-----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 Litomyšl
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-----

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to