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@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 _______________________________________________ App_rpt-users mailing list App_rpt-users@lists.allstarlink.org http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users
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