Displaying 20 results from an estimated 22 matches for "interoffice".
2010 Apr 06
1
IAX Problem
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [s at macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
"IAX2/InterOffice/210,300,tr") in new stack
-- Called InterOffice/210
-- Hungup 'IAX2/InterOffice-7578'
== Everyone is busy/congested at this time (1:0/0/1)
The only difference I am aware of is that one server has a public IP
address, the other is behind a NAT.
The trunk from the server...
2005 Sep 20
6
iax2 trunking wackyness
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
The setup is IAX2 trunking using GSM codec.
Is there any obvious reason I am overlooking to figure out why
there is such a big difference between the two.?
I am using CVS-head September 3rd, maybe there is a version
skew?
Any suggestions will be appreciated.
Thanks
Clive
2006 Jun 28
2
point to point T hookup?
...n where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead of over the voiceT at each location to maximize savings on
interoffice calls.
An alternative is to put a router and switch at each end and extend a data
network to the other site for SIP traffic. Would that result in better
quality calls?
What configuration areas are there to be set and how are they diffent than
just a standard PRI, which I have working now?
Th...
2007 May 01
3
Display Caller ID of called party
Not sure if this can be done or not, but I can't seem to find it
anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I
would like to have the caller id of the person I am dialing displayed
and not the number I just dialed. Is this possible? So, if extension
4023 is John Doe, and I dial 4023, my display should read John Doe and
not 4023. I am using a Polycom 501 by the way in case that matters...
2005 May 04
4
OpenSwan traffic shaping with HTB & sfq
Hi All,
I''ve got an interoffice IPSEC VPN in place that I''m trying to give
priority to terminal service (tcp 3389) traffic.
I''ve created rules at each end, but have hit a bit of a dillemma. As
the data is encrypted I must also give highest priority to protocol 50
otherwise the priority is lost as the packet get...
2003 Oct 10
1
Marketing Digium/Asterisk
...mpanies who need advanced features such as ip trunking, IVR,
Conference bridging, etc., etc.
I would like to find a way to identify multi-location companies who would
benefit from IP trunking - perhaps by industry segment. ie what type of
company is likely to have multiiple locations with lots of interoffice
communications.
Also, I would love to target Centrex customers. I put in an Asterisk
system for a local company that had been on Centrex. In short, it will save
them $18,000.00 per year (and did pretty well for myself too:)). Any way to
ID centrex customers?
I'd be glad to share any tips...
2006 Nov 08
1
I LOVE IT
After about one weeks time I've gone from no VoIP to a completely
configured system for two of our offices to be able to page/communicate
interoffice as well as handle existing PSTN communications (okay,
waiting onf hardware for the PSTN side and I've likely jinxed myself
now).
I was sweating getting the two boxes talking to each other and I knocked
that out in no time without even needing to look up online, FreePBX
makes it to easy.
Onc...
2005 Jul 11
1
Snom 360 NOTIFY syntax
...nstallation with snom 360s in the near future.
Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a
snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I
have the 360's set up to subscribe and notify for the line use lights,
which works like a charm for interoffice calling (between the 360's,
anyway. The IAXy, 200 and, softphone will be used by less phone
dependant types) but what I can't figure out from the Wiki is if it's
possible to have the ZAP lines notify for the outbound lines so we can
see how many lines are in use.
My configuratio...
2008 Aug 11
1
Phone system layout suggestions
...any's phone "layout" and would like
to get comments from people who have done something similar.
Currently, we have 3 locations - each with their own Asterisk PBX. The
corporate office has a PRI. Each remote location has a SIP provider for
5 channels of SIP going to their own PBX. Interoffice calls use the PSTN.
Most inbound calls come to the headquarters via a toll free number.
We are adding another remote office and are planning a MPLS network to
connect all locations for DATA. As we really want a "centralized"
switchboard
anyway, I thought with QoS and an MPLS, I could el...
2008 May 28
3
Asterisk VoIP in Dubai/UAE?
Dear All,
We have a customer who is opening a new office in Dubai and we know
that VoIP is blocked over there.
Has anyone a solution to getting VoIP back out (we want interoffice
calls back to the UK)? We we're thinking of IAX trunking, but not sure
if that is blocked or just SIP etc.
A VPN works, but is not great. We have seen:
http://www.speed-voip.com/voiceguard.html
At the moment it is a very small satellite office with one user, but
will be bigger soon. Ideally,...
