Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: New... on March 22, 2010, 06:52:54 AM
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Scenario: SIP server connected with mediant. Need to make a call from SIP extn to mediant and send that call back to another SIP extn (same SIP server)
Any pointers would be helpful.
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Hi there!
It's there simple task.
1. create outbound trunk or Extension DN that point to Mediant
2. On Mediant create translate rule to rewrite destination number from dialed by your agent to new internal number
3 On Mediant create routing rule to route created number to you next SIP server.
WBR.
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Hi Timur,
Thanks for the input. Am new to Mediant. I understand the SIP configuration (pt.1). Can you help me where exactly the rules to be defined in Mediant? Also, in this case of loop back, do I need to connect the trunks using RJ48?
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[quote author=RJO link=topic=5421.msg23584#msg23584 date=1269244294]
Hi Timur,
Thanks for the input. Am new to Mediant. I understand the SIP configuration (pt.1). Can you help me where exactly the rules to be defined in Mediant? Also, in this case of loop back, do I need to connect the trunks using RJ48?
[/quote]
well, coz mediant did't have abiliti to pass-throgh sip2sip conversation you may need create an external hardware loop to connect 2 pstn ports together.
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Hi Timur,
I have connected loop back connector from trunk1 to 2. Channels are now in active state. But when call is made (trunkphonenumber@GW-IP:5060), am getting the response for INVITE message as 486 Busy Here (cause=3) :(
Am now trying to make a call from one SJphone to GW and receive the call in another SJPhone (SJPhone setup in 2 different IPs).
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[quote author=RJO link=topic=5421.msg23669#msg23669 date=1269577091]
Hi Timur,
I have connected loop back connector from trunk1 to 2. Channels are now in active state. But when call is made (trunkphonenumber@GW-IP:5060), am getting the response for INVITE message as 486 Busy Here (cause=3) :(
Am now trying to make a call from one SJphone to GW and receive the call in another SJPhone (SJPhone setup in 2 different IPs).
[/quote]
check the hardware options - especially network timer source
check the routing and translate rule
try to debug the call