search for: smadi

...PSTN channel 2 C calls confroom over SIP/Ethernet then i have all of them talking and the media stream mixed by asterisk. However, if i hang up A, channel 1 is still ocuppied (i try dialing inbound again on channel and it continues to give a busy siganl) any idea how that can be resolved? m. smadi

any idea on how we can setup threeway calling in * thanks moe smadi

hi; does any body know what are the physical dimension of a digium care 400pm for example? thanks m.smadi

...; Does asterisk handle the "refer method"? If we use asterisk as both a sip proxy and a GW what would happen to POTS leg (SIP --> * --> POTS) after that send client send a "refer" request which refers the POTS destination to let's say another SIP phone? thanks m. smadi

...find out the delay between two events: 1) the instance a call is recieved on an FXO port and the 2) the instance a SIP INVITE is sent to the SIP destination. i need to attach timestamps to the events before logging them. How can i: 1) log `ALL' events. 2) Attach timestamps to them? thanks m.smadi

If i leave my asterisk server running for a long time then try to dial outbound on the zaptel channel i get this high pitch static noise and won't dial out. This behavior is happening over two different servers i am using. Rebooting asterisk does not sovle the problem. I rmmod the zaptel driver then reload and that solved the problem. But i cannot continue to do that. Also sip to sip

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change

Hello, I'm trying to setup an Asterix PBX solution in our office. We plan to have 5 active lines open available at any point in time. We'd like to use VoIP Phones, and possibly Software Based phone (*NIX/Windows enviroment). I was researching the various cards and I think I'd want to go with the Digium TDM40B - 4-port. However, I can't figure the differences between FXS &

Hello, I have an Asterisk with one local Cisco ATA and one remote Cisco ATA connected to the Asterisk, the remore connection is a satellite link with an 900ms delay. I can make calls from the remote site to the local site, but when try to call from local to remote it doesn't work. The Asterisk timesout, it sais no one answered and can?t establish the connection. Can anybody help me with

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing

How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",

i have tried to connect my asterisk server to vonage like this: Sip.conf: register => 1<yournumber>:<secret>@atlas-east.vonage.net:5060 [vonage] type=friend username=1<yournumber> secret=<secret> host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1<yournumber> fromdomain=atlas-east.vonage.net canreinvite=no nat=yes