asterisk users - Mar 2004 - Using MultiTech MVP-210 as FXO/FXS gateway

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
*       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
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I tested Multitech with the same scenario and it works.

Stephen Foster wrote:
> 
> 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.
> 
No so simple. At least you must to elaborate the following windows:
IP, Voice/Fax, Interface, Phone Book configuration,Outbound Phone Book, 
Inbound Phone Book.

> 
> Here is the current version of my sip.conf
> 
>  
> 
> [multitech]
> 
> context=local
> 
My sip.conf:

[multitech]
context=default
type=friend
host=192.168.YY.XX     ; multitech IP
dtmfmode=inband        ; we use alaw


Hope this help.

Jorge

I am using an MVP-210 as FXS -- I haven't tried FXO.

Here's my sip.conf entry:

[mvp-x303]
type=friend
host=192.168.1.93
username=303
dtmfmode=rfc2833
context=fs1
disallow=all
allow=ulaw

(Not sure if dtmfmode is correct.)

Username must be an extension number that appears in the MVP210's inbound 
phone book.

Here's an MVP210 outbound phone book entry to call x352 on the Asterisk
server:

Destination Pattern: 352
Total Digits: 3
IP Address: 192.168.1.94        (the Asterisk server)
Protocol Type: SIP
Transport Protocol: UDP (MVP210 defaults to TCP)
SIP Port Number: 5060


At 01:23 PM 3/11/2004 -0500, Stephen Foster wrote:
>Hi all,
>
>             Im 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 IPs.
>
>
>
>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 dont 
>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 havent 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 cant figure out why that worked 
>for them but I cant 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