<div dir="ltr">I had no problem with FXS to x-lite, do you have the latest SVN trunk or one of the tarballs? We may have fixed some issues if you have an older release.<br><br><br>We have an application you can use in the dialplan called set_user<br>
<br>for example put this as the very first extension in the dialplan<br><br><extension name="set_openzap_user" continue="true"><br> <condition field="source" expression="mod_openzap"/><br>
<action application="set_user" data="${caller_id_number}@$${domain}"/><br> </condition><br></extension><br><br>if you put that in an extension at the top of your dialplan just for calls from openzap then you<br>
will make that call assume the settings of the user in the directory with an id that matches the caller id num set in openzap then fall through to the rest of the dialplan.<br><br>most things you *think* are not possible are just not forced into place =D<br>
<br>as for astribank, I have heard there are a few issues with them and I do not have one to test them. The guy who supports them is around on IRC but he tends to favor the AST that's why he calls them ASTribank so we can try to work with him if he's willing.<br>
<br><br><br><br><br><br><br><div class="gmail_quote">On Tue, Aug 12, 2008 at 5:31 PM, David Baggett <span dir="ltr"><<a href="mailto:freeswitch@baggett.org">freeswitch@baggett.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I've gotten FreeSWITCH running on this config:<br>
<br>
- Intel Atom D945GCLF in 1U rackmount<br>
- RedHat FC8<br>
- Xorcom USB Astribank-8 (FXO)<br>
- Xorcom USB Astribank-8 (FXS)<br>
- Trunk FreeSWITCH code from a few days ago<br>
- zaptel-1.4.9.2.xpp.r5566<br>
- Grandstream BT-100 SIP phones<br>
- Grandstream GXV-3000 SIP videophones<br>
- Various analog phone extensions<br>
<br>
First I tried installing on FC9. This didn't work because FC9 couldn't<br>
deal with the on-board NIC card on the D945GCLF. FC8 installed fine,<br>
though, so I just used that instead.<br>
<br>
Overall I am very impressed with FreeSWITCH. It is vastly easier to set<br>
up than Asterisk, and much cleaner. The XML config files are *so* much<br>
better than the ad hoc Asterisk config syntax.<br>
<br>
I like that I can make it run like a regular RedHat service, though I<br>
could not get it to run as user freeswitch -- it seems to want to run as<br>
root. I read through a bunch of the code and it is really well done.<br>
Thanks for a great open source project!<br>
<br>
Most things work: I set up the SIP phones and added the analog<br>
extensions, and can dial out on any phone over my POTS lines via<br>
Astribank FXO. I can receive incoming calls as well. I can<br>
videoconference between the videophones by using the code in the sample<br>
dialplan (default.xml) for "intercom" mode. Neat!<br>
<br>
Problems:<br>
<br>
Weirndess with FXS-originated calls<br>
-----------------------------------<br>
<br>
If I call from an analog (FXS) extension to a BT-100, it rings normally.<br>
But if I then pick up the SIP phone, I can hear the ring on the analog<br>
extension get choppy. Soon after, the call is dropped.<br>
<br>
If I call from an analog (FXS) extension to a GXV-3000, it rings<br>
normally, but when I pick up the SIP phone, I get an immediate busy on<br>
the analog extension.<br>
<br>
If I call from an analog (FXS) extension to an X-Lite SoftPhone, it<br>
rings normally, but when I pick up the SIP phone, no audio comes through<br>
from the analog extension.<br>
<br>
If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to*<br>
an analog (FXS) extension, it works fine.<br>
<br>
The dialplan is simple (note that I have 1- and 2-digit extensions):<br>
<br>
<br>
<br>
<br>
<!-- dial an OpenZAP<br>
channel number to get the corresponding analog extension --><br>
<br>
<br>
<br>
<extension name="OpenZAP extensions"><br>
<br>
<br>
<br>
<condition field="destination_number" expression="^(9|1[0-6])$"><br>
<br>
<br>
<br>
<action application="set" data="dialed_ext=$1"/><br>
<br>
<br>
<br>
<action application="bridge" data="OpenZAP/${dialed_ext}/1"/><br>
<br>
<br>
<br>
</condition><br>
<br>
<br>
<br>
</extension><br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<!-- dial a number in the directory to get the corresponding SIP<br>
extension; use video if possible --><br>
<br>
<br>
<extension name="local-extension"><br>
<br>
<br>
<br>
<condition field="destination_number" expression="^([3-6]\d)$"><br>
<br>
<br>
<br>
<action application="set" data="dialed_ext=$1"/><br>
<br>
<br>
<br>
<action<br>
application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action><br>
<br>
<br>
<br>
<action application="export"<br>
data="sip_invite_params=intercom=true"/><br>
