search for: olaifa

...t > > attached to the zhone. > > > > > > This makes it quite impossible to recieve call from outside or dial > > out > > form the asterisk. > > > > suggestions will be appreciated. > > > > > > > > regards > > -- > > Olaifa Augustine > > tel:- 234-2-8105156 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Olaifa Augustine General Data Engineering Services Ltd 18b o...

...signalling = fxs_ks context =incoming channel=1 signalling =fxo_ks context= internal channel2 extensions.conf [incoming] exten=>s,1,Dial,Zap/1|20 [outgoing] exten => 125,1,Dial,Zap/2|20 [internal] include=incoming include=outgoing i am running this on RH8.0. Someone Please assist. -- Olaifa Augustine General Data Engineering Services Ltd

...onf': Found == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.5.1) [skipping chan_zap.so] Asterisk Ready. *CLI> As you will see in the last line "skipping chan_zap.so" before now it gave me the following error message -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156

...;Zap/7-1","1") in new stack. -- executing..... it gives the same warning message even when the amphaenol calbe is not attached to the zhone. This makes it quite impossible to recieve call from outside or dial out form the asterisk. suggestions will be appreciated. regards -- Olaifa Augustine tel:- 234-2-8105156

Hi, Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

...e most especially the dialtone parameters using export VPB_TONE= DIAL,C,437,100,2000 but there was no change. Is there anyone out there that has got the voicetronix Openswitch working? i would like to share the working configs, and learn how to get rid of the Tone Detect : Grunt regards -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156

I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I have read that the Zaptel devices share an IRQ, is this causing the problem? Is there a way that I can manually change the IRQs of

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

...there it says that asterisk is running but never gets to the normal asterisk prompt (*CLI>). when i do "asterisk -vr" from another machine or terminal it just stays there and does not get to the prompt and all the phones still do not have tone. Any suggestions please! Regards -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156