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Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

...eases delay, may disrupt some functions, is a hassle to administer, etc. But it's better than nothing. I am considering adding MG capability to the * MGCP stack. Do you or does anyone have an interest in helping with this? --Stewart > Date: Fri, 18 Jun 2004 10:30:15 -0400 > From: "Kubat, Philip" <pkubat@kubat.com> > Subject: [Asterisk-Users] ATT CallVantage & Asterisk > I am trying to connect directly to AT&T VoIP service CallVanage. I have > ATT's ATA (D-Link DVG-1120M). They use mgcp. I have traces of the connects > from the Dlink and hopin...

It looks like Asterisk's mgcp, defaults as "connect to" endpoints or gateways. Is there a means to have it act as the endpoint or gateway? I have another system that needs to "connect to" a mgcp endpoint and I would like that to be asterisk. Thanks! Philip Kubat

This may be a newbie SIP/NAT question. If so I am sorry. But any help would be appreciated. My Asterisk server is behind an ipchains box and I am trying to connect to Broadvoice. All works fine without the NAT. I have a global nat=yes prior to my register, but the sip debug allows shows "no nat)". Is this "label" issue, and am I barking up the wrong tree? Sip.conf....

I have a SIP connection to Broadvoice and sometimes when I make outgoing calls from a SIP ATA-188 (could be the same number) (the ATA-188, is currently the only extension), there is no audio passed for 5-10 secs. I have set all the codec the same to 711u and also ensured canreinvite is set to no. Any suggestions? Places to look for? -------------- next part -------------- An HTML

Hi, i have a problem trying to register the codec G729, my licence is valid but when i try to Register i got the following error: "Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient

What is the relationship between the peer definitions and the register command? In reviewing the sample sip.conf it seems that you can place the "sip_proxY" peer as the hostname. Is this correct? This question adds the the Broadvoice thread and where to place the dtmfmode variable. sip.conf --- (asterisk sample) -------------------------------- ;register =>

I am trying to connect directly to AT&T VoIP service CallVanage. I have ATT's ATA (D-Link DVG-1120M). They use mgcp. I have traces of the connects from the Dlink and hoping to setup Asterisk the same. It looks like I need to have Asterisk be a MGCP endpoint (gateway). How do I configure this? Does the mgcp.conf support "register" like sip etc? What is the syntax?

I have a Cisco ATA-188 with two POTS phones and latest stable cvs. In any situations with call waiting (existing connection and calling again) the second call cause both calls to drop. This is the same for "internal" extensions and from external (ZAP and SIP). It seems to be a "502 - The transaction could not be executed, because the endpoint does not have sufficient