[Freeswitch-users] sip_exclude_contact in mad boss scenario does not work

Lesley Pervis lesley.pervis at gmail.com
Fri Mar 30 20:17:08 MSD 2012


Thanks Michael,

The paste is at: http://pastebin.freeswitch.org/18788

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 at 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 at 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 at 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.
>>
>> http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix
>>
>> 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
>>
>>
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