<div>Dear sir,</div>
<div> Thankx for the Mail.I have built with asterisk with jitter buffer patch for 1.2.9 for the latest one there was no change it was the same.But when u said me to compile with chan_ss7 with -DAST_JB added to the CFLAGS of the chan_ss7 Makefile if you have built Asterisk
1.2.x with the jitterbuffer patch applied this is not clear for me to Check it sir.As per my application we have Dell Power Edge with Digium card the server has 8 Gbram and Dual processor with Raid 5 .We are using the SS7 with asterisk for my calling cards business
sir.People are not buying the cards because of the Audio lost or gichhy problem of audio.we are using Completely Sip platfrom sir.If u can make the things little easy that would be fine for me sir i hope u will not mind in making the things clear.
</div>
<div> </div>
<div>regards</div>
<div>surender<br><br><br> </div>
<div><span class="gmail_quote">On 6/25/06, <b class="gmail_sendername">Patrick</b> <<a href="mailto:asterisk-list@puzzled.xs4all.nl">asterisk-list@puzzled.xs4all.nl</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">On Sun, 2006-06-25 at 06:30 +0530, Mr.Surender Reddy wrote:<br>> Hello sir,<br>> I have audio lost problem in ss7 is there any way that i
<br>> can over come this.I have tried to what ever i could do which is said<br>> in all the forums like increase the jitter buffer<br>> 4*160 like this and Clocking and all but iam unable to get a way that<br>> i can over come this
problem.If any one could give me some help that<br>> would be a great help for me sir.When there is no audio lost the<br>> Quality is just like a PSTN and there is no comparision PSTN and VOip<br>> call when the audio lost comes the voice gets choppy and
gitchhy.If<br>> anyone can give me a solution on commercial basis also is most<br>> welcome.<br>><br>> regards<br>> surender<br><br>Hello Surender,<br><br>First you need to compile chan_ss7 with -DAST_JB added to the CFLAGS of
<br>the chan_ss7 Makefile if you have built Asterisk 1.2.x with the<br>jitterbuffer patch applied.<br><br>I'm not sure if chan_ss7 supports the jitterbuffer. I was told that the<br>jitterbuffer patch is specifically for SIP-SIP calls and ZAP-SIP/SIP-ZAP
<br>calls and I don't know if the normal ZAP channels are used for SS7 too.<br><br>Perhaps you could ask the folks at Sifira if they can implement the<br>jitterbuffer on SS7 channels if you pay for it. Or ask the original
<br>author slav [a.t] <a href="http://securax.org">securax.org</a>.<br><br>Alternatively try building Asterisk without the jitterbuffer patch and<br>see what happens.<br><br>Regards,<br>Patrick<br><br>_______________________________________________
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