<br>
<br>
<br>
<action application="export" data="sip_auto_answer=true"/><br>
<br>
<br>
<br>
<action application="bridge"<br>
data="user/${dialed_ext}@$${domain}"/><br>
<br>
<br>
<br>
</condition><br>
<br>
<br>
<br>
</extension><br>
<br>
<br>
<br>
<br>
In general, I found it a bit strange that you can't (it seems) put<br>
analog extensions into the directory. Is it true that you have to define<br>
analog extensions manually in the dialplan? That's kind of<br>
counterintuitive, especially for Asterisk users; the Asterisk Now GUI<br>
makes analog and SIP extensions look the same for all practical purposes.<br>
<br>
Echo<br>
----<br>
<br>
I was getting massive distortion on analog connections involving any<br>
GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The<br>
default value was 64. I changed it to 0 and the distortion went away,<br>
but of course I get lots of annoying echo now. Other SIP phones don't<br>
seem to have this problem. I have read in earlier posts to this list<br>
that the GXV-3000 has known problems. But the interesting thing is that<br>
it was working fine with Asterisk and the Astribanks (though on a<br>
different CPU & M/B).<br>
<br>
The Astribank seems to have its own EC, but I can't figure out how to<br>
turn it on. And I have no idea what to set the value in zt.conf to; 32<br>
seems to work better than either 16 or 64 (!).<br>
<br>
I tried fxotune but that didn't do anything: it ran for a long time and<br>
generated an /etc/fxotone that seemed to have all zero values.<br>
<br>
Basically, it would be great to have an "echo cancellation HOWTO" for<br>
OpenZap users since EC seems to be offered in several different layers.<br>
I also notice a bunch of different EC modes in the OpenZap code -- does<br>
anyone know how to try different ones out?<br>
<br>
FAX detection<br>
-------------<br>
<br>
This just doesn't seem to work. Here's the relevant code from my<br>
diaplan. (It looks a little odd because I send all unanswered calls from<br>
my POTS lines to the ext. 31 voicemail box.)<br>
<br>
<!-- ring all extensions for 20 seconds, then send to voicemail for<br>
extension 31 --><br>
<br>
<br>
<!-- we answer immediately to prevent the alarm from emitting a<br>
fax/modem tone in some cases before a previous call has settled --><br>
<br>
<br>
<extension name="incoming-astribank"><br>
<br>
<br>
<br>
<condition field="destination_number" expression="^([1-4])$"><br>
<br>
<br>
<br>
<action application="answer" /><br>
<br>
<br>
<br>
<action application="tone_detect" data="fax 1100 r +5000 transfer<br>
fax XML default" /><br>
<br>
<br>
<action application="set" data="dialed_ext=31"/><br>
<br>
<br>
<br>
<action application="export" data="dialed_ext=31"/><br>
<br>
<br>
<br>
<action application="bind_meta_app" data="1 b s<br>
execute_extension::dx XML features"/><br>
<br>
<br>
<action application="bind_meta_app" data="2 b s<br>
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/><br>
<br>
<br>
<action application="bind_meta_app" data="3 b s<br>
execute_extension::cf XML features"/><br>
<br>
<br>
<action application="set" data="transfer_ringback=${us-ring}"/><br>
<br>
<br>
<br>
<action application="set" data="hangup_after_bridge=true"/><br>
<br>
<br>
<br>
<action application="set" data="continue_on_fail=true"/><br>
<br>
<br>
<br>
<action application="db"<br>
data="insert/call_return/${dialed_ext}/${caller_id_number}"/><br>
<br>
<br>
<br>
<action application="set" data="call_timeout=20"/><br>
<br>
<br>
<br>
<action application="sleep" data="1000"/><br>
<br>
<br>
<br>
<action application="bridge"<br>
data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/36@$${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/11/1"/><br>
<br>
<br>
<action application="sleep" data="1000"/><br>
<br>
<br>
<br>
<action application="voicemail" data="default $${domain}<br>
${dialed_ext}"/><br>
<br>
<br>
</condition><br>
<br>
<br>
<br>
</extension><br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<!-- called when the tone_detect above triggers --><br>
<br>
<br>
<br>
<extension name="fax"><br>
<br>
<br>
<br>
<condition field="destination_number" expression="^fax$"><br>
<br>
<br>
<br>
<action application="set" data="transfer_ringback=${us-ring}"/><br>
<br>
<br>
<br>
<action application="set" data="hangup_after_bridge=true"/><br>
<br>
<br>
<br>
<action application="set" data="call_timeout=20"/><br>
<br>
<br>
<br>
<action application="bridge" data="OpenZAP/12/1"/><br>
<br>
<br>
<br>
</condition><br>
<br>
<br>
<br>
</extension><br>
<br>
Thanks for any help with these issues.<br>
<br>
Dave<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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</div>