<div>Dear sir,</div>
<div> Thankx for ur time and ur concern on me.we have planned to replace the system or to find people who can do this Steve from <a href="http://Astcode.com">Astcode.com</a> is also trying really hard to solve this problem may be he will solve this
soon.thankx again sir.</div>
<div> </div>
<div>regards</div>
<div>surender<br><br> </div>
<div><span class="gmail_quote">On 6/27/06, <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 Mon, 2006-06-26 at 19:51 +0530, Mr.Surender Reddy wrote:<br>> Dear sir,<br>> Thankx for the
Mail.I have built with asterisk with<br>> jitter buffer patch for 1.2.9 for the latest one there was no change<br>> it was the same.But when u said me to compile with chan_ss7 with<br>> -DAST_JB added to the CFLAGS of the chan_ss7 Makefile if you have
<br>> built Asterisk 1.2.x with the jitterbuffer patch applied this is not<br>> clear for me to<br><br>If you have built & installed asterisk 1.2.9 *with* the jitterbuffer<br>patch then I think you need to add -DAST_JB to the CFLAGS in the
<br>Makefile of chan_ss7 and rebuild and install it. If you *don't* use the<br>jitterbuffer patch with asterisk 1.2.9 then you do *not* need to add<br>-DAST_JB to the CFLAGS of the chan_ss7 Makefile. Hope this is clearer<br>
now.<br><br>> Check it sir.As per my application we have Dell Power Edge with Digium<br>> card the server has 8 Gbram and Dual processor with Raid 5 .<br><br>Make sure that the Digium card is on it's own interrupt and that the
<br>Digium card is *not sharing* an interrupt with something else. I think<br>there is background information about this on <a href="http://www.asteriskguru.com">www.asteriskguru.com</a><br><br>> We are using the SS7 with asterisk for my calling cards business
<br>> sir.People are not buying the cards because of the Audio lost or<br>> gichhy problem of audio.we are using Completely Sip platfrom sir.If u<br>> can make the things little easy that would be fine for me sir i hope u
<br>> will not mind in making the things clear.<br><br>If you continue to have these problems I suggest you hire a local<br>Asterisk consultant to help you find and hopefully fix the problems.<br><br>Regards,<br>Patrick
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