Hi All, The debug/verbose are also attached along with the extensions.conf Warm Regards Venugopal G ************************************************************************ ************************************************************************ ************************************************* -----Original Message----- From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of peterpet Sent: Tuesday, February 09, 2010 7:30 PM To: asterisk-ss7 at lists.digium.com Cc: Singh Deepesh-SNGD001 Subject: Re: No CallerID Display for calls orginating from Asterisk Gopalakrishnaiyer Venugopal-Q16770 wrote: > 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 > ********************************************************************** > ********************************************************************** > **************************************************** > > Put debug/verbose here from these call -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -------------- next part -------------- A non-text attachment was scrubbed... Name: extensions.conf Type: application/octet-stream Size: 1207 bytes Desc: extensions.conf Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100209/42a42d11/attachment-0001.obj -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: asterisk_log.txt Url: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100209/42a42d11/attachment-0001.txt
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