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.
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
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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, 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 gshfre...@gmail.comwrote:
When a call is hangup, I get SERVER_DISCONNECTED Event over and over again
instead of CHANNEL_HANGUP event. Has anyone
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 gshfre...@gmail.com wrote:
What does linger do? I do not see it documented anywhere.
Why would I get
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.
, Shameem Shiek gshfre...@gmail.com 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
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. This is
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
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: uuid [-bleg|-both] dest-exten [dialplan] [context]
/b
On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:
Hello,
Good work
, Brian West br...@freeswitch.org 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 did on the asterisk land.
In asterisk land, Caller
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