Quintum ax800 inbound calls asterisk


#1

Hi,

Trying to set up a Quintum AX800 to terminat PSTN calls to *.
Can’t get the * to register at the AX800.
When trying with following sip.conf:

[tenor]
type=peer
host=dynamic ;192.168.1.110
username=tenor
secret=tenor
disallow=all
allow=ulaw
allow=alaw
context=incoming

The * sees incoming SIP calls, but doesn’t answer them, here extensions.conf:

[general]
autofallthrough=yes
static=yes

[globals]
201=SIP/201
202=SIP/202
600=SIP/600

[default]
exten => 201,1,Dial(${201}) ;
exten => 202,1,Dial(${202}) ;
exten => tenor,1,Dial(${201});

[incoming]
exten => s,1,Playback(360111)
exten => tenor,1,Playback(360111);Dial(${201});
include default

The * just does not take the call which comes from the AX800, see SIP debug message:

— (13 headers 10 lines)—
Using INVITE request as basis request - call-0029678A-6972-DA11-0001-3B@192.168.1.110
Sending to 192.168.1.110 : 5060 (non-NAT)
Found peer 'tenor’
Dec 23 08:34:41 DEBUG[2744]: chan_sip.c:7096 check_user_full: Setting NAT on RTP to 0
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.110:10362
Dec 23 08:34:41 DEBUG[2744]: chan_sip.c:3501 process_sdp: Peer audio RTP is at port 192.168.1.110:10362
Found description format pcmu
Found description format pcma
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Dec 23 08:34:41 DEBUG[2744]: chan_sip.c:10322 handle_request_invite: Checking SIP call limits for device
Looking for 192.168.1.108 in incoming (domain )
Reliably Transmitting (no NAT) to 192.168.1.110:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.110;branch=z9hG4bK-tenor-c0a8-016e-0076;received=192.168.1.110
From: sip:192.168.1.110;tag=c0a8016e-58
To: sip:192.168.1.108;tag=as716d3e04
Call-ID: call-0029678A-6972-DA11-0001-3B@192.168.1.110
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ax-Forwards: 70
Contact: sip:192.168.1.108@192.168.1.108
Content-Length: 0


asterisk1*CLI>
<-- SIP read from 192.168.1.110:5060:
ACK sip:192.168.1.108@192.168.1.108 SIP/2.0
CSeq: 1 ACK
Call-ID: call-0029678A-6972-DA11-0001-3B@192.168.1.110
Contact: sip:192.168.1.110
From: sip:192.168.1.110;tag=c0a8016e-58
Session-GUID: 875782499-912536929-859398499-909730618
To: sip:192.168.1.108;tag=as716d3e04
Via: SIP/2.0/UDP 192.168.1.110;branch=z9hG4bK-tenor-c0a8-016e-0076
User-Agent: Quintum/1.0.0
Quintum: 0c01030b0237320501001301000f0b413032322d333030374338
Max-Forwards: 70

Anyone knows what’s wrong here? Just can’t get it to work.

Regards,
Jonathan


#2

Your AX800 is looking for extension 192.168.1.108 on your Asterisk box. You could figure out why your AX800 is doing this or you could add the following extension to your incoming context.

exten => 192.168.1.108,1,Goto(s,1)


#3

Hello,

I am trying to register a axg800 on asterisk and be able to make some calls on it ; this is the sip.conf I have, shouldn’t it work ? Any thing
special should be put on the tenor to call/register correctly ? We have
set the server IP, the username and password and the port (5060).
Thanks in advance.

disallow=all
allow=lpc10
allow=g729
allow=gsm
disallow=speex
allow=ilbc
allow=ulaw

[4754693]
type=peer
username=4754693
context=inbound
secret=
nat=yes
canreinvite=no
qualify=yes
host=dynamic
language=en
dtmfmode=inband