Hello guys, I have a some problem with ringing multiple phones simultanously.
I would like to set one extension which should call multiple phones (in other office) simultaneously, something like Dial(SIP/exten1&SIP/exten2). In my case I have two astersik sever and I use IAX between asterisks. Let’s assume that serverA has extensions range 3XXX and severB extensions range 4XXX. The phones which I want to call are connected to serverB,f.e.: 4004 and 4003. I need to create extension 4445(4447) on serverA which call 4003 and 4004. I use dundi for advertising extensions (I have other asterisk servers). I just want to add that all normal calls work in both directions from (serverA-serverB and serverB-serverA). I tried set this 4445/(4447) extension using Local channel variables in two ways: with Dial() application and with Queue(). Here are a parts of configuraton files from serverA:
===extensions.conf====
[Queues]
exten => 4447,1,Answer()
exten => 4447,n,Queue(itemerg)
exten => 4447,n,Hangup()
[it-emergency]
exten => 4445,1,Dial(Local/100@it-emerg&Local/101@it-emerg)
exten => 4445,2,Hangup()
[it-emerg]
exten => 100,1,Goto(lookup-dundi,4003,1)
exten => 101,1,Goto(lookup-dundi,4004,1)
[users]
include => lookup-dundi
include => it-emergency
include => Queues
include => outgoing-local
====queues.conf=========
[code]
[general]
autofill=yes
shared_lastcall=yes
[itemerg]
musicclass=default
;strategy=rrmemory
strategy=ringall
joinempty=yes
leavewhenempty=yes
ringinuse=no
member => Local/100@it-emerg
member => Local/101@it-emerg[/code]
It works until the DND is enabled on f.e. 4003. When DND is enabled and I dial 4445/4447 from phone with 3099 number, then in aster CLI> I got something like this:
[Nov 30 17:02:13] -- Executing [4445@users:1] Dial("SIP/3099-0000010a", "Local/100@it-emerg&Local/101@it-emerg,30") in new stack
[Nov 30 17:02:13] -- Called 100@it-emerg
[Nov 30 17:02:13] -- Executing [100@it-emerg:1] Goto("Local/100@it-emerg-4861;2", "users,4003,1") in new stack
[Nov 30 17:02:13] -- Goto (users,4003,1)
[Nov 30 17:02:13] -- Called 101@it-emerg
[Nov 30 17:02:13] -- Executing [101@it-emerg:1] Goto("Local/101@it-emerg-e651;2", "users,4004,1") in new stack
[Nov 30 17:02:13] -- Goto (users,4004,1)
[Nov 30 17:02:13] -- Called iaxuser:+pDtIhClxxA+29MQxAR9eg==@IP_serverB/4003
[Nov 30 17:02:13] -- Call accepted by IPADDR (format alaw)
[Nov 30 17:02:13] -- Format for call is alaw
[Nov 30 17:02:13] -- IAX2/IP_serverB:4569-561 is ringing
[Nov 30 17:02:13] -- Local/100@it-emerg-4861;1 is ringing
[Nov 30 17:02:13] -- Called iaxuser:+pDtIhClxxA+29MQxAR9eg==@IP_serverB/4004
[Nov 30 17:02:13] -- Call accepted by IP_serverB (format alaw)
[Nov 30 17:02:13] -- Format for call is alaw
[Nov 30 17:02:13] -- IAX2/IP_serverB:4569-2709 answered Local/101@it-emerg-e651;2
[Nov 30 17:02:13] -- Local/101@it-emerg-e651;1 answered SIP/3099-0000010a
[Nov 30 17:02:13] -- Hungup 'IAX2/IP_serverB:4569-561'
[Nov 30 17:02:13] == Spawn extension (users, 4003, 1) exited non-zero on 'Local/100@it-emerg-4861;2'
[Nov 30 17:02:14] -- Hungup 'IAX2/IP_serverB:4569-2709'
[Nov 30 17:02:14] == Spawn extension (users, 4004, 1) exited non-zero on 'Local/101@it-emerg-e651;2'
[Nov 30 17:02:14] == Spawn extension (users, 4445, 1) exited non-zero on 'SIP/3099-0000010a'
[Nov 30 17:02:30] == Using SIP RTP TOS bits 184
[Nov 30 17:02:30] == Using SIP RTP CoS mark 5
[Nov 30 17:02:30] == Using SIP VRTP TOS bits 136
[Nov 30 17:02:30] == Using SIP VRTP CoS mark 6
[Nov 30 17:02:30] -- Executing [4447@users:1] Answer("SIP/3099-0000010b", "") in new stack
[Nov 30 17:02:30] -- Executing [4447@users:2] Queue("SIP/3099-0000010b", "itemerg") in new stack
[Nov 30 17:02:30] -- Started music on hold, class 'default', on SIP/3099-0000010b
[Nov 30 17:02:30] -- Executing [100@it-emerg:1] Goto("Local/100@it-emerg-f054;2", "users,4003,1") in new stack
[Nov 30 17:02:30] -- Goto (users,4003,1)
[Nov 30 17:02:30] -- Executing [101@it-emerg:1] Goto("Local/101@it-emerg-5af9;2", "users,4004,1") in new stack
[Nov 30 17:02:30] -- Goto (users,4004,1)
[Nov 30 17:02:30] -- Called iaxuser:+pDtIhClxxA+29MQxAR9eg==@IP_serverB/4003
[Nov 30 17:02:30] -- Called iaxuser:+pDtIhClxxA+29MQxAR9eg==@IP_serverB/4004
[Nov 30 17:02:30] -- Call accepted by IP_serverB (format alaw)
[Nov 30 17:02:30] -- Format for call is alaw
[Nov 30 17:02:30] -- Call accepted by IP_serverB (format alaw)
[Nov 30 17:02:30] -- Format for call is alaw
[Nov 30 17:02:30] -- IAX2/IP_serverB:4569-3357 answered Local/101@it-emerg-5af9;2
[Nov 30 17:02:30] -- Local/101@it-emerg-5af9;1 answered SIP/3099-0000010b
[Nov 30 17:02:30] -- Hungup 'IAX2/IP_serverB:4569-641'
[Nov 30 17:02:30] -- Stopped music on hold on SIP/3099-0000010b
and on the phone display there is “Hung up” or “Call Ended” message.
Could someone please help me?
Does anyoone has an idea what could it be related to?