 2.0 404 Not Found. But now with your settings I am getting SIP/2.0
407
Proxy Authentication Required.

And yea for some unknown reason the TRUNK name and USER CONTEXT has to
be
same don't know why... 

But still no success for inbound calls... 

This is my sip.conf

[vonage]
username=<<VonageDID>>
type=friend
secret=blah
port=5061
nat=yes
insecure=very
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=inband
context=from-pstn
canreinvite=no
auth=md5

[vonage-out]
username=<<VonageDID>>
type=friend
secret=blah
port=5061
nat=yes
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
auth=md5

Any help would be appreciated.... Thanks Dean!

Nitesh



-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of dean
collins
Sent: Wednesday, February 23, 2005 1:50 PM
To: nitesh@vipernetworks.com; Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk@home sourceforge forum, you'll probably be able to work out
how to set it up from there.


Cheers,
Dean


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hello All,

I know this question has already been answered but for some reason I am
failing to receive inbound calls.

I got my Asterisk@home 0.6 setup for inbound and outbound and I can make
successful outbound calls. My Asterisk is registered with Vonage
successfully also. 

The only problem I am facing is inbound calls. I enabled the "sip
debug"
and
I can see inbound calls coming in but then asterisk destroys it for some
unknown reason. 

Does anyone have a working config for inbound call?

Thanks Jay for all your help

Nitesh


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Spam detection software, running on the system "zeus.avanzada7.com",
has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  Hi Nitesh, check out my config that I have for the 
  Faktortel config in the asterisk@home sourceforge forum, you'll 
  probably be able to work out how to set it up from there. Cheers, Dean


Content analysis details:   (1.1 points, 5.0 required)

 pts rule name              description
---- ----------------------
--------------------------------------------------
 0.1 FORGED_RCVD_HELO       Received: contains a forged HELO
 1.0 RCVD_IN_NJABL_PROXY    RBL: NJABL: sender is an open proxy
                            [68.174.172.170 listed in
combined.njabl.org]




_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
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Nitesh Divecha

2005-Feb-24 12:53 UTC

head link

[Asterisk-Users] Vonage <---> Asterisk Complete Config

Hello Asterisk Users, 

After Brain storming for couple of hours, days, and weeks, finally got
Asterisk to work with Vonage for Inbound and Outbound calls.

Requirement: -
1) Vonage Softphone account
2) Asterisk
3) Couple of SIP Phones

Here is my sip.conf

[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = <<Local IP>>    ; Address to bind to (all addresses on
machine)
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=<<External IP>>
localnet=<<Local IP>>
localmask=<<Local mask>>
nat=yes

register=<<VonageDID>>:password@sphone.vopr.vonage.net:5061/202

[vonage-out]
username=<<VonageDID>>
type=friend
secret=<<password>>
port=5061
nat=yes
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
auth=md5

[vonage202]
username=<<VonageDID>>
type=friend
secret=<<password>>
port=5061
nat=yes
insecure=very
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=inband
context=from-pstn
canreinvite=no
auth=md5

Here is my extension.conf

[ext-did]
exten => <<VonageDID>,1,Goto(ext-local,202,1) 
or 
exten => <<VonageDID>>,1,Goto(aa_1,s,1) If you are sending the
call to IVR.

For some this configuration might vary as my Asterisk is behind NAT. 

Asterisk Rocks!!! Enjoy

Many thanks to Jay & Dean

Neel

Randy Johnson

2005-Feb-25 05:26 UTC

head link

[Asterisk-Users] Vonage <---> Asterisk Complete Config

I thought Vonage did not allow this?


-Randy


Nitesh Divecha wrote:
>Hello Asterisk Users, 
>
>After Brain storming for couple of hours, days, and weeks, finally got
>Asterisk to work with Vonage for Inbound and Outbound calls.
>
>Requirement: -
>1) Vonage Softphone account
>2) Asterisk
>3) Couple of SIP Phones
>
>Here is my sip.conf
>
>[general]
>port = 5060           ; Port to bind to (SIP is 5060)
>bindaddr = <<Local IP>>    ; Address to bind to (all addresses
on machine)
>context=incoming
>disallow=all
>allow=ulaw
>allow=alaw
>allow=g729
>allow=g723
>externip=<<External IP>>
>localnet=<<Local IP>>
>localmask=<<Local mask>>
>nat=yes
>
>register=<<VonageDID>>:password@sphone.vopr.vonage.net:5061/202
>
>[vonage-out]
>username=<<VonageDID>>
>type=friend
>secret=<<password>>
>port=5061
>nat=yes
>host=sphone.vopr.vonage.net
>fromuser=<<VonageDID>>
>fromdomain=sphone.vopr.vonage.net
>dtmfmode=rfc2833
>auth=md5
>
>[vonage202]
>username=<<VonageDID>>
>type=friend
>secret=<<password>>
>port=5061
>nat=yes
>insecure=very
>host=sphone.vopr.vonage.net
>fromuser=<<VonageDID>>
>fromdomain=sphone.vopr.vonage.net
>dtmfmode=inband
>context=from-pstn
>canreinvite=no
>auth=md5
>
>Here is my extension.conf
>
>[ext-did]
>exten => <<VonageDID>,1,Goto(ext-local,202,1) 
>or 
>exten => <<VonageDID>>,1,Goto(aa_1,s,1) If you are sending
the call to IVR.
>
>For some this configuration might vary as my Asterisk is behind NAT. 
>
>Asterisk Rocks!!! Enjoy
>
>Many thanks to Jay & Dean
>
>Neel
>
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>  
>

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