Unable to make outbound calls

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Unable to make outbound calls

Postby Boutchou » 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
VERSION: 2.14-730a
BUILD: 191121-2256
DB Schema Version: 1579
asterisk-13.17.2-vici
Boutchou
 
Posts: 14
Joined: Wed Oct 16, 2019 4:32 am

Re: Unable to make outbound calls

Postby carpenox » Mon Apr 27, 2020 12:38 am

not smart to put your Ip's SIP attacks suck, js. stupid question but does your SIP trunk have active channels that youve paid for?
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
Posts: 2230
Joined: Wed Apr 08, 2020 2:02 am
Location: Coral Springs, FL


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