# Sending empty rtp packets

**URL:** <https://community.asterisk.org/t/sending-empty-rtp-packets/17355>\
**Category:** Asterisk Support\
**Created:** [November 1, 2007, 2:33pm UTC](https://community.asterisk.org/t/sending-empty-rtp-packets/17355 "2007-11-01T14:33:09Z")\
**Posts on this page:** 5\
**Page:** 1

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**Author:** ![aaelghat](https://avatars.discourse-cdn.com/v4/letter/a/c5a1d2/32.png) [@aaelghat](https://community.asterisk.org/u/aaelghat)\
**Post date:** [November 1, 2007, 2:33pm UTC](https://community.asterisk.org/t/sending-empty-rtp-packets/17355/1 "2007-11-01T14:33:09Z")

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I run a voicemail service, and I’m running into a situation where the line disconnects after five minutes of leaving a voicemail. It’s not that the voicemail application allow them to leave a longer message, the line just goes dead. This problem happens with my local DID numbers, but not my toll-free numbers and appears to be carrier-related.

The Asterisk CLI indicates the cause of the disconnection is that the user hung up. I’ve tried adding these settings to SIP\_Custom.conf, but that hasn’t cured the issue.

rtptimeout=1200  
rtpholdtimeout=1800  
rtpkeepalive=20

In conversations with Vitelity where I describe what’s going on and that the settings above aren’t working but should, they have said, “We have clients that do exactly what you are doing - They have custom scripts that send out blank/silent rtp packets out every x amount of seconds. Unfortunately I’m not quite sure how it’s done myself or I would definately provide you with the info… i’m sure you could find something if you dig around”.

So, how do you keep the line alive? Because it’s voicemail I can’t play beeps or a recorded message, and the rtp settings aren’t working for me. Thanks.

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**Author:** ![agni\_suresh896](https://avatars.discourse-cdn.com/v4/letter/a/f4b2a3/32.png) [@agni\_suresh896](https://community.asterisk.org/u/agni_suresh896)\
**Post date:** [November 1, 2007, 3:21pm UTC](https://community.asterisk.org/t/sending-empty-rtp-packets/17355/2 "2007-11-01T15:21:06Z")

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Could you please post your asterisk console messages.

Thanks,  
Suresh.

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**Author:** ![ianplain](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ianplain/32/1776_2.png) [@ianplain](https://community.asterisk.org/u/ianplain)\
**Post date:** [November 1, 2007, 3:38pm UTC](https://community.asterisk.org/t/sending-empty-rtp-packets/17355/3 "2007-11-01T15:38:20Z")

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This was fixed in 1.2

add  
[options]  
transmit\_silence\_during\_record = yes

to the top of your asterisk.conf

Ian

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**Author:** ![aaelghat](https://avatars.discourse-cdn.com/v4/letter/a/c5a1d2/32.png) [@aaelghat](https://community.asterisk.org/u/aaelghat)\
**Post date:** [November 1, 2007, 5:29pm UTC](https://community.asterisk.org/t/sending-empty-rtp-packets/17355/4 "2007-11-01T17:29:29Z")

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That did the trick.

I can do some testing, but do you know if technically if that setting replaces the rtpkeepalive setting in SIP.conf? Are they meant to do the same thing?

Thanks!

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**Author:** ![ianplain](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ianplain/32/1776_2.png) [@ianplain](https://community.asterisk.org/u/ianplain)\
**Post date:** [November 1, 2007, 11:04pm UTC](https://community.asterisk.org/t/sending-empty-rtp-packets/17355/5 "2007-11-01T23:04:01Z")

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Hi.

Im not sure. You will have to look at the source, but this is the setting that I use, as it works for SIP and IAX

If you go back in Bugzilla you will find the birth of this fix

Ian