Unable to make outbound calls
Posted: Mon Oct 21, 2019 6:42 am
Hi guys, am able to make inbound calls but outbound calls , please help
here is my trunk configuration:
Registration string: register => username:password@sip ip:5060
account entry:
[voip]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
type=friend
username=username
secret=password
host=XXXXXXXX
dtmfmode=rfc2833
context=trunkinbound
Dial plan entry:
exten => _0237.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0237.,3,Dial(${VOIPTRUNK}/${EXTEN:2},,tTo)
exten => _0237.,4,Hangup
sip show registratry =>ok
sip show peers =>ok
here is the CLI output:
= Using SIP RTP CoS mark 5
> 0x7f6aac50a3c0 -- Strict RTP learning after remote address set to: 154.72.162.24:2242
-- Executing [0237691654943@trunkinbound:1] AGI("SIP/300-00004c00", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- <SIP/300-00004c00>AGI Script agi-DID_route.agi completed, returning 0
-- Executing [9998811112@default:1] Wait("SIP/300-00004c00", "2") in new stack
-- Executing [9998811112@default:2] Answer("SIP/136.243.57.214-00004bff", "") in new stack
-- Executing [9998811112@default:3] Playback("SIP/136.243.57.214-00004bff", "ss-noservice") in new stack
-- <SIP/136.243.57.214-00004bff> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [9998811112@default:2] Answer("SIP/300-00004c00", "") in new stack
> 0x7f6aac50a3c0 -- Strict RTP switching source address to 154.72.162.228:8790
-- Executing [9998811112@default:3] Playback("SIP/300-00004c00", "ss-noservice") in new stack
-- <SIP/300-00004c00> Playing 'ss-noservice.gsm' (language 'en')
> 0x7f6aac50a3c0 -- Strict RTP learning complete - Locking on source address 154.72.162.228:8790
here is my trunk configuration:
Registration string: register => username:password@sip ip:5060
account entry:
[voip]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
type=friend
username=username
secret=password
host=XXXXXXXX
dtmfmode=rfc2833
context=trunkinbound
Dial plan entry:
exten => _0237.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0237.,3,Dial(${VOIPTRUNK}/${EXTEN:2},,tTo)
exten => _0237.,4,Hangup
sip show registratry =>ok
sip show peers =>ok
here is the CLI output:
= Using SIP RTP CoS mark 5
> 0x7f6aac50a3c0 -- Strict RTP learning after remote address set to: 154.72.162.24:2242
-- Executing [0237691654943@trunkinbound:1] AGI("SIP/300-00004c00", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- <SIP/300-00004c00>AGI Script agi-DID_route.agi completed, returning 0
-- Executing [9998811112@default:1] Wait("SIP/300-00004c00", "2") in new stack
-- Executing [9998811112@default:2] Answer("SIP/136.243.57.214-00004bff", "") in new stack
-- Executing [9998811112@default:3] Playback("SIP/136.243.57.214-00004bff", "ss-noservice") in new stack
-- <SIP/136.243.57.214-00004bff> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [9998811112@default:2] Answer("SIP/300-00004c00", "") in new stack
> 0x7f6aac50a3c0 -- Strict RTP switching source address to 154.72.162.228:8790
-- Executing [9998811112@default:3] Playback("SIP/300-00004c00", "ss-noservice") in new stack
-- <SIP/300-00004c00> Playing 'ss-noservice.gsm' (language 'en')
> 0x7f6aac50a3c0 -- Strict RTP learning complete - Locking on source address 154.72.162.228:8790