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Any have any success using a patton smartnode 4118/js/eiu fxs gateway with asterisk? We we're able to get the unit to register with asterisk, but when trying to place a call, no codec was compatible, even though I had all of the following enabled on the patton ... # G.711 A-Law/?-Law (64kbps) # G.726 (ADPCM 40, 32, 24, 16 kpbs) # G.723.1 (5.3 or 6.3 kbps) # G.729ab (8kbps) the link to this

Hi All, I face some problems when I try to introduce Asterisk to my customers / friends. They are not convince as they are currently using Skype and asking me what is/are the different between this two. Does anyone in the community can provide such a comparison chart? What's your opinion ? Thanks and Regards, Stephen

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hi All, How can I lock the extension in Asterisk? For example , my extension is 1000 and I am away for business trip. I want to lock my extension during my absence. Can it be done in Asterisk? regards, Stephen

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley

Hi , Does anyone encounter this problem ? We have installed Asterisk at Site A and have 128k Frame Relay over to Site B. We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A. We are using Ulaw at Site A and G729 at Site B. When the calls are originated from Site A to Site B, party at Site A can hear Site B voice clearly and no breaking up voice. But Site B user hears Site A