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Hi all, I’m
trying to use my 2-port multi-tech VoIP gateway to
talk to asterisk. Ideally I want to put it in a remote location with a POTS
line one port1 and an analog phone on port2 to call that location. Both the MultiTech and Asterisk have non-natted
static IP’s. I have tried every different type of configuration possible
for the sip.conf file. I can call from the analog
phone on the multitech to a local asterisk extension
and it rings, but when I
pickup I get a busy signal at both ends. When I try and call from asterisk to the phone on the multitech, I don’t even get that far. I receive this
from the CLI: --
Starting simple switch on 'Zap/10-1' --
Executing Dial("Zap/10-1", "SIP/multitech") in new stack --
Called multitech --
Got SIP response 486 "Busy Here" back from 122.33.44.55 --
SIP/multitech-964c is busy == Everyone is busy at this time n
Hungup 'Zap/10-1' The MultiTech seems pretty simple
to configure, just the IP of asterisk, username and pass. The only field I
haven’t tried its SIP URL. I was recently at a MultiTech
show and I saw them use x-lite to call to the MultiTech. Since neither is a sip proxy, I can’t
figure out why that worked for them but I can’t get this working with
asterisk. Here is the current version of my sip.conf [multitech] context=local ;disallow=all allow=all ;disallow=all allow=gsm allow=ulaw allow=alaw type=friend username=multitech secret=pass nat=no ;mailbox=200 host=dynamic reinvite=no ;canreinvite=yes qualify=1000 dtmfmode=info canreinvite=no callerid="Multi
Tech" ;defualtip=1.2.3.4 Thanks everyone, Steve |
