Loss of RTP stream during DTMF collection

Home » Asterisk Users » Loss of RTP stream during DTMF collection
Asterisk Users 1 Comment

Hi Kevin,

I have two asterisk boxes with the same issues.
Box 1: asterisk ver
Box 2: Asterisk

CDMA Phone <> CDMA Media Gateway WCM Asterisk voice mail

The calls are SIP Based. DTMF collection is when the user is entering a
password for voice mail access or voucher to recharge their account.

voice mail:
user is prompted for a password. After password is entered I can see
asterisk playing the voice mail but no audio is heard on the phone.

Other scenario user dials into a voucher menu (Asterisk2billing) and is
prompted for a voucher. No audio after the voucher is entered.

The CDMA guys did a trace on their end and this is what they explained
is happening:

The voicemail problem is due to the time stamp jump on the RTP steam
sending WCM to BSC. There are about 5 seconds gap between two
consecutive RTP packets. It was caused by Asterisk not sending any RTP
packet to WCM.

How can I enable the option to allow asterisk to maintain the RTP stream
during DTMF collection?


> ——– Original Message ——–
> Subject: Re: [asterisk-users] Loss of RTP stream during DTMF collection
> From: “Kevin P. Fleming”
> Date: Fri, May 25, 2012 5:38 pm
> To: asterisk-users@lists.digium.com
> On 05/25/2012 04:30 PM, Dave George wrote:
> > I am using asterisk for voice mail. During DTMF collection Asterisk
> > stop sending any RTP Packets. The gap between two consecutive packets
> > are 4 seconds, which is huge enough to screw up the jitter buffer. When
> > ever asterisk stops to receive DTMF, the RTP stream is cut and we loose
> > audio.
> >
> > I don’t have this issue when calling from a SIP phone. I only have this
> > issue when calling from one media gateway to the asterisk box.
> >
> > Any suggestions welcome. Can I play some file in the back while
> > collecting DTMF?
> You are missing quite a lot of crucial information required for anyone
> to help you. First, what version of Asterisk are you using? Second, what
> type of channel is being used to connect to Asterisk? You mention it
> works from a SIP phone, but not from a media gateway.. is that gateway
> also using SIP, or something else? What does ‘during DTMF collection’
> mean? Do you mean after a prompt has been played and the voicemail
> application is waiting for input, or is this during prompt playback, or
> something else?
> Quite some time ago Asterisk was changed to ensure that silence would be
> sent while an application was running and waiting for input from the
> caller; if your version is older than this, then that could explain what
> you are seeing. That’s just a mildly-educated guess though.
> —
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW – Huntsville, AL 35806 – USA
> Check us out at www.digium.com & www.asterisk.org
> —
> _____________________________________________________________________
> — Bandwidth and Colocation Provided by http://www.api-digital.com
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users

One thought on - Loss of RTP stream during DTMF collection

  • If it’s the problem I hypothesized it was, you can set
    ‘transmit_silence=yes’ in your asterisk.conf file.