2003 Nov 30
1
Rate file formats: a standard?
...andardization as to how I would populate an LCR
system on anything even remotely approaching "automatically". I get
Excel spreadsheets via email, text files off the web, badly formatted
RTF, faxes, snail mail, everything except for cuneiform - though I
expect my first clay tablets in interoffice mail any day now. As an
IP-centric person and someone who has been exposed to using data
networks as a transport mechanism, it seems odd to me that an
RFC-like document describing rate table formats has not come across
my searches to this point. So...
====
The question is: Is there a standar...
2005 Jun 27
0
Failover Design
Hello All,
I've been investigating and playing with asterisk to see how it would
work out as a small-medium business pbx to handle mostly
interoffice/branch communication and a possibly communication out to
pstn in later stages of implementation. (All communication would be VoIP
internally with possibly 4 pstn lines at the HQ with either a TDM04B or
spa3000s and 1-2 pstn lines across 7-8 branches with spa3000s).
Still evaluating my options abou...
2006 Apr 12
1
free video (soft) phone available?
I am using eyebeam and I am happy with it. However, it is boring just to
talk to my son in the other room.
Whenever I try to convince somebody to buy eyebeam, they are scared of
the price.
Is there a free video soft phone available, that will work with eyebeam
/ asterisk?
bye
Ronald Wiplinger
2006 Nov 22
2
G722?
In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products
like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions.
Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be
2005 Jun 06
1
RE: LOA for CFA . . work up "pencil copy"
...numerous occasions yet it doesn't seem to be moving anywhere. In the
conference call last week, my understanding was that it just needed to be
signed by an authorized SC Telcom employee like Steve and then sent to us.
Please take care of this ASAP as we need to get the ball rolling on ordering
interoffice transport circuits. Not only do we need these for Indy,
Coffeyville, and our ATM DS3 from ASI, but we also need them for Augusta and
Andover to order DS3s back to Broadway for the collocation equipment moves.
Regards,
Shaun
-----Original Message-----
From: David Henning [mailto:dhenning@sctelco...
2004 Jul 05
4
IAX Call Pickup
I've looked in the obvious places but haven't found a definitive
answer to the following: can an IAX extension (an Iaxy phone, for
instance) do call pickup via *8?
Adolfo
2009 Jun 27
0
Audio distorted local side only
...und on the
call. We hear it very loud, almost painfully and you have to hold the
handset away from your ear as you talk. The remote party says they
hear nothing at all wrong. It happens on both handset and speaker
phone, so it shouldn't be a speaker issue.
This never seems to happen on interoffice calls, only on calls going
out over our POTS lines. Also, if we put the call on hold or transfer
the call, usually the distortion is gone once the call is picked back
up (although it may come back, but not as bad). Flip side, sometimes a
good call that gets put on hold or transfered will ha...
2003 Dec 22
1
Authentication
Dear all,
I have a question regarding the configuration of *. I have 3 port FXS, and 2
port FXO. I have 4 users that use analog phone connected to FXS (I have 3
phones). I need to limit the user's capability (user A can call
International, user B can call long distance, etc). I want to implement the
password say to call , he/she needs to puch 9(for outgoing call)2-4 digits
password,then
2006 Jun 30
0
FOSS, Science, and Public activism
...kindness. Obviously, feel free to write back, copy, or
send these comments along to anyone else as you see fit.
Regards,
Michael L. Love Ph.D
Department of Biophysics and Biophysical Chemistry
School of Medicine
Johns Hopkins University
725 N. Wolfe Street
Room 608B WBSB
Baltimore MD 21205-2185
Interoffice Mail: 608B WBSB, SoM
office: 410-614-2267
lab: 410-614-3179
fax: 410-502-6910
cell: 443-824-3451
http://www.gnu-darwin.org/
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2006 Jul 01
0
FOSS, Science, and Public activism
...kindness. Obviously, feel free to write back, copy, or
send these comments along to anyone else as you see fit.
Regards,
Michael L. Love Ph.D
Department of Biophysics and Biophysical Chemistry
School of Medicine
Johns Hopkins University
725 N. Wolfe Street
Room 608B WBSB
Baltimore MD 21205-2185
Interoffice Mail: 608B WBSB, SoM
office: 410-614-2267
lab: 410-614-3179
fax: 410-502-6910
cell: 443-824-3451
http://www.gnu-darwin.org/
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