vicidial.org

We have installed Asterisk, Apache, and ViciDial on Slackware 10.2 but for some reason our softphone from another machine with a defferent public static ip is unable to connect with our Asterisk server. The softphone gives an error message saying that the server rejected the connection. We are able to use the softphone when we directly put our telecom provider's IP in it but not when we put the Asterisk server's ip. We are using Express Talk SIP phone.

How do we double check if we configured Asterisk or ViciDial correctly to interface with the softphone? Thanks

You are going to have to explain your network setup in more detail.

Do you have a firewall or NAT in any way?

are you using IP addresses or domain names to connect?

what is the traceroute from your clcient PC to the Asterisk server?

We do Not have a Firewall on either one of the machines.

Our NAT is Disabled.

We are using static public IP addresses to connect.

I can check and provide you with the trace routes. Both machines are in the same office and use the same ISP but are NOT on one Network. They both use separate static public ip's.

Make sure your subnet settings are correct!

We used the default subnet settings from scratch install because I think that's how we had our own machine configured. Please advice.

whats your asterisk CLI when client attempting to connect to your asterisk server...

can you please post also your sip.conf

Below are Asterisk CLI and our sip.conf settings:

*CLI> == Refreshing DNS lookups.
Sep 14 14:00:50 NOTICE[4172]: chan_iax2.c:7875 iax2_poke_noanswer:
Peer 'expres s talk' is now UNREACHABLE!
Time: 0
Sep 14 14:01:49 NOTICE[4168]: chan_sip.c:11045
handle_request_register: Registration from
'<sip:talk@ourasteriskserverip>' failed for 'oursoftphoneip' -
Username/auth name mismatch
-- Registered SIP 'spa2001' at oursoftphoneip port 5061 expires 3600
-- Saved useragent "Express Talk 2.02" for peer spa2001
-- Got SIP response 481 "Call/Transaction Does Not Exist" back
from oursoftphoneip
Sep 14 14:04:55 NOTICE[4168]: chan_sip.c:11564 sip_poke_noanswer: Peer
'spa2001' is now UNREACHABLE! Last qualify: 1


Below is our sip.conf setting:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-pstn
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=rfc2833
; qualify=yes

[gs102]
disallow=all
allow=ulaw
allow=alaw
type=peer
# username=gs102
# secret=test
host=ourtelecomproviderip
dtmfmode=inband
defaultip=ourtelecomproviderip
qualify=1000
mailbox=102
[spa2000]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=spa2000
secret=test
host=dynamic
dtmfmode=inband
defaultip=oursoftphoneip
qualify=1000
mailbox=2000
[spa2001]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=spa2001
secret=test
host=dynamic
dtmfmode=inband
defaultip=oursoftphoneip
qualify=1000
mailbox=2001

Please advice

can you explain the another machine that you are reffering to? is it on the LAN?

[spa2001]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=spa2001
secret=test
host=dynamic
dtmfmode=inband
qualify=1000
mailbox=2001
canreinvite=no

try deleting the default ip and adding canreinvite=no and see if it works..

The other machine is Not on LAN. Actually, neither the machine with the sofphone nor our Asterisk server are on LAN. They are both connected independently to the same broadband provider but are assigned separate static Public IP's. I will try your suggestion right now.

By the way, where in the sip.conf setting do we put our Telecom/Termination provider's IP? Or do we put that some place else?

Also, our telecom provider had mentioned the following:

context=from-pstn
dtmfmode=rfc2833

Scratch Install has:
dtmfmode=Inband
and
Nothing for context

Should we include context=from-pstn and include dtmfmode=rfc2833?

Thanks Again.

you will put your provider's voip in sip.conf if sip, if iax in iax.conf

dtmfmode depends on what type of voip phone you are using. try using rfc if it works.

context is where to start in the dialplan when this phone calls.

Thanks for the feedback.

Which field in sip.conf corresponds to the telecom provider's ip? There are 2 or 3 places where it asks for default or host ip.

Which field relates to the softphone ip in sip.conf?