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as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

i've just spent the pass 2 days trying to get AGI to work with PHP; i made a lot of silly mistakes along the way which could have been avoided if only there were some kinda howto or samples. at the risk of looking stupid, i decided to shared my experience in hopes that it might help some newbie get going with PHP. 1. first order of business is to be aware of your php environment; i m NOT

...ow is it not economical? > I already have the PBXs on both sides. > If I switch to * I'll need to get a channel bank > > Am I wrong? > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of hkirrc.patrick > Sent: Wednesday, November 05, 2003 8:36 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Using Asterisk as a VOIP gateway > > yes you can but may not be all that economical though. > on the other hand, if you can replace or do away with > at le...

i've already tried to change the indications.conf to the following: dial = 0/1500 but the dial tone still persists i am using the following workaround but obviously not a clean b'cos it just replace dial tone with some other tone. in zapata.conf context=spec immediate=yes signalling=fxo_ks channel=>2 ; TDM400p-1 in extensions.conf [spec] exten =>

the above-message keep popping up every second during a conversation between a zap(fxs) channel and sip channel. * eventually hung after a long while we can talk to each other and we can ring one another without any problem. i've had x-lite and x-pro register with * without this problem. furthermore, i have ask my friend to turn off all codec expect g.711MLAW; that did not help i then

i've tried to set dial = 0/1500 in the indications.conf but still getting a dial tone. is this a bug with *? or did i do something wrong. thank you, patrick

would some kind guru please point to me where i can pick up all the various required and optional labels required by each manager's action or command. i've looked high and low, and found very little info. i can login an originate a call thanks to: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API