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Hello, I have X100P card but exten => _9X.,1,Dial(Zap/1/${EXTEN:1}) does not work. The error is app_dial.c:803 dial_exec: Unable to create channel of type 'Zap' (cause 0) Zap show channel, lspci etc show everything is normal. Because X100P is Intel 537 modem, so can I use it as true modem for test? Did any one use X100P as a modem? Thank you very much. -------------- next part

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have been successful in getting Digest authentication to work with my Mitel 5055 IP Phones, however I'm wondering if Asterisk still supports Simple authentication? I know it has been depreciated in the RFC, but I have some phones with don't support Digest. Or how about an ACL? PB -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.5

Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? PB