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...ated Asterisk a s a NT mode [ 212.226050] wctdm24xxp 0000:0a:01.0: xhfc: Configuring port 0 span 1 in NT mode with termination resistance ENABLED. It could help by checking parameters on dahdi-channels if signalling = bri_cpe_ptmp # The signalling for TE mode Best regards Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card. Mc GRATH Ricardo

Roberto Could you provide more details about Panasonic PBX test? model unit and configuration details? Best regards Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

Can be done calls from each system? How about to capture data with Wireshark? I have experiences Asterisk with Panasonic with H323 without any problem. Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

hi, anybody know how to make a sip trunk between trixbox and panasonic kx tde 100? i've tried, but always failed when calling from trixbox to panasonic. thanks,,, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130312/1dd5170b/attachment.htm>

...for preventing to let to PSTN start call handler process during asterisk starting process or stop. Another point it seems on your log information a call have been answered, so it seems is working. Good luck PD: 0x00 it doesn't mean idle state (it mean force release) Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

...#66 (new software configurations file etc.) when device boot up. If user need to configures directory and others "My Allworx manager " it should made through system ip address or domain name. I think better idea to choose another brand, save your time! good luck. Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

...ng for openvox B400P card. Each port could be configured as CO or extension port as Point-Point or PTMP in both mode. Regarding to dial timeout these it becomes from ISDN dial mode overlap or En-bloc, by the way you can close dialling number by using # key. Best regards Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Tim At first should take a look to cable pinout (RAD documents) as pin 1,2, Transmit (output) and 4, 5 Receive (input) for Digium card you should use a straight cable (try to test with new cable one too). Second check Dahdi configuration parameters, use dahdi commands as; dahdi show status, service dahdi restart and check result, (could a mistake on parameter value on system.conf). Mc