Dear Freeswitch users,
I am building an app where the extensions map to external callers and there
are no registered users. For example, the extension 1001 would map to an
external number. In that case, does it make sense to use the Mod voicemail
or should I build a voicemail solution using dialpl
Dear Freeswitch Users,
I am looking for a SIP Provider who can provide a DID with unlimited
channels. Currently I am using junction networks but they have a high
2.9c/minute charge. I am looking for someone who has a flat rate for X
minutes.
Any advise would be much appreciated.
Thanks.
__
I found the linger command on a old freeswitch user's email thread. Updated
the Event socket outbound wiki with the command.
On Thu, Sep 17, 2009 at 3:08 PM, Shameem Shiek wrote:
> What does "linger" do? I do not see it documented anywhere.
>
> Why would I get a SERVER
at 2:56 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> try sending the linger command
>
> linger\n\n
>
>
>
>
> On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote:
>
>> When a call is hangup, I get SERVER_DISCONNECTED Event over and over aga
When a call is hangup, I get SERVER_DISCONNECTED Event over and over again
instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am
using freeswitch 1.04.
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Hi Michael,
Why is it not recommended to do the brdge app right in the script? The
reason I ask this, I did have lot of trouble using Park/Fifo app in the
script and the whole thing started working after I did the UUID transfer and
have the things I wanted executed as part of the Dial plan.
Also
This is one did the trick.
Thanks.
On Tue, Aug 25, 2009 at 11:07 PM, SP wrote:
> You can try playing with this in your gateway profile
>
>
> Not sure what it'll do to your registration, give it a try
>
> On Tue, Aug 25, 2009 at 21:51, Shameem Shiek wrote:
>
>
I am setting the caller id like this in my ESL script:
@con.sendRecv("api originate
{origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1...@number}
&park()")
And the caller id comes out as all zeroes. The sip trace shows the "from"
as shown in the sofia status command. Thi
ation?
On Tue, Aug 11, 2009 at 3:40 PM, Brian West wrote:
>
> On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:
>
> > Thanks for the explanation. I was trying to find the correct way to
> > do this as I saw several ways of doing this. I will try to explain
> > how I d
both B and C to park using the -
> both then hangup on C and then uuid_bridge A and C. Because once you
> break the bridge on B and C by hanging up on B you left C hanging so
> its naturally going to hangup.
>
> -USAGE: [-bleg|-both] [] []
>
> /b
>
> On Aug 11, 2009
Hello,
Good work guys. I am having good fun using freeswitch so far. Currently, I
am having a serious issue on making a call transfer happen. The scenario is
simple.
1. Caller A arrives on extension 1 and is waiting on a fifo queue.
2. Caller B arrives on extension 2 and dial plan bridges the ca
Thangappa,
Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.
On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice wrote:
> You need to look at using ODBC... That is what pretty much everything
> uses... Connect FS to ODBC and ODB
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