asterisk users - Mar 2008 - No voice in one direction, SIP, call manager

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

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 Litomy?l
Public GPG key: http://edas.visaci.cz/#gpgkeys

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (GNU/Linux)
Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org

iD8DBQFH8KRoqmMakYm+VJ8RAh/gAKCsObn2hmsvuMqkrsnp9RJoYRBKNQCfSJzv
rEkCQaLp6e0GOknasykg3K0=zaws
-----END PGP SIGNATURE-----