Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch => Realtime/macro-conference exten => s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --) exten => s,n,Answer exten => s,n,Wait(1) exten => s,n,MeetMe(|cdIps) exten => s,n,Playback(vm-goodbye) exten => s,n,Hangup Thank for any help. Kind Regards Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080130/2cf68cf0/attachment.htm
ztdummy can give you issues as a timing device. Any way you could try using a Digium card just as a timing device to see if this helps? ----- Original Message ----- From: "Tomasz Zieleniewski" <tzieleniewski at gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Wednesday, January 30, 2008 11:23:57 AM (GMT-0500) America/New_York Subject: [asterisk-users] Meetme voice quality problems Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch => Realtime/macro-conference exten => s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --) exten => s,n,Answer exten => s,n,Wait(1) exten => s,n,MeetMe(|cdIps) exten => s,n,Playback(vm-goodbye) exten => s,n,Hangup Thank for any help. Kind Regards Tomasz -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 fwebb at imminc.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080130/2c6bdd5c/attachment.htm
Tomasz Zieleniewski wrote:> I am using Debian OS kernel 2.6.22-3-amd64 > and zaptel driver 1.4 with ztdummy module for meetme application. > I use meetme with SIP channels. > > I have such problem that when one connects to the conference voice is > "cut". > Each voice sequence is disturbed. > > Does any one have similar issue and could give me some advice??Tomasz, Have you run zttest on the system? It verifies the accuracy of your timing source. Digium recommends an accuracy of at least 99.98%. If your accuracy is less than that it's probably the source of your problem. Luckily, it's a problem with multiple solutions. The following thread documents some kernel configuration changes that you can make to improve the quality of ztdummy as a timing source: Recommendations for kernel config <http://lists.digium.com/pipermail/asterisk-users/2007-October/197778.html> My preferred solution is to use an empty TDM400P as a timing source. It will cost you a little bit of money, but it's an easy way to reliably solve your problem. You'll find a few posts about it if you search the list, but this one has most of the information you'll need: Empty Wildcard TDM400P as a MeetMe timer. <http://lists.digium.com/pipermail/asterisk-users/2007-March/182005.html> Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer
Franklin wrote:> ztdummy can give you issues as a timing device.Yes and no. See below> Any way you could try using a Digium card just > as a timing device to see if this helps?Tomasz wrote:>> I am using Debian OS kernel 2.6.22-3-amd64 >> and zaptel driver 1.4 with ztdummy module for meetme >> application. I use meetme with SIP channels.Your kernel is new enough that you should be able to leverage hi-res timers (you might need to patch ztdummy), or at least a RTC set to 8192 ticks/sec. What does dmesg show after ztdummy is loaded?>> I have such problem that when one connects to the >> conference voice is "cut". Each voice sequence is >> disturbed.Do you have internal_timing=yes in asterisk.conf? This option allows Asterisk to time the RTP stream based on zaptel/ztdummy clock and not on the received RTP stream. In a MeetMe, where callers might mute themselves, the received RTP stream is all but useless for timing. Dan
On Jan 30, 2008 10:35 PM, Dan Austin <Dan_Austin at phoenix.com> wrote:> Franklin wrote: > > ztdummy can give you issues as a timing device. > Yes and no. See below > > > Any way you could try using a Digium card just > > as a timing device to see if this helps? > > > Tomasz wrote: > >> I am using Debian OS kernel 2.6.22-3-amd64 > >> and zaptel driver 1.4 with ztdummy module for meetme > >> application. I use meetme with SIP channels. > > Your kernel is new enough that you should be able to > leverage hi-res timers (you might need to patch ztdummy), > or at least a RTC set to 8192 ticks/sec. What does > dmesg show after ztdummy is loaded?it is 1024 Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.4-r3748 Zaptel Echo Canceller: MG2 ztdummy: RTC rate is 1024 how can I increase it?> > > >> I have such problem that when one connects to the > >> conference voice is "cut". Each voice sequence is > >> disturbed. > Do you have internal_timing=yes in asterisk.conf? > This option allows Asterisk to time the RTP stream > based on zaptel/ztdummy clock and not on the received > RTP stream. In a MeetMe, where callers might mute > themselves, the received RTP stream is all but useless > for timing.Yes I have it set.> > > Dan > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080131/45826b7d/attachment.htm