<div>OOOOPSS</div><div><br></div><a href="http://bit.ly/ywiwzt">http://bit.ly/ywiwzt</a><br><br><div class="gmail_quote">On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Google or click this link <a href="http://bit.ly/ywiwzteve" target="_blank">http://bit.ly/ywiwzteve</a> " Steve Totaro IAX" and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s).<div>
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Forget IAX2 and save yourself time you will never get back.</div><div><br></div><div>IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world.</div><div><br></div>
<div>Thanks for IAX2 Digium!</div><div><br></div><div>Thanks,</div><div>Steve Totaro<div><div class="h5"><br><br><div class="gmail_quote">On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford <span dir="ltr"><<a href="mailto:ttelford.groups@gmail.com" target="_blank">ttelford.groups@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?)<div>
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On 2012-02-28 21:12:48 +0000, Noah Engelberth said:<br>
<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that "couldn't hear" (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote callers) fixed the call quality issue.<br>
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<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.<u></u>digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-<u></u>bounces@lists.digium.com</a>] On Behalf Of Troy Telford<br>
Sent: Tuesday, February 28, 2012 4:08 PM<br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.<u></u>com</a><br>
Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great<br>
<br>
On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity.<br>
<br>
Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem:<br>
<br>
With IAX2:<br>
- Incoming Voice from my Provider -> Asterisk = Sounds great<br>
- Outgoing Voice from Asterisk -> my Provider = Sounds terrible<br>
<br>
By "terrible," I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw).<br>
<br>
On the other hand:<br>
With SIP:<br>
- Incoming Voice from my Provider -> Asterisk = Sounds great<br>
- Outgoing Voice from Asterisk -> my Provider = Sounds great<br>
<br>
The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system).<br>
<br>
The server for my provider is identical in either case. So I figure it's one of a few things:<br>
- misconfiguration<br>
- My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP<br>
- If there's something I can do here, I'd like to know, but I doubt it.<br>
- a problem with my provider<br>
- In which I'll contact them.<br>
<br>
For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward:<br>
[general]<br>
bandwidth=low<br>
jitterbuffer=yes<br>
forcejitterbuffer=no<br>
encryption = yes<br>
autokill=yes<br>
maxcallnumbers=12<br>
maxcallnumbers_nonvalidated=4<br>
<br>
[guest]<br>
type=user<br>
context=default<br>
callerid="Guest IAX User"<br>
<br>
[myprovider]<br>
type=friend<br></div></div>
usernamesecretcontext=<u></u>somecontext<div><div><br>
host=provider_server<br>
qualify=1000<br>
disallow=all<br>
allow=g729<br>
allow=ulaw<br>
auth=md5,rsa<br>
requirecalltoken=yes<br>
trunk=yes<br>
<br>
Firewall:<br>
Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to "just work" - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing).<br>
<br>
I have configured my firewall to forward incoming connections on port<br>
4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic).<br>
<br>
Outgoing connections are fairly typical for a NAT setup - anything can go out.<br>
<br>
Any other ideas before I give up on using IAX?<br>
Thanks<br>
--<br>
Troy Telford<br>
<br>
<br>
<br>
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The message does not contain any threats<br>
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</blockquote><div><div>
<br>
<br>
-- <br>
Troy Telford<br>
<br>
<br>
<br>
--<br>
______________________________<u></u>______________________________<u></u>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
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asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
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</blockquote></div><br>