On Fri, Dec 30, 2011 at 3:46 PM, Dave Redmore < dave.redm...@spigotnetworks.com
> wrote:
What, specifically, do you see as the shortcomings of Freeswitch, as
implemented in sipX, as an SBC?
I'm just curious.
- Original Message -
From: "Michael Picher" &
What, specifically, do you see as the shortcomings of Freeswitch, as
implemented in sipX, as an SBC?
I'm just curious.
- Original Message -
From: "Michael Picher"
To: "Discussion list for users of sipXecs software"
Sent: Friday, December 30, 2011 2:18:55 PM GMT -06:00 US/Canada
You really should avoid the term "SIP PRI" - it just doesn't make sense, in
terms how sipX is going to deal with it. From the user end (you) - it is a SIP
trunk, period. The concept of a SIP PRI is only relevant from the providers
end, in terms of how they are going to deliver and bill for the
Any chance you want to share who you have found/tried as an ITSP that supports
T.38?
I've been a long time customer of Teliax and have found them to provide a solid
service, but a bit on the pricey side. They will set up an account to handle
t.38 and have been using them to terminate t.38 fax
Hi Tony,
I might be able to help. I know I've talked about Teliax in the past. They have
a T.38 service via their wholesale service and they have set up a PAYG account
for me on that. I could temporarily have them add your IP to the ACL for the
account if it helps. I haven't tested sipX yet di
Changing the gateway (Freeswitch) to listen on port 5060 fixes the problem. So,
it seems that sipXbridge does not like to use a gateway that uses port 5090? Is
this a bug? Can anyone confirm this? Does anyone care?
Dave
- Original Message -
From: "Dave Redmore"
To: &
Message -
From: "Dave Redmore"
To: "Discussion list for users of sipXecs software"
Sent: Wednesday, October 6, 2010 10:58:15 AM GMT -06:00 US/Canada Central
Subject: Re: [sipx-users] 500 Internal Server Error on Invite
Call trace is attached.
Looks to me like sipXb
rewriting the headers, so I think the issue is going to be "port based". I
normally do this with a SBC, because they rewrite headers. I am not sure if FS
does that without a lot of work, they have an example on their wiki:
http://wiki.freeswitch.org/wiki/SBC_Setup
On Wed, Oct 6, 20
es that without a lot of work, they have an example on their wiki:
http://wiki.freeswitch.org/wiki/SBC_Setup
On Wed, Oct 6, 2010 at 9:32 AM, Dave Redmore < dave.redm...@spigotsystems.com >
wrote:
Anyone have any thoughts on this? I'm not sure where to start debugging.
Somet
My next step is to rework Freeswitch to listen
on 5060 and see if that helps - maybe then is a problem with specifying a
different port in the gateway?
Dave .
- Original Message -
From: "Dave Redmore"
To: "Discussion list for users of sipXecs software"
Sent: Tuesday, Oc
at the phone sends an Invite to sipX, sipX
returns "100 Trying" and then returns "500 Server Internal Error, with
sipfrag". The Message Body of the packet says "Sipfrag" and "Exception Info
Unexpected error creatin
rns "500 Server Internal Error, with
sipfrag". The Message Body of the packet says "Sipfrag" and "Exception Info
Unexpected error creating INVITE at Siputilities.java:839"
I have googled this error, but not finding any info.
Anyone have any ideas?
Sonicwall NSA 240. I have NAT policies which forward ports UDP
5080, UDP&TCP 5060-5061 & UDP 3-31000 untranslated to the sipX server
(we're a small shop so everything is running on one server). Are you saying
that the invite actually comes to UDP port 37678?
Stiles
Da
My settings for the gateway are all default - Under "Configuration", I defined
"Address" as "den.teliax.net" - Under "CallerID" I set the "Default Caller ID"
to my incoming phone number - under "ITSP Account" I defined "Username"
("Authentication Username" is left blank), "Password" and checked
I can report that I have 4.2.1 installed and working very nicely with Teliax. I
have configured a gateway using very "plain vanilla" settings and it worked
pretty much "right out of the box". Incoming calls and outgoing. MOH and
transfers all seem to work fine. I currently have a Grandstream GXP
need to do requires anything more and then I don't have
to deal with driver issues) in about 30 minutes.
Attaching the scripts, but not sure if they will make it through, so email me
directly if you need them.
Dave Redmore
Spigot Networks, Inc.
- Original Message -
From: m...@g
nsfers the call to an extension
at the office (signalling is between the Polycom and sipX/Tranferee Extension,
when call is answered, RTP is between ITSP and Transferee Extension).
Is this what the "goal" would be - if I have correctly configured my topology
and devices?
Thanks so m
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