Recent content by wagtech

Is there ports for SLT's installed, if so then Program 110 Station Type should allow you to pick from the following... In case of selecting SLT type, there are 6 types as below:  SLT (DTMF)  SLT (Pulse)  SLT (DTMF VOL-MW)  SLT (Pulse-MW)  SLT (DTMF FSK-MW)  SLT (DTMF POL-MW)

Very helpful, by your use of 'probably' probably means you don't know. eMG80/800 DO pulse dialing as long as the appropriate boards are included. Suggest if YOU don't know then don't comment.

Thanks for the info, greatly appreciated.

Would the leading digit table cause issue with the outgoing CO digit 0 if the system also looks for 000? Daniel_RO if you try this would be good to know how you go.

International Subscriber Dialing, Australia has 0011 or 0014 for international outgoing calls followed by country code + area code + phone number (not that anyone would pay to use it if they use chat programs now days!).

I see a problem with that in dialing 0 then 00 with an 0 added by the system you then bar access to ISD (001x). As soon as you dial 0 then 00 the pbx will add 0 unless you specifically assign another digit/s for ISD or not allow ISD.

If you try this link and then I can email you the config details direct- http://www.nontaurus.com/msgs/ToAdmin.asp If you are party to this... Phil

1) Only one (trial at present, just ordered two licenses) 2) Extn setup is basic as below- Ext cfg exten => 201,1,Macro(exten-vm,novm,201) exten => 201,hint,SIP/201 Sip cfg [201] username=201 type=friend record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=yes mailbox=201@device...

It looks like the ipo is failing on i/c calls starting with 8xxx. But aren't you stripping 8 in the *? Maybe try turning on SIP Tx and SIP Rx on Monitor to see what the two units are sending each other in the SIP packets. Post here if you want.

Oh and I forgot to mention having an incoming call route on the IPO for each ext to map the * ext (or group) number to the IPO ext/group.

Ah, so you DO have audio both ways on a call, my problem was one way audio on a call. On my setup I dont use registration on the * only set the ip address for the extension (the IPO's ip), in fact I have all the IPO's exts existing on the * with the same IP so you can dial any ext on the IPO...

Ditto for the setup. My setup IPO with no NAT, * with NAT. IPO only calls the * or through it to outside world. To receive calls I have calls route via the * ext. to the IPO Are you able to make a call out with audio both ways from the * direct? Check on the * monitor how the call is...

eholm, would have to do with turning on NAT on the asterisk side? What I just found was the asterisk side was sending its public IP to the SOE causing the SOE to reply back to the public address. Now that I have turned on NAT on the asterisk side it now sends only its LAN ip addy. It now works...

After a bit more investigation voice is going out of the SOE but not in. So may not be a codec issue!?

Having a problem with audio between SOE Sip trunk and Asterisk PBX where audio seems to be one way then after a while the call disconnects (most likely to quiet timeout). Sounds to me to be a Codec issue, does anyone know which codec is best on either side? Unless someone has other ideas... Phil