I have been having a problem with my outgoing caller ID coming through as
Private or Restricted using a PRI + Openzap. My provider claims that it
must be a configuration on my end. Is there something I might be missing?
Setup is essential the default FreeSWITCH configuration. I realize this is
Scratch that, I had my Openzap configured for national, not NI2.
Thanks,
-AF
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Tuesday, December 15, 2009 1:03 PM
To: freeswitch-users
I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I
have found with a stable release. It was very easy to use, I would
recommend it if you just want a quick base system with standard features.
Though, I ended up switching to a compiled version of FreeSWITCH in order to
I am still new to freeswitch, but I would think you could achieve this by
just passing the call to an IVR application that plays the message instead
of passing it to the voicemail application.
-AF
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Have you checked out Redfone? While I haven't attempted to implement it yet,
my Redfone foneBridge2 claims to be able to handle load balancing and
failover between two Asterisk/Freeswitch servers.
-AF
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
FusionPBX, FreePBX v3, and wikiPBX are the three that I have found in the
past. However they all seem to be in the early stages of development, and
not 100% stable. I can say this for sure about FreePBX and FusionPBX, but I
have not actually tried wikiPBX.
-AF
-Original Message-
From:
Awesome, thanks Andrew, I will have to keep an eye out for that patch.
To continue, last night I decided to tackle the business hours and holiday
routing on my FreeSWITCH system. It turned out to not be quite as simple
with the XML dialplan as I thought. After being up until 1am banging my head
Samuel,
FreeSWITCH has a Skype module that uses Skype client instances to connect to
the Skype network, you can read about it at
http://wiki.freeswitch.org/wiki/Skypiax
As far as an official Skype module for non-Asterisk PBX-es, it looks like it
is in beta right now -
Is there any way to set the origination_caller_id for a FIFO outbound call
to an on-hook agent? I can't find anything in the wiki about a FIFO or
member variable to set this. It seems to be set to 'Queue' by default, and
appears to be hardcoded in the module source. It would be nice to be able
Is there a standard module for FreeSWITCH out there that people use for
routing calls based on business hours and a holiday schedule? Or is everyone
just creating their own in the XML dialplan?(which seems pretty simple)
I can't seem to find anything on the wiki, but might just be searching
On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote:
Is there a standard module for FreeSWITCH out there that people use for
routing calls based on business hours and a holiday schedule? Or is
everyone
just creating their own in the XML dialplan?(which seems pretty simple)
I can't
Has anyone used a Polycom SoundPoint IP501 or similar hard phone with
FreeSWITCH? I configured one to register with my FreeSWITCH server using one
of the default sip profiles to test and I get [DEBUG] sofia_reg.c:1688 SIP
username 1001 does not match auth username in the log file and the phone
am using Polycoms (430 and 501) with FreeSwitch. How do you provision
them? Via WEB or config files?
If you use config files than I can send you some sample files.
Regards, __Yehavi:
On Nov 12, 2009, at 11:41 AM, Adam Ford wrote:
Has anyone used a Polycom
Hi everybody,
I have setup a FreeSWITCH IP-PBX for my office using a T1 and Redfone
foneBridge2, which uses Openzap, for my connection to the PSTN. I am trying
to figure out if it is possible to forward a call that comes in through the
T1/Openzap, back out to a PSTN number.
An example would be,
] On Behalf Of Russell
Mosemann
Sent: Wednesday, November 11, 2009 6:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Forwarding calls to an outside number -
OpenZAP
Adam Ford wrote:
I have also noted that I can simply bridge the call out another line on
the T1 through
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