asterisk users - Jun 2007 - X-Lite problems on basic asterisk setup

I'm trying to setup my first Asterisk setup on a CentOS 5 installation
on VMWare Workstation 6.  Got two Linksys SPA941s working fine.  But 
X-Lite softphones can't answer phone calls, and when one of them calls 
on of the Linksys phones they "connect" but neither party can hear
hear
the other.  I noticed that the Linksys phones are connected via Native 
bridging while the X-Lite ones are connected via Packet2Packet bridging.

Also, on the X-Lite phones there is a about a 30 second lag between when 
the X-Lite client hits dial/call and when the called party starts ringing.


::Asterisk setup::
Asterisk 1.4.4
Zaptel 1.4.3 (only ztdummy compiled)
Asterisk Addons 1.4.1
CentOS 5
VMWare Workstation 6


::sip.conf::
[Linksys01]
type=friend
secret=ledzep
context=default
host=dynamic
mailbox=6445

[X-Lite01]
type=friend
secret=rammerjammer
context=default
host=dynamic
dtmfmode=rfc2833
mailbox=2070
canreinvite=yes
nat=no

[Linksys02]
type=friend
secret=bigben
context=default
host=dynamic
mailbox=6368
qualify=yes


::extenstions.conf::
[default]
include => demo

exten => 6445,1,Dial(SIP/Linksys01,20)
exten => 6445,n,Voicemail(u6445)

exten => 2070,1,Dial(SIP/X-Lite01,20)
exten => 2070,n,Voicemail(u2070)
exten => 2070,n,HangUp()

exten => 6368,1,Answer
exten => 6368,n,Ringing
exten => 6368,n,Dial(SIP/Linksys02,20)
exten => 6368,n,Voicemail(u6368)
exten => 6368,n,HangUp()




-------------------
Andrew Stewart

This typically happens when the phone is natting or there is a firewall
between the phone and the asterisk server. The connection is made via
sip (5060), but the voice is over ports 10000-20000 (RTP). Most likely,
the sip connection is succeeding, since you are connecting, but the
actual voice is failing to transfer over RTP.

if this is the case, I would aim to use IAX since it was made for this
type of use.

If the phone is on the same network as the asterisk server, and you are
still having issues, use a packet sniffer and watch the traffic on both
ends. You should be able to receive every packet that is sent. Most
likely in this case though, you will only see those 5060 packets making it.

Rob


Andrew Stewart wrote:
> I'm trying to setup my first Asterisk setup on a CentOS 5 installation
> on VMWare Workstation 6.  Got two Linksys SPA941s working fine.  But 
> X-Lite softphones can't answer phone calls, and when one of them calls 
> on of the Linksys phones they "connect" but neither party can
hear hear
> the other.  I noticed that the Linksys phones are connected via Native 
> bridging while the X-Lite ones are connected via Packet2Packet bridging.
>
> Also, on the X-Lite phones there is a about a 30 second lag between when 
> the X-Lite client hits dial/call and when the called party starts ringing.
>
>
> ::Asterisk setup::
> Asterisk 1.4.4
> Zaptel 1.4.3 (only ztdummy compiled)
> Asterisk Addons 1.4.1
> CentOS 5
> VMWare Workstation 6
>
>
> ::sip.conf::
> [Linksys01]
> type=friend
> secret=ledzep
> context=default
> host=dynamic
> mailbox=6445
>
> [X-Lite01]
> type=friend
> secret=rammerjammer
> context=default
> host=dynamic
> dtmfmode=rfc2833
> mailbox=2070
> canreinvite=yes
> nat=no
>
> [Linksys02]
> type=friend
> secret=bigben
> context=default
> host=dynamic
> mailbox=6368
> qualify=yes
>
>
> ::extenstions.conf::
> [default]
> include => demo
>
> exten => 6445,1,Dial(SIP/Linksys01,20)
> exten => 6445,n,Voicemail(u6445)
>
> exten => 2070,1,Dial(SIP/X-Lite01,20)
> exten => 2070,n,Voicemail(u2070)
> exten => 2070,n,HangUp()
>
> exten => 6368,1,Answer
> exten => 6368,n,Ringing
> exten => 6368,n,Dial(SIP/Linksys02,20)
> exten => 6368,n,Voicemail(u6368)
> exten => 6368,n,HangUp()
>
>
>
>
> -------------------
> Andrew Stewart
>
>
>
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