FROM ARGENTINA
Good evening !! I´m newbie in Asterix and I´m stuck in a problem since two days ago.
My problem is:
I have my device registered, and I can see it through the web, not only for the fxo port but for the fxs one as well. I can also make outgoing calls perfectly.
The problem is with incoming calls, which are detected by the device and forwarded to the corresponding extension (the one that holds the Menu). I can watch this through the Cli console but my headache is that no sound s played. In the console I can see the final message (Goodbye message) but obviously I can´t hear it. Inmediately it closes the call.
The most curious issue (unless for me) is that if I call the extension from another inner extension, I can hear perfectly the message. The same happens (the sound is played and I can hear the message perfectly) if I call from one DDI incoming line that I have configured.
Any ideas or help will be appreciated !!
Thanks in advance
CLI CONSOLE
– Goto (day-menu,s,1)
– Executing [s@day-menu:1] Answer(“SIP/ata-fxo-00000116”, “”) in new stack
– Executing [s@day-menu:2] Wait(“SIP/ata-fxo-00000116”, “0.5”) in new stack
– SIP/troncalVOIPMS-00000115 answered SIP/200-00000114
– Executing [s@day-menu:3] BackGround(“SIP/ata-fxo-00000116”, “tt-monkeys”) in new stack
– <SIP/ata-fxo-00000116> Playing ‘tt-monkeys.alaw’ (language ‘es’)
== Spawn extension (users, 46375627, 1) exited non-zero on ‘SIP/200-00000114’
– Executing [s@day-menu:4] WaitExten(“SIP/ata-fxo-00000116”, “5”) in new stack
– Timeout on SIP/ata-fxo-00000116, going to ‘t’
– Executing [t@day-menu:1] Playback(“SIP/ata-fxo-00000116”, “goodbye”) in new stack
– <SIP/ata-fxo-00000116> Playing ‘goodbye.gsm’ (language ‘es’)
– Executing [t@day-menu:2] Hangup(“SIP/ata-fxo-00000116”, “”) in new stack
== Spawn extension (day-menu, t, 2) exited non-zero on ‘SIP/ata-fxo-00000116’
Archivo sip
[general]
context=noasignado
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
disallow=all
allow=alaw
language=es
[ata-fxo]
type=friend
host=dynamic
secret=S3cr3t0
context=features
mailbox=400@default
qualify=yes
Configuraciones HT 503
STATUS
Product Model: HT-503 V1.1B
Software Version: Program-- 1.0.5.5 Bootloader-- 1.0.0.15 Core-- 1.0.5.2 Base-- 1.0.5.2
System Up Time: 09:46:05 up 1:06
PPPoE Link Up: Disabled
NAT:
Port Status:
Port Hook Registration DND Forward Busy Forward Delayed Forward
FXS On Hook Registered No
FXO Idle Registered No
PUERTO FXO
Preferred DTMF method: 1 Prioridad RFC2833
Preferred Vocoder:
(in listed order)
choice 1: pcmu
choice 2: pcma
choice 3: 723
choice 4: 729
choice 5: 726-32
choice 6: ilbc
choice 7: g729e
choice 8: aal2
PUERTO FXS
unregister on reboot yes
Prefered DTMF method Priority 1 RFC2833
uncondicional call forward to VOIP
507 67.22.XXXX 5060
DE NUEVO MUCHAS GRACIAS,