On 4/12/18 11:03 AM, 2E0SIP wrote:
> Hi All
>
> I currently have an issue where by if a SIP client has 'dialled' into a node
> and another SIP client then dials in, the first call is terminated with a BYE.

I don't run a SIP client into my nodes, but this should work.

Question, is there NAT involved?

What's your sip.conf look like?

> The logs, even with full debugging enabled, don't really reveal why this is
> occurring.
>
> Someone else in a Facebook group reported the same issue with an IAX client.

IAX is totally different.  Post the details as a new thread on this list.

>
> Here's the context the SIP clients get dropped into:
>
> [sip-client]
> exten => ${NODE},1,Answer
> exten => ${NODE},n,Set(NODENUM=${CALLERID(number)})
> exten => ${NODE},n,Set(CALLERID(num)=0)
> exten => ${NODE},n,Rpt,${NODE}|Pv|${CALLERID(name)}
> exten => ${NODE},n,Hangup
> exten => ${NODE},n(hangit),Answer
> exten => ${NODE},n(hangit),Wait(1)
> exten => ${NODE},n(hangit),Hangup

Looks ok, and it should support concurrency.


--
Bryan Fields

727-409-1194 - Voice
http://bryanfields.net
_______________________________________________
App_rpt-users mailing list
App_rpt-users@lists.allstarlink.org
http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users

To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the "Unsubscribe or edit options button"
You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.