Thanks Michael,
Here is my extension:
<extension name="mad_boss_intercom">
<condition field="destination_number" expression="^0911$">
<action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/>
<action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
<action application="set" data="conference_auto_outcall_timeout=60"/>
<action application="set" data="sip_exclude_contact=${network_addr}"/>
<action application="conference_set_auto_outcall" data="[sip_auto_answer=true]user/1003@${domain_name}"/>
<action application="conference_set_auto_outcall" data="user/1004@${domain_name}"/>
<action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/>
</condition>
</extension>
When I call from, for example, 1002, both 1003 and 1004 pick up and broadcast the page, and when I hang up, all legs get torn down.
When I call from 1003, however, deaf legs to both 1003 and 1004 are set up, but I expect the sip_exclude_contact setting to ensure no leg is set up to 1003. When I hang up, the "page" leg hangs up, but the deaf legs are still up, and the phone at 1003 has its deaf leg on hold. I need to resume the leg and hang it up for the conference to end.
The paste is from this second scenario.
I looked in the code, and it looks src/mod/endpoints/mod_sofia/mod_sofia.c does an sql lookup and excludes anything "like" the network_addr in the contact header, but I couldn't trace back how this is actually used. Maybe I'm using this extension wrong. Does this work only with groups? (Note that I haven't figured out yet how to use groups properly, since the mad boss groups example in dialplan/default.xml does not work with the groups as defined in directory/default.xml. That "pointer" seems to break things.) In the meantime, I'll look more closely at that.
Thanks for any insight you might have on this.
Les
On Thu, Mar 29, 2012 at 8:03 PM, Michael Collins
<msc@freeswitch.org> wrote:
Grab a console debug log and throw it on pastebin.freeswitch.org. Use "FreeSWITCH Log" for syntax highlighting and then give us the pb URL in this thread.
-MC
On Thu, Mar 29, 2012 at 5:02 PM, Lesley Pervis
<lesley.pervis@gmail.com> wrote:
I'm using a small variation on the mad boss extension in the default dialplan, much like this, except instead of using groups, I'm using multiple user/100X@${domain} lines.
The problem is that the sip_exclude contact does not work as I expect. I expect that the conference is NOT bridged to the caller, but it is. I can see in the logs that a conference leg to the calling phone is set up, and the phone answers and puts it on hold. This means there are two conference legs up on the calling phone: the one that initiated the page, and another deaf leg. When the first leg hangs up, the deaf one is on hold, and to end the conference, I have to resume that leg and hang it up.
Anyone know why this isn't working for me?
FreeSWITCH version: 1.0.head (git-d8d4d20 2012-03-14 19-00-26 +0000)
Thanks,
Les
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org