No voice in one direction, SIP, call manager

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No voice in one direction, SIP, call manager
\ Martin Edlman (31 Mar 2008)

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Subject:	No voice in one direction, SIP, call manager
Group:	Asterisk-users
From:	Martin Edlman
Date:	31 Mar 2008

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Hello,

I have a problem with Asterisk 1.4.x and the call manager. When I
originate a call by the call manager or by a dot-call file only the
calling party can hear the called party, not vice versa. When I dial the
same number directly from the SIP phone (Cisco 7960) everything is OK.
The same configuration worked with Asterisk 1.2 last week before
switching to 1.4.

There is a gateway (Patton) to the telecom operator communicating with
the Asterisk via SIP.
I've checked the SIP channels with "sip show channels" and it's the
same when the call is originated by the phone or the call manager.

Is there something special to be set to make call manager originated
calls working again?

Dot-call used:

# calling party
Channel: SIP/CiscoPhone
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: sip
Priority: 1
# called party
Extension: +420phonenumber

Call manager commands used:

Action: login
Username: call_manager
Secret: call_password
Events: off

Action: originate
Channel: SIP/CiscoPhone
Context: sip
Priority: 1
Timeout: 30000
CallerID: Martin Edlman <38>
Exten: +420phonenumber

- --
Ragards,

Martin Edlman
Fortech, spol. s r.o,
Ropkova 51, 57001 Litomyšl
Public GPG key: http://edas.visaci.cz/#gpgkeys

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