[Freeswitch-users] Installation: good, but some issues
Brian West
brian at freeswitch.org
Tue Aug 12 16:23:08 PDT 2008
I would recommend you open a Jira on the audio issue. http://jira.freeswitch.org
so we don't loose track of the issue. I know there is a problem
with the astribank not hanging up properly.
/b
On Aug 12, 2008, at 5:31 PM, David Baggett wrote:
> I've gotten FreeSWITCH running on this config:
>
> - Intel Atom D945GCLF in 1U rackmount
> - RedHat FC8
> - Xorcom USB Astribank-8 (FXO)
> - Xorcom USB Astribank-8 (FXS)
> - Trunk FreeSWITCH code from a few days ago
> - zaptel-1.4.9.2.xpp.r5566
> - Grandstream BT-100 SIP phones
> - Grandstream GXV-3000 SIP videophones
> - Various analog phone extensions
>
> First I tried installing on FC9. This didn't work because FC9 couldn't
> deal with the on-board NIC card on the D945GCLF. FC8 installed fine,
> though, so I just used that instead.
>
> Overall I am very impressed with FreeSWITCH. It is vastly easier to
> set
> up than Asterisk, and much cleaner. The XML config files are *so* much
> better than the ad hoc Asterisk config syntax.
>
> I like that I can make it run like a regular RedHat service, though I
> could not get it to run as user freeswitch -- it seems to want to
> run as
> root. I read through a bunch of the code and it is really well done.
> Thanks for a great open source project!
>
> Most things work: I set up the SIP phones and added the analog
> extensions, and can dial out on any phone over my POTS lines via
> Astribank FXO. I can receive incoming calls as well. I can
> videoconference between the videophones by using the code in the
> sample
> dialplan (default.xml) for "intercom" mode. Neat!
>
> Problems:
>
> Weirndess with FXS-originated calls
> -----------------------------------
>
> If I call from an analog (FXS) extension to a BT-100, it rings
> normally.
> But if I then pick up the SIP phone, I can hear the ring on the analog
> extension get choppy. Soon after, the call is dropped.
>
> If I call from an analog (FXS) extension to a GXV-3000, it rings
> normally, but when I pick up the SIP phone, I get an immediate busy on
> the analog extension.
>
> If I call from an analog (FXS) extension to an X-Lite SoftPhone, it
> rings normally, but when I pick up the SIP phone, no audio comes
> through
> from the analog extension.
>
> If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone
> *to*
> an analog (FXS) extension, it works fine.
>
> The dialplan is simple (note that I have 1- and 2-digit extensions):
>
>
>
>
> <!-- dial an OpenZAP
> channel number to get the corresponding analog extension -->
>
>
>
> <extension name="OpenZAP extensions">
>
>
>
> <condition field="destination_number" expression="^(9|1[0-6])$">
>
>
>
> <action application="set" data="dialed_ext=$1"/>
>
>
>
> <action application="bridge" data="OpenZAP/${dialed_ext}/1"/>
>
>
>
> </condition>
>
>
>
> </extension>
>
>
>
>
>
>
>
> <!-- dial a number in the directory to get the corresponding SIP
> extension; use video if possible -->
>
>
> <extension name="local-extension">
>
>
>
> <condition field="destination_number" expression="^([3-6]\d)$">
>
>
>
> <action application="set" data="dialed_ext=$1"/>
>
>
>
> <action
> application="export"><![CDATA[sip_h_Call-Info=<sip:$$
> {domain}>;answer-after=0]]></action>
>
>
>
> <action application="export"
> data="sip_invite_params=intercom=true"/>
>
>
>
> <action application="export" data="sip_auto_answer=true"/>
>
>
>
> <action application="bridge"
> data="user/${dialed_ext}@$${domain}"/>
>
>
>
> </condition>
>
>
>
> </extension>
>
>
>
>
> In general, I found it a bit strange that you can't (it seems) put
> analog extensions into the directory. Is it true that you have to
> define
> analog extensions manually in the dialplan? That's kind of
> counterintuitive, especially for Asterisk users; the Asterisk Now GUI
> makes analog and SIP extensions look the same for all practical
> purposes.
>
> Echo
> ----
>
> I was getting massive distortion on analog connections involving any
> GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf.
> The
> default value was 64. I changed it to 0 and the distortion went away,
> but of course I get lots of annoying echo now. Other SIP phones don't
> seem to have this problem. I have read in earlier posts to this list
> that the GXV-3000 has known problems. But the interesting thing is
> that
> it was working fine with Asterisk and the Astribanks (though on a
> different CPU & M/B).
>
> The Astribank seems to have its own EC, but I can't figure out how to
> turn it on. And I have no idea what to set the value in zt.conf to; 32
> seems to work better than either 16 or 64 (!).
>
> I tried fxotune but that didn't do anything: it ran for a long time
> and
> generated an /etc/fxotone that seemed to have all zero values.
>
> Basically, it would be great to have an "echo cancellation HOWTO" for
> OpenZap users since EC seems to be offered in several different
> layers.
> I also notice a bunch of different EC modes in the OpenZap code --
> does
> anyone know how to try different ones out?
>
> FAX detection
> -------------
>
> This just doesn't seem to work. Here's the relevant code from my
> diaplan. (It looks a little odd because I send all unanswered calls
> from
> my POTS lines to the ext. 31 voicemail box.)
>
> <!-- ring all extensions for 20 seconds, then send to voicemail for
> extension 31 -->
>
>
> <!-- we answer immediately to prevent the alarm from emitting a
> fax/modem tone in some cases before a previous call has settled -->
>
>
> <extension name="incoming-astribank">
>
>
>
> <condition field="destination_number" expression="^([1-4])$">
>
>
>
> <action application="answer" />
>
>
>
> <action application="tone_detect" data="fax 1100 r +5000
> transfer
> fax XML default" />
>
>
> <action application="set" data="dialed_ext=31"/>
>
>
>
> <action application="export" data="dialed_ext=31"/>
>
>
>
> <action application="bind_meta_app" data="1 b s
> execute_extension::dx XML features"/>
>
>
> <action application="bind_meta_app" data="2 b s
> record_session::$${base_dir}/recordings/${caller_id_number}.$
> {strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>
>
> <action application="bind_meta_app" data="3 b s
> execute_extension::cf XML features"/>
>
>
> <action application="set" data="transfer_ringback=${us-ring}"/>
>
>
>
> <action application="set" data="hangup_after_bridge=true"/>
>
>
>
> <action application="set" data="continue_on_fail=true"/>
>
>
>
> <action application="db"
> data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
>
>
>
> <action application="set" data="call_timeout=20"/>
>
>
>
> <action application="sleep" data="1000"/>
>
>
>
> <action application="bridge"
> data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/
> 36@$${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/
> 11/1"/>
>
>
> <action application="sleep" data="1000"/>
>
>
>
> <action application="voicemail" data="default $${domain}
> ${dialed_ext}"/>
>
>
> </condition>
>
>
>
> </extension>
>
>
>
>
>
>
>
> <!-- called when the tone_detect above triggers -->
>
>
>
> <extension name="fax">
>
>
>
> <condition field="destination_number" expression="^fax$">
>
>
>
> <action application="set" data="transfer_ringback=${us-ring}"/>
>
>
>
> <action application="set" data="hangup_after_bridge=true"/>
>
>
>
> <action application="set" data="call_timeout=20"/>
>
>
>
> <action application="bridge" data="OpenZAP/12/1"/>
>
>
>
> </condition>
>
>
>
> </extension>
>
> Thanks for any help with these issues.
>
> Dave
>
>
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Brian West
sip:brian at freeswitch.org
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