asterisk users - Feb 2020 - Predictive call - agent talking to a customer, then suddenly talking to another customer

Hi, do you have NAT between Asterisk and agent phones? 

S pozdravem
Tomáš Holý

Hi Tomas

Thanks for replying.

Yes, the phones are in one location in a LAN and are then NATed to enable them
to contact the Asterisk which is hosted in the cloud.

A typical sip.conf phone configuration on the remote server for the site is

[general]
session-timers=refuse
disallow=all
allow=g729:20
allow=ulaw
allow=alaw
fromuser=xxx
useragent=xxx
callcounter=yes
alwaysauthreject=yes
allowguest=no
jbnable=yes
jbforce=no
jbimpl=adaptive
jblog=no
jbmaxsize=200
jbresyncthreshold=1000
externaddr=xx.xx.xx.xx
localnet=xx.xx.xx.xx/255.255.255.0

[xxxx]
type=peer
user=xxxx
secret=xxxx
host=dynamic
disallow=all
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=xxxx
call-limit=1
limitonpeers=yes
callgroup=1
pickupgroup=1
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx/255.255.255.255
nat=force_rport,comedia

Not sure if that helps much.

Thanks for the reply!
> 13. 2. 2020 v 19:06, Stefan Viljoen <viljoens at verishare.co.za>:
> 
> 
> Hi all
>  
> Asterisk 13 instance - I’ve got a situation in an agent queue that an agent
will be talking to one person, then suddenly the same agent will be talking to
another person who was talking to another agent.
>  
> The calls do not switch around between the two agents, the “losing” agent
will just suddenly have silence in his handset and the other agent will now be
talking to “his” customer.
>  
> The original customer is simply cut off instantly the moment this happens.
>  
> This happens randomly. I have yet to collect log output and capture the
CLI, etc. but anybody ever heard of this happening?
>  
> It is as if agent channels get randomly reassigned / lose their audio
channel from outside the Asterisk - one channel is disconneted that a customer
was on and certain agents are suddenly talking to someone else and lose the
original caller they were busy with.
>  
> Where can I even being to look?
>  
> Thx!
> <image001.png>

Hi Stefan,
IMHO NAT would be that problem. Check if you have enabled SIP helpers on router,
try it disable, try set static port redirection to phones and set RTP ports in
phone configuration (every phone must have different ports range!). Even better,
make VPN between Asterisk and phones and leave it in one network without NAT.


S pozdravem
Tom Hol

> 14. 2. 2020 v 9:45, Stefan Viljoen <viljoens at verishare.co.za>:
> 
> Hi, do you have NAT between Asterisk and agent phones? 
> 
> S pozdravem
> Tom Hol
> 
> Hi Tomas
> 
> Thanks for replying.
> 
> Yes, the phones are in one location in a LAN and are then NATed to enable
them to contact the Asterisk which is hosted in the cloud.
> 
> A typical sip.conf phone configuration on the remote server for the site is
> 
> [general]
> session-timers=refuse
> disallow=all
> allow=g729:20
> allow=ulaw
> allow=alaw
> fromuser=xxx
> useragent=xxx
> callcounter=yes
> alwaysauthreject=yes
> allowguest=no
> jbnable=yes
> jbforce=no
> jbimpl=adaptive
> jblog=no
> jbmaxsize=200
> jbresyncthreshold=1000
> externaddr=xx.xx.xx.xx
> localnet=xx.xx.xx.xx/255.255.255.0
> 
> [xxxx]
> type=peer
> user=xxxx
> secret=xxxx
> host=dynamic
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> dtmfmode=rfc2833
> context=xxxx
> call-limit=1
> limitonpeers=yes
> callgroup=1
> pickupgroup=1
> deny=0.0.0.0/0.0.0.0
> permit=xxx.xxx.xxx.xxx/255.255.255.255
> nat=force_rport,comedia
> 
> Not sure if that helps much.
> 
> Thanks for the reply!
> 
>> 13. 2. 2020 v 19:06, Stefan Viljoen <viljoens at
verishare.co.za>:
>> 
>> 
>> Hi all
>> 
>> Asterisk 13 instance - Ive got a situation in an agent queue that an
agent will be talking to one person, then suddenly the same agent will be
talking to another person who was talking to another agent.
>> 
>> The calls do not switch around between the two agents, the losing agent
will just suddenly have silence in his handset and the other agent will now be
talking to his customer.
>> 
>> The original customer is simply cut off instantly the moment this
happens.
>> 
>> This happens randomly. I have yet to collect log output and capture the
CLI, etc. but anybody ever heard of this happening?
>> 
>> It is as if agent channels get randomly reassigned / lose their audio
channel from outside the Asterisk - one channel is disconneted that a customer
was on and certain agents are suddenly talking to someone else and lose the
original caller they were busy with.
>> 
>> Where can I even being to look?
>> 
>> Thx!
>> <image001.png>
> 
> 
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