Anybody tried with Trunk between asterisk and freeswitch...?
Hi, Any body tried with Trunkig between freeswitch and asterisk . If any body tried and its working fine . Please share the details. Regards Samba DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.
sambasivarao Vemula wrote:
Hi,
Any body tried with Trunkig between freeswitch and asterisk .
If any body tried and its working fine .
Please share the details.
Why would it not? We use a SIP trunk from our freeswitch to asterisk. Asterisk being used ONLY as a ZAP gateway. Works fine. Its like setting up any other SIP trunk. -- Faraz R Khan Zivios::Open source Enterprise Management www.zivios.org
http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk /b On Nov 5, 2008, at 12:31 AM, sambasivarao Vemula wrote:
Hi, Any body tried with Trunkig between freeswitch and asterisk . If any body tried and its working fine . Please share the details. Regards Samba
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It's rather simple - Setup a sip user on asterisk with username/password - Setup a gateway in freeswitch with the asterisk user credentials (ip, username, password of asterisk) - Define a route in the dialplan (e.g. default.xml) to route certain numbers to the asterisk gateway e.g. <extension name="Long Distance -Asterisk"> <condition field="destination_number" expression="^(0[2-9]\d{4,13})$"> <action application="set" data="effective_caller_id_number="/> <action application="export" data="sip_secure_media=true"/> <action application="bridge" data="sofia/gateway/asterisk/$1@asterisk.domain"/> </condition> </extension> You should already be able to make outgoing calls via asterisk. Best regards Peter sambasivarao Vemula schrieb:
Hi,
Any body tried with Trunkig between freeswitch and asterisk .
If any body tried and its working fine .
Please share the details.
Regards
Samba
DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.
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Was this not helpful? http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk /b On Nov 6, 2008, at 8:23 AM, Peter P GMX wrote:
It's rather simple - Setup a sip user on asterisk with username/password - Setup a gateway in freeswitch with the asterisk user credentials (ip, username, password of asterisk) - Define a route in the dialplan (e.g. default.xml) to route certain numbers to the asterisk gateway e.g. <extension name="Long Distance -Asterisk"> <condition field="destination_number" expression="^(0[2-9]\d{4,13})$"> <action application="set" data="effective_caller_id_number="/> <action application="export" data="sip_secure_media=true"/> <action application="bridge" data="sofia/gateway/asterisk/$1@asterisk.domain"/> </condition> </extension>
You should already be able to make outgoing calls via asterisk.
Best regards Peter
sambasivarao Vemula schrieb:
Hi,
Any body tried with Trunkig between freeswitch and asterisk .
If any body tried and its working fine .
Please share the details.
Regards
Samba
DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.
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participants (4)
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Brian West
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Faraz Khan
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Peter P GMX
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sambasivarao Vemula