search for: psnt

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2007 Aug 06
1
help: H323 and SIP
...well. Now I have to perform call from h323 to sip and viceversa. How can I do it ???? I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone: - PSTN ==> CiscoVoiceGW(h323) ==> Asterisk ==> SIP - SIP ==> Asterisk ==> CiscoVoiceGW(h323) ==> PSNT I've now idea how to configure asterisk (conf file) and softphones... Thanks for all! -- AxR. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070806/c8c01970/attachment.htm
2006 Jan 31
3
ZAP <--> sip(polycom301) can not hear each other
please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bi...
2008 Mar 01
7
ASTCC installation error install: invalid user `apache'
I am attempting a fresh install of ASTCC on Ubuntu. Getting install invalid user as bellow. Has any one seen this? Can some one help out? /usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi >/dev/null DBI