[asterisk-users] Voicemail cutting out after about 30 seconds

[Chris Bagnall]

Greetings list,

I've had problems on a few of our asterisk boxes where voicemail tends to cut out after about 30 seconds, despite the maximum message length being set at 240s (4m).

I've tried reducing the silence detection threshold from 128 down to 32, which has helped, but not resolved the issue entirely.

The calls in question are being delivered to the boxes via IAX or SIP, so it's not a Zaptel gain issue.

Has anyone else had similar problems? Is it "safe" to drop the silence detection threshold even lower?

Thanks in advance.

Regards,

Chris

[Steven Howes]

Hi,

We saw this between Asterisk and an Audiocodes gateway. Whilst the
voicemail is being recorded asterisk is not sending *ANY* rtp. Silence
detection will always detect silence if it listens to this side of the
conversation. Adjusting the threshold wont work, you need to find the
timeout for the gateway which is doing it (not always easy...)

Steve

Post by Chris Bagnall
Greetings list,
I've had problems on a few of our asterisk boxes where voicemail
tends to cut out after about 30 seconds, despite the maximum message
length being set at 240s (4m).
I've tried reducing the silence detection threshold from 128 down to
32, which has helped, but not resolved the issue entirely.
The calls in question are being delivered to the boxes via IAX or
SIP, so it's not a Zaptel gain issue.
Has anyone else had similar problems? Is it "safe" to drop the
silence detection threshold even lower?
Thanks in advance.
Regards,
Chris
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[Gordon Henderson]

Post by Chris Bagnall
Greetings list,
I've had problems on a few of our asterisk boxes where voicemail tends
to cut out after about 30 seconds, despite the maximum message length
being set at 240s (4m).
I've tried reducing the silence detection threshold from 128 down to 32,
which has helped, but not resolved the issue entirely.
The calls in question are being delivered to the boxes via IAX or SIP,
so it's not a Zaptel gain issue.
Has anyone else had similar problems? Is it "safe" to drop the silence
detection threshold even lower?

Try:

In /etc/asterisk/asterisk.conf, under

[options]

add

transmit_silence_during_record = yes

That stopped a remote site I was taking VoIP calls from hanging up after
30 seconds in a custom recording setup...

Gordon