Re: [Sipwitch-devel] sipwitch call dropped after 16 seconds

[Haakon Meland Eriksen]

Hi Paul,

David suggests using Wireshark to find out what really happens in your network, 
he writes:

"SIP depends on a 3 step handshake to establish a call.  There is an invite, 
an answer, and a ack.  Many sip clients establish audio for the call on the 
answer.  However, if the do not receive the ack within 16 seconds, they then 
disconnect.

SIP witch sends the ack back to the client when it receives it from the 
destination.  This probably will require wireshark to figure out if the 
destination sent back the ack, and then if/where sipwitch sent it to the 
caller.  Most often when this happens it is a network routing or client config 
issue, though."

Yours,
Haakon


Onsdag 10. august 2011 09.48.51 skrev Paul Mathebula :
> Hi,
> 
> I have downloaded, compiled ad installed sip witch release candidate 1.1.1
> i had previously installed other version. My calls are setup up properly
> HOWEVER all my active calls are terminated after 16 or so seconds. All
> previous versions I had installed from 0.8.4 exhibit the same limitations.
> What is the cause of this and how do i resolve it ? Thanks in advance.
> Additional info;
> 
> 
>    - ubuntu natty
>    - running sipwitch as root
>    - my domain in sipwitch.conf is fully qualified and ends with a fullstop
>    - no localnames specified
>    - realm is gnuvoip which is also used in client voip softphones
> 
> my sipwitch.conf:
> 
> <?xml version="1.0"?>
> <sipwitch>
> <!-- my test users -->
> <provision>
> <user id="bumblebee">
>      <secret>sam001</secret> <extension>101</extension>
> <display>Bumblebee</display>
>    </user>
> <user id="megatron">
>      <secret>allspark</secret> <extension>102</extension>
> <display>Megatron</display>
>    </user>
> </provision>
> <access>
> </access>
> 
> <stack>
> 
>   <domain>fqdn.</domain><!--my fully qualified domain name, withheld-->
> 
>   <mapped>900</mapped>
>   <threading>4</threading>
>   <interface>*</interface>
>   <dumping>false</dumping>
> 
>   <system>system</system>
>   <anon>anonymous</anon>
> </stack>
> 
> <timers>
>   <!-- ring every 4 seconds -->
>   <ring>4</ring>
>   <!-- call forward no answer after x rings -->
>   <cfna>4</cfna>
>   <!-- call reset to clear cid in stack, 6 seconds -->
>   <reset>6</reset>
> </timers>
> 
> <registry>
> 
>   <prefix>100</prefix>
>   <range>1000</range>
>   <keysize>77</keysize>
>   <mapped>1800</mapped>
>   <realm>gnuvoip</realm>
> </registry>
> 
> <routing>
> </routing>
> </sipwitch>
> 
> 
> my /etc/default/sipwitch contents:
> 
> # Default values for daemon operation.  This should be edited and is
> invoked # by init script.
> 
> # install specifc plugins, or use "auto" to auto-load whatever is
> installed...
> PLUGINS="zeroconf scripting subscriber forward"
> 
> # runtime priority, recommended realtime for high capacity
> PRIORITY="1"
> 
> # can be used to adjust pthread concurrency...
> CONCURRENCY=0
> 
> # can be used to specify running effective user/group id for the server
> GROUP="sipwitch"
> 
> # set server errlog history buffer, typical may be 100, default is none...
> HISTORY=10000
> 
> # set UID mapping for automatic extension numbers, or 0 to disable
> FIRSTUID="1000"
> 
> # set group for automatic sip users, or - to disable
> SIPUSERS="sipusers"
> 
> # specify security model, desktop or server.
> SECURITY="server"