I have two links in my linkset which make it 60 channels. what I am asking of is the whether the cause of my asterisk service stopping is due to bad oh323 or the version of asterisk i am using which is 1.2.5. I have tested my chan_ss7 using h323callgen and I populated it with over 64 calls though without any voice and its was successfull. I am using oh323 version 0.7.3, asterisk 1.2.5 and latest version of chan_ss7. goksie On 4/28/06, Tom Chandler <tchandle@xxxxxxxxxx> wrote: > > If you are using SS7 and E1, then one channel is taken for signaling, > leaving 30 channels for voice. SS7 will not send more calls than available > channels. You should be getting a message on the console stating no CIC was > available........ > > Tom Chandler > > > ----- Original Message ----- > *From:* Goke Aruna <goksie@xxxxxxxxx> > *To:* asterisk-ss7@xxxxxxxxxxxxxxxx > *Sent:* Friday, April 28, 2006 1:14 PM > *Subject:* Re: [asterisk-ss7] asterisk oh323 - chan-ss7 echo problem > > Can someone suggest any other h323 protocol that could > use with asterisk instead of oh323? > > I had similar problem with oh323 and each time it happens the asterisk > will stop at a call going above 30 simultaneus calls > > However, I read from voip-info and which give oh323 some ratings to oh323. > > I used A104D sangoma card with chan_ss7 and no echo. > > goksie > > On 4/28/06, Anton <anton.vazir@xxxxxxxxx> wrote: > > > > Dear Jacob, > > > > Any chance to give us any timing approximation for a next > > version? > > > > Regards, > > Anton. > > > > On 28 April 2006 17:26, Jacob Tinning wrote: > > > On Tue, 18 Apr 2006, leonimar cape wrote: > > > > Can someone give a suggestion on what I should do to > > > > omit the echo. Here is may scenario > > > > > > > > --- h323 --- chan-ss7 --- > > > > A---| |--------| |--------------| |----B > > > > --- --- --- > > > > Nextone Asterisk DMS > > > > > > The next version of chan_ss7 will include > > > enabling/disabling of the zaptel echo-canceller, which > > > probably will solve your problem. > > > > > > > The calling party can hear every words he/she say > > > > after 1 to 2 seconds. But the called party can hear A > > > > with no echo and the quality is clear. I have already > > > > tried it in both digium (TE410P) and sangoma card > > > > (AT104) and the results where the same. Is there any > > > > way that I can cancel the echo? Any particular > > > > settings that I have to change in the settings of the > > > > asterisk? > > > > > > The current version of chan_ss7 does not do anything to > > > avoid echo. It just passes the audio through the > > > channels. > > > > > > The problem is when A calls an old analog phone B. Old > > > analog phones typically bleed some of the audio back to > > > the sender. > > > > > > Ordinary synchronous telephone-networks doesn't delay the > > > audio very much ( < 10ms) so the caller will not notice > > > any echo. > > > > > > Unfortunately, ip-networks induce more delay ( > 70 ms ) > > > which will clearly sound as echo. > > > > > > To avoid the echo, the only (as far as I know) solution > > > is to insert some kind of echo-canceller "in" or "to the > > > right" of Asterisk in your drawing. > > > > > > As noted above, the next version of chan_ss7 will start > > > the zaptel echo-cancellation when a new call is made > > > (this is configurable, if you don't want > > > echo-cancellation), and stop it again when the call is > > > finished. > > > > > > Mvh. Jacob > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-ss7 mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20060428/76ad040b/attachment.htm
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