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Clive, cool - winter is getting quite near ova here...<br>
<br>
Well, how would I find out what is happening - I mean how do I know
what * is connecting with to net2phone. <br>
<pre wrap=""> "...They have their own proprietry protocol..."
</pre>
I thought it was because of the G723.1 codec and passthrough - but the
I must take the voice prompts way.<span class="moz-smiley-s1"><span>
:-) </span></span><br>
(Didn't thought that it'll cause a problem - just the warnings and
notices but continue still...) Thank you for that tip.<br>
<pre> "...For G723.1 passthrough, you just allow it..."
</pre>
-------------------------------------------------------------------<br>
So that is in "sip.conf"<br>
[general]<br>
disallow=all;<br>
allow=G723;<br>
allow=ulaw;<br>
allow=alaw;<br>
allow=gsm;<br>
<br>
(some text later)<br>
<br>
[net2phone]<br>
(some text)<br>
canreinvite=yes;<br>
(some text)<br>
-------------------------------------------------------------------<br>
<br>
Sources for net2phone:<br>
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Net2phone">http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Net2phone</a><br>
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/tiki-index.php?page=Net2phone">http://www.voip-info.org/tiki-index.php?page=Net2phone</a><br>
<br>
PS - I do get a frame error about expecting 4 getting 256 when * is
trying to initiate to call through to net2phone device MAX IP-10 through<br>
the net2phone network - could be that protocall you were talking
about or have I completely missed the plot?<br>
<pre class="moz-signature" cols="72">Kind Regards
Etienne
</pre>
<a class="moz-txt-link-abbreviated" href="mailto:clive@sentechsa.com">clive@sentechsa.com</a> wrote:
<blockquote cite="mid425CE507.25884.59297B@localhost" type="cite">
<pre wrap="">Etienne, howzit
I am not 100% sure about this, but Net2phone do not always use
standard SIP as the protocol. They have their own proprietry
protocol as well, so perhaps your phone is trying to talk on the
proprietry protocol.
For G723.1 passthrough, you just allow it, and it should work fine, as
long as you do not try playing any voice prompts to the channel.
good luck.
regards
Clive
=====================
Phone I.T.
<a class="moz-txt-link-freetext" href="http://www.phonehome.co.za">http://www.phonehome.co.za</a>
On 13 Apr 2005 at 8:52, Etienne Pretorius wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hello all,
I came a cross a problem yesterday that I don't quite know how to solve.
I am trying to use * to connect to net2phone, and have a net2phone MAX
IP-10 connect to net2phone. From the settings on
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/">http://www.voip-info.org/</a> it was easy to get asterisk to connect to the
network - acting like a net2phone device/user. Anyway the problem arose
when attempting to call the MAX IP-10 device through the net2phone
network. They seem to be using the G732.1 codec. I have in my settings
in sip.conf allow=G732.1 or what ever flavour of the like and still I
can not talk to the two devices. I googled a bit and came across the
fact of * being able to do a pass through - well I was not successful
and this subject is either simple or not well documented. The devices
are using SIP and there is a bridge initiated, but there is no audio and
no voice being passed through... I have tried connecting as the
receiving device a GrandStrem Budge Tone-100 and still no luck. So all
that I am inquiring is has anyone successfully done a pass through and
if so can someone please guide me through some of the settings. I have
set the [net2phone] with a canreinvite=yes - that a post on a forum also
suggested, and that also did not work.
On a separate issue: When the Grandstream Budge Tone-100 is connected on
the internal network then the audio and the voice in both directions
work fine. But when the device is connected on a separate network - ie
on an other ADSL line, then the device doesn't send voice packets
although is receives packets. I have opened up IPTABLES, to allow udp
5060 and udp 10000:20000 in both directions on any interface and the
problem still persists. (SIP phone: Grandstream Budge Tone 100 connects
to * and the call is answered by a Softphone X-Lite with all the codecs
enabled. As far as I can tell thy both are "speaking" with a G711 codec
ULaw/ALaw).
So can anyone please give me a guideline or some advise on where to look
to solve the problem.
--
Kind Regards
Etienne
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</blockquote>
<pre wrap=""><!---->
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