Hi All, I have an asterisk 1.6.1.6 with PSTN lines connected to it via digium cards.When I try to call a SIP phone the caller ID is not displayed and is shown as Unavailable/Out of area.Need the expert advice in resolution of the same.the extension.conf is attached... Warm Regards Venugopal G ************************************************************************ ************************************************************************ ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100209/92b712b2/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: extensions.conf.presentation Type: application/octet-stream Size: 1276 bytes Desc: extensions.conf.presentation Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100209/92b712b2/attachment.obj
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