Voicemail works internally but not externally

Hey guys,

Kinda stuck on this one… Voicemail working fine when calling from an internal extension to another internal extension… But when I call from my cell the extension rings and when Voicemail is expected to pick up I get a fast busy signal and the call hangs up…

Any help greatly appreciated as always —

Call from External here –

Using SIP RTP CoS mark 5
> 0x7f3bc806f1f0 – Strict RTP learning after remote address set to: XXX.XXX.XXX.XXX:18510
– Executing [XXX2520999@from-trunk:1] Dial(“SIP/VOIP-00000040”, “SIP/XXX2520999,30,r”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/XXX2520999
– SIP/XXX2520999-00000041 is ringing
== Spawn extension (from-trunk, XXX2520999, 1) exited non-zero on ‘SIP/VOIP-00000040’

========================================================

Call from internal here –

Using SIP RTP CoS mark 5
– Called SIP/XXX2520999@VOIP
== Using SIP RTP CoS mark 5
> 0x7f3bc80606c0 – Strict RTP learning after remote address set to: XXX.XXX.XXX.XXX:14852
– Executing [XXX2520999@from-trunk:1] Dial(“SIP/VOIP-00000052”, “SIP/XXX2520999,30,r”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/XXX2520999
– SIP/VOIPMS-00000051 is ringing
– Got SIP response 486 “Busy Here” back from XXX.XXX.XXX.XXX:1379
– SIP/XXX2520999-00000053 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [XXX2520999@from-trunk:2] VoiceMail(“SIP/VOIP-00000052”, “XXX2520999@default,u”) in new stack
> 0x5598774d7100 – Strict RTP learning after remote address set to: XXX.XXX.XXX.XXX:11716
– SIP/VOIP-00000051 answered SIP/XXX3920220-00000050
– Channel SIP/VOIP-00000051 joined ‘simple_bridge’ basic-bridge <87ebb9d6-8b58-4e4e-a13e-283305d0db8e>
– Channel SIP/XXX3920220-00000050 joined ‘simple_bridge’ basic-bridge <87ebb9d6-8b58-4e4e-a13e-283305d0db8e>
> Bridge 87ebb9d6-8b58-4e4e-a13e-283305d0db8e: switching from simple_bridge technology to native_rtp
> Locally RTP bridged ‘SIP/XXX3920220-00000050’ and ‘SIP/VOIP-00000051’ in stack
> 0x7f3bc806f1f0 – Strict RTP qualifying stream type: audio
> 0x7f3bc806f1f0 – Strict RTP switching source address to XXX.XXX.XXX.XXX:5004
> 0x7f3bc80606c0 – Strict RTP switching to RTP target address XXX.XXX.XXX.XXX:14852 as source
– <SIP/VOIP-00000052> Playing ‘vm-theperson.ulaw’ (language ‘en’)
> 0x5598774d7100 – Strict RTP switching to RTP target address XXX.XXX.XXX.XXX:11716 as source
> 0x7f3bc806f1f0 – Strict RTP learning complete - Locking on source address XXX.XXX.XXX.XXX:5004
> 0x7f3bc80606c0 – Strict RTP learning complete - Locking on source address XXX.XXX.XXX.XXX:14852
– <SIP/VOIP-00000052> Playing ‘digits/2.ulaw’ (language ‘en’)
> 0x5598774d7100 – Strict RTP learning complete - Locking on source address XXX.XXX.XXX.XXX:11716
– <SIP/VOIP-00000052> Playing ‘digits/5.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘digits/2.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘digits/0.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘digits/9.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘digits/9.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘digits/9.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘vm-isunavail.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘vm-intro.ulaw’ (language ‘en’)
– <SIP/VOIP-00000052> Playing ‘beep.ulaw’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/XXX2520999/tmp/L0XiUy format: wav49, 0x7f3bb4006450
– x=1, open writing: /var/spool/asterisk/voicemail/default/XXX2520999/tmp/L0XiUy format: gsm, 0x7f3bb4019240
– x=2, open writing: /var/spool/asterisk/voicemail/default/XXX2520999/tmp/L0XiUy format: wav, 0x7f3bb400be20
– User ended message by pressing #
– <SIP/VOIP-00000052> Playing ‘auth-thankyou.ulaw’ (language ‘en’)
– Executing [XXX2520999@from-trunk:3] Hangup(“SIP/VOIP-00000052”, “”) in new stack
== Spawn extension (from-trunk, XXX2520999, 3) exited non-zero on ‘SIP/VOIP-00000052’
– Channel SIP/VOIP-00000051 left ‘native_rtp’ basic-bridge <87ebb9d6-8b58-4e4e-a13e-283305d0db8e>
– Channel SIP/XXX3920220-00000050 left ‘native_rtp’ basic-bridge <87ebb9d6-8b58-4e4e-a13e-283305d0db8e>
== Spawn extension (from-internal, XXX2520999, 23) exited non-zero on ‘SIP/XXX3920220-00000050’
====================

extensions.conf

[from-internal]

include=>parkedcalls

exten=>XXX2520999,hint,SIP/XXX2520999
exten=>XXX2520999,1,Dial(SIP/XXX2520999,30,r)
exten=>XXX2520999,n,Voicemail(XXX2520999@default,u)
exten=>XXX2520999,n,Hangup

[from-trunk]

exten => s,1,Set(HEADER_TO=${SIP_HEADER(To)})
exten => s,n,Set(CALLED_NUMBER=${HEADER_TO:5:12})
exten => s,n,Goto(${CALLED_NUMBER:2},1)

exten=>XXX2520999,1,Dial(SIP/XXX2520999,30,r)
exten=>XXX2520999,n,Voicemail(XXX2520999@default,u)
exten=>XXX2520999,n,Hangup


sip.conf

[friends_internal](!)
disallow=all
type=peer
nat=force_rport,comedia
host=dynamic
context=from-internal
;allow=all
allow=ulaw
directmedia=no
busylevel=1
allowsubscribe=yes

[XXX2520999](friends_internal)
secret=
mailbox=XXX2520999@default
qualify=yes
keepalive=yes

Your dial timeout is 30 seconds . Mobile networks, at least, tend to timeout after 20 seconds.

Thank you David… Now I feel stupid LOL :slight_smile: changed the timeout to 20 seconds and works fine now.

Another thing I just noticed… guess I missed it because I was too close to it… by adding

exten=>XXX2520999,1,Answer()

at the top of the order… then the timeout doesn’t matter as much because Asterisk already answered the call as far as the external source is concerned…

The disadvantage, there is that the caller may get billed.