[Freeswitch-users] openzap / dialplan / Sangoma A204 questions

[Anthony Minessale]

Can you try upgrading to the latest svn build of FS.
There are several fixes to openzap in there that I know will fix your issue.

BTW
Let me know how it's going, we have not actually seen anyone try openzap on
BSD before.


On Mon, Jun 30, 2008 at 7:29 PM, John Wehle <john at feith.com> wrote:

> I have a A204 with hardware echo cancelling and two FXO modules
> on FreeBSD 6.2 connected to tip-ring lines from a PBX.  ztcfg reports:
>
>  Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>  Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>  Channel 03: FXS Kewlstart (Default) (Slaves: 03)
>  Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>
> Hooking a plain old telephone to the tip-ring lines from the PBX
> works fine.  On startup freeswitch reports:
>
>  [DEBUG] zap_io.c:1951 load_config() found config for span
>  [DEBUG] zap_io.c:1978 load_config() created span 1 of type zt
>  [DEBUG] zap_io.c:1991 load_config() span 1 [name]=[OpenZAP]
>  [DEBUG] zap_io.c:1991 load_config() span 1 [number]=[551]
>  [DEBUG] zap_io.c:1991 load_config() span 1 [fxo-channel]=[1]
>  [DEBUG] zap_io.c:2020 load_config() setting trunk type to 'FXO'
> start(KEWL)
>  [WARNING] zap_zt.c:135 zt_open_range() this ioctl fails on older zaptel
> but is harmless if you used ztcfg
>  [device /dev/zap/channel chan 1 fd 26 (Inappropriate ioctl for device)]
>  [INFO] zap_zt.c:170 zt_open_range() configuring device /dev/zap/channel
> channel 1 as OpenZAP device 1:1 fd:25
>
> Ultimately I want freeswitch to run a script when any of the FXO lines
> receive a call.  Playing around produced some questions:
>
>  1) I have a dialplan of:
>
>    <extension name="outgoing-fxo">
>      <condition field="destination_number" expression="^55[1-4]$">
>        <action application="set" data="dialed_ext=482"/>
>        <action application="bridge" data="openzap/1/1/${dialed_ext}"/>
>      </condition>
>    </extension>
>
>    which I'm assuming will cause freeswitch to use the fxo to dial 482
>    on the PBX when routing a call to 551.  When I dial 551 from a VoIP
>    phone I see:
>
>    [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel
> OpenZAP/1:1/482 [edf17e96-0247-dd11-9800-001fc6ab49e2]
>    [WARNING] zap_analog.c:52 analog_fxo_outgoing_call() VETO Changing state
> on 1:1 from DOWN to DIALING
>    [WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:1
>
>    however I don't hear anything on the VoIP phone (i.e. no ringing) and
>    extension 482 which is right next to the VoIP doesn't ring.
>
>  2) What would my dialplan look like so that dialing 551 bridges the call
>     to the FXO with the FXO just going off hook ... not dialing?
>     I.e. dialing 551 just gets me a PBX line with dialtone.
>
>  3) What condition would I use in my dialplan to match an FXO line ringing?
>     I.e. when the FXO line rings I want to invoke javascript.
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495  |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080701/5ad4df54/attachment-0002.html