Hi, I am restricting the caller ID from the SONUS class 5 server towards the asterisk. But Asterisk is not considering it to be restricted and it shows the caller ID of the SIP phone on the DAHDI lines. Kindly let me know what all logs.config info are required for this?
I am attaching the chan_dahdi.conf and extensions.conf.Let me know if you need any more of the config. Warm Regards Venugopal G Cell : +91-99723-99437 ************************************************************************ ************************************************************************ ************************************************* Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 ************************************************************************ ************************************************************************ ************************************************* -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Tzafrir Cohen Sent: Thursday, March 11, 2010 3:09 PM To: [email protected] Subject: Re: [asterisk-users] CallerID presented in Asterisk On Thu, Mar 11, 2010 at 04:18:21PM +0800, Gopalakrishnaiyer Venugopal-Q16770 wrote: > Hi, > > I am not using FreePBX.I am using Asterisk 1.6.1.6 and TDM800P cards > for analog lines. This is an analog line. Either caller ID is sent or it isn't. Even if it is sent, you can choose to ignore it. Could you please be more specific about what you expect Asterisk to do? -- Tzafrir Cohen icq#16849755 jabber:[email protected] +972-50-7952406 mailto:[email protected] http://www.xorcom.com iax:[email protected]/tzafrir -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
extensions.conf
Description: extensions.conf
[channels] group = 1 echocancel = yes signalling = pri_cpe switchtype = euroisdn context = Internal channel = 1-15,17-31 usecallerid=yes hidecallerid=no hidecalleridname = no callwaiting=no callwaitingcallerid=yes cidsignalling=bell cidstart=ring threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echotraining=yes rxgain=0.0 txgain=0.0 callprogress=no callerid=asreceived pickupgroup=1 pridialplan=national ;FXS module group = 5 signalling = fxo_ks context = Internal channel = 33-39 usecallerid = yes hidecallerid = no callerid = PSTN <8001234000> callwaiting = no
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
