[App_rpt-users] Connect multiple SIP or IAX clients to a node

[2E0SIP]

Hi Bryan,

Thanks for the response. Here's the relevant section of sip.conf:

[sip-client]
type = peer
context = sip-client
host = 127.0.0.1
insecure=very
disallow = all
allow = ulaw
allow = alaw
allow = gsm
transfer = no

[audio-bridge]
type = peer
context = sip-client
host = 217.163.11.214
insecure=very
disallow = all
allow = ulaw
allow = alaw
allow = gsm
transfer = no

There is no NAT in between the client and AllStar Asterisk Node

Thanks

Matthew
2E0SIP

On Thu, Apr 12, 2018 at 9:36 PM, Bryan Fields <Bryan at bryanfields.net> wrote:

> 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
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