Loss of RTP stream during DTMF collection
I have two asterisk boxes with the same issues.
Box 1: asterisk ver 126.96.36.199
Box 2: Asterisk 188.8.131.52
CDMA Phone <> CDMA Media Gateway WCM
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.
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
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: firstname.lastname@example.org
> 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: email@example.com | SIP: firstname.lastname@example.org | Skype: kpfleming
> 445 Jan Davis Drive NW – Huntsville, AL 35806 – USA
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