Why do we keep letting these providers get away with this? Smh<br><br>On Sunday, March 1, 2015, ik <<a href="mailto:idokan@gmail.com">idokan@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Feb 27, 2015 at 2:50 PM, I put the Who? in Mishehu <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','mishehu@freeswitch.org');" target="_blank">mishehu@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<div>I see after all these years it's still
the popular thing to think that Bezeq is the most terrible thing
in the world. :-) Be thankful you don't have to deal with some
American incumbent providers... Bezeq can be a walk in the
park... :-)<br></div></div></blockquote><div><br></div><div>You mean like answering with 200 OK on INVITE, getting 404 instead of "authentication require", dropping requests because I offer both Alaw and Ulaw and other horror stories around other SIP providers that I have worked with ? :P<br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000"><div>
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DTMF handling can be a fun little source of headache in general.<br>
<br>
To clarify, are you saying that midway through a call that the
DTMF mode changes? Or are you saying that the DTMF mode for one
call may be 2833 and the next call can randomly be inband audio?
If you are saying that the mode changes during the call, what
indication do you receive that this has happened?<br></div></div></blockquote><div><br></div><div>Different types of payloads in RTP, you can see it also in wireshark when capturing with tcpdump <br></div><div><br></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000"><div>
<br>
You can try the "liberal-dtmf" setting and see if that fixes the
issue for you, but I believe that only allows SIP NOTIFY and 2833,
and I don't believe it handles inband. (Hopefully somebody else
will correct me if I'm mistaken.)<span><font color="#888888"><br></font></span></div></div></blockquote><div><br></div><div>I'll try this<br></div><div>Thank you<br><br></div><div>Ido<br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000"><div><span><font color="#888888">
<pre cols="72">--
Yossi Neiman
On 02/27/2015 05:08 AM, ik wrote:
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<div>Hello,<br>
<br>
</div>
I have the misfortune of forced to use a telco named
Bezeq - the biggest telco in Israel.<br>
</div>
The SIP trunk they provide is very problematic.<br>
<br>
</div>
<div>The one that I cannot overcome is that sometimes
doing the call, it switches DTMF between Inband and
rfc2833, and sometimes the whole DTMF sending is either
inband or rfc2833 doing the entire call.<br>
</div>
<br>
</div>
I cannot make them to be more stable in this matter (I have
tried talking with them), and I cannot replace them (I have
tried to do so as well).<br>
</div>
<br>
How can I deal with such mess ?<br>
<br>
</div>
Thanks,<br>
Ido<br>
</div>
<br>
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</div></div><span><pre>_________________________________________________________________________
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</blockquote><br><br>-- <br><div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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