radhakrishnan.vijay@wipro.com
2005-Jan-02 23:51 UTC
[Asterisk-Users] Configuration details for Asterisk interaction with Vocal
I have seen a number of people in this newsgroup asking for information
regarding asterisk interworking with Vocal. I was able to configure
Vocal and Asterisk so that calls originating from vocal can land on an
extension in Asterisk. I would like to share this info with the group
The scenario that I tested was as follows.
A call was originated from extn. 1001 on Vocal and the call was made to
land on extension 12456 hanging from an asterisk pbx registered with
vocal. Vocal was configured such that, if the dialed digits are 8500,
the call will be routed to asterisk to which will handle the call on
extension 12346 hanging off asterisk
Configuration on Vocal's side
------------------------------------
All calls made to 8500 are forwarded to asterisk. This is achieved by
adding the following entry to the dial plan of Vocal using the provgui
as mentioned below:
Add a new entry in the dial plan as shown below
Key
-----
^sip:8500
Contact
---------
^sip:8500@<ip-address:port>
ipaddress:port is where the asterisk pbx is listening for sip messages.
Also make a ummy entry for extension 8500 in Vocal (so that vocal thinks
this is an extn. Connected to vocal).
Configuration Changes in asterisk
-----------------------------------
A register entry as shown below is added to the [general] section of the
sip.conf file in the asterisk server
register =8500:password@vocal/12346
This would register the extension 8500 with the vocal server using the
vocal tag. All calls received from vocal would terminate in extension
1246.
The following is the information contained in the vocal tag.
[vocal]
type=friend ; either "friend" (peer+user),
"peer" or
"user"
callerid=Test 1 <12346>
host=10.117.4.236 ; we have a static but private IP
address
port=5065
This indicates that the calls will be received from the Vocal server
running on host 10.117.4.236 on port 5065 (port where Marshall server is
running).
The following details are included in the extensions.conf file so that
calls originating from vocal can be answered by the extension 12346.
exten => 12346,1,NoOp(.call for .${EXTEN})
exten => 12346,2,Dial(SIP/${EXTEN},60,tr)
exten => 12346,3,Congestion
The following is the configuration information for the extension 12346
(Xlite UA client running on Windows) in the sip.conf file.
[12346] ; X-Lite client 12346
type=friend
secret=blah
auth=md5
nat=no ; we assume clients are not behind NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=inband
callerid="Test 1" <12346>
disallow=all
allow=gsm ; add whatever other codecs we fancy
context=test1 ; use a context that exists ;-)
A corresponding entry for the context test1 is required in
extensions.conf to complete configuration of extension 12346
[test1]
exten => _[123456789]XXXX,1,NoOp(.call for .${EXTEN})
exten => _[123456789]XXXX,2,Dial(SIP/${EXTEN},60,tr)
exten => _[123456789]XXXX,3,Congestion
This should be suffecient for you to land all calls originating from
Vocal with dialed digits 8500 to land on extn 12346 in asterisk.
Tip:
To ensure that asterisk has correctly registered with vocal just run the
show sip registry from the asterisk console to show all the peers in the
network
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Jim Van Meggelen
2005-Jan-03 00:33 UTC
[Asterisk-Users] Configuration details for Asterisk interactionwith Vocal
Please put this in the Wiki (HYPERLINK
"http://www.voip-info.org"www.voip-info.org). It is valuable
information
and will reach a wider audience there.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
radhakrishnan.vijay@wipro.com
Sent: January 3, 2005 1:52 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Configuration details for Asterisk
interactionwith Vocal
I have seen a number of people in this newsgroup asking for information
regarding asterisk interworking with Vocal. I was able to configure
Vocal and Asterisk so that calls originating from vocal can land on an
extension in Asterisk. I would like to share this info with the group
The scenario that I tested was as follows.
A call was originated from extn. 1001 on Vocal and the call was made to
land on extension 12456 hanging from an asterisk pbx registered with
vocal. Vocal was configured such that, if the dialed digits are 8500,
the call will be routed to asterisk to which will handle the call on
extension 12346 hanging off asterisk
Configuration on Vocal?s side
------------------------------------
All calls made to 8500 are forwarded to asterisk. This is achieved by
adding the following entry to the dial plan of Vocal using the provgui
as mentioned below:
Add a new entry in the dial plan as shown below
Key
-----
^sip:8500
Contact
---------
^sip:8500@<ip-address:port>
ipaddress:port is where the asterisk pbx is listening for sip messages.
Also make a ummy entry for extension 8500 in Vocal (so that vocal thinks
this is an extn. Connected to vocal).
Configuration Changes in asterisk
-----------------------------------
A register entry as shown below is added to the [general] section of the
sip.conf file in the asterisk server
register =8500:password@vocal/12346
This would register the extension 8500 with the vocal server using the
vocal tag. All calls received from vocal would terminate in extension
1246.
The following is the information contained in the vocal tag.
[vocal]
type=friend ; either "friend" (peer+user),
"peer" or
"user"
callerid=Test 1 <12346>
host=10.117.4.236 ; we have a static but private IP
address
port=5065
This indicates that the calls will be received from the Vocal server
running on host 10.117.4.236 on port 5065 (port where Marshall server is
running).
The following details are included in the extensions.conf file so that
calls originating from vocal can be answered by the extension 12346.
exten => 12346,1,NoOp(.call for .${EXTEN})
exten => 12346,2,Dial(SIP/${EXTEN},60,tr)
exten => 12346,3,Congestion
The following is the configuration information for the extension 12346
(Xlite UA client running on Windows) in the sip.conf file.
[12346] ; X-Lite client 12346
type=friend
secret=blah
auth=md5
nat=no ; we assume clients are not behind NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=inband
callerid="Test 1" <12346>
disallow=all
allow=gsm ; add whatever other codecs we fancy
context=test1 ; use a context that exists ;-)
A corresponding entry for the context test1 is required in
extensions.conf to complete configuration of extension 12346
[test1]
exten => _[123456789]XXXX,1,NoOp(.call for ..${EXTEN})
exten => _[123456789]XXXX,2,Dial(SIP/${EXTEN},60,tr)
exten => _[123456789]XXXX,3,Congestion
This should be suffecient for you to land all calls originating from
Vocal with dialed digits 8500 to land on extn 12346 in asterisk.
Tip:
To ensure that asterisk has correctly registered with vocal just run the
show sip registry from the asterisk console to show all the peers in the
network
Confidentiality Notice
The information contained in this electronic message and any attachments
to this message are intended
for the exclusive use of the addressee(s) and may contain confidential
or privileged information. If
you are not the intended recipient, please notify the sender at Wipro or
Mailadmin@wipro.com immediately
and destroy all copies of this message and any attachments.
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