Hi Kevin, I have two asterisk boxes with the same issues. Box 1: asterisk ver 1.4.21.2 Box 2: Asterisk 1.8.7.1
setup: CDMA Phone <> CDMA Media Gateway WCM <sip> 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? Thanks, Dave > -------- Original Message -------- > Subject: Re: [asterisk-users] Loss of RTP stream during DTMF collection > From: "Kevin P. Fleming" <kpflem...@digium.com> > 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: kflem...@digium.com | SIP: kpflem...@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 -- _____________________________________________________________________ -- 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