Those boxes run around $50k USD, I've only seen them once back in the late
1990s.
At work for customer consulting we have very expensive site licenses for
Prognosis IPT Assessor which generate great looking pdf reports.
We also use Cisco IOS IP SLA however it doesn't have a reporting
I'm not aware of an open source speech product.
Some great examples where speech recognition works well are 1-800-USA-RAIL,
Microsoft/Cisco corporate directory 425-882-8080 you can say the employees name
and be connected and those works great, 1-800-Goog-411 also works well.
Windows 7
WireShark does a good job showing the T38 communication. Most products you can
also set packet redundancy to send 2 packets.
Your setup was T.38 ATA to T.38 Gateway with PRI/ANALOG/PSTN/G.711. I've heard
various problems with SIP/PSTN and faxing, around jitter/packet loss and it's
not
I normally work with other 3rd party IVRs, usually once the Agent is Reserved
we signal the phone system to play Music on Hold while it's ringing the Agent.
The trick here is to replace the Music on Hold with a fake ring file.
-Original Message-
From:
I think you need to ask your SIP provider about Redirecting Header, ask what
they support and how-to.
I work more with Cisco CallManager and SIP Rediversion Header is new in
CallManager 8x. Not sure about Asterisk. We have this same problem with Cisco
Mobility/Single Number Reach, providers
I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323.
Trying to find a way I could use my laptop to send a fax over H323 to the
BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface
on a Cisco router and setup a h323 dial peer to the BrookTrout, but I
I'm experiencing