[asterisk-users] X-Lite problems on basic asterisk setup

[Andrew Stewart]

Packet sniffer found the problem.  RTP was firewalled on the Asterisk 
box.  Fixed it using the Asterisk firewall rules page on the wiki 
<http://www.voip-info.org/wiki-Asterisk+firewall+rules>.

The 30 second lag on the dialing has something to do with using the 
domain name instead of the IP address of the asterisk server in the SIP 
config on X-Lite.  The call goes immediately when I set the domain to 
the IP address of the asterisk box.

Thanks for your help.

Rob Schall wrote:
> 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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-- 
-------------------
Andrew Stewart
astewart at notre1.com
(205) 585-2980 - cell