Dave George
2012-May-25 21:30 UTC
[asterisk-users] Loss of RTP stream during DTMF collection
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? Dave
Kevin P. Fleming
2012-May-25 21:38 UTC
[asterisk-users] Loss of RTP stream during DTMF collection
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 at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
Dave George
2012-May-25 23:30 UTC
[asterisk-users] Loss of RTP stream during DTMF collection
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" <kpfleming at digium.com> > Date: Fri, May 25, 2012 5:38 pm > To: asterisk-users at 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 at digium.com | SIP: kpfleming at 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