Voicemail works internally but not externally

Asterisk Asterisk General
1

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
2

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

3

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

4

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…

5

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

6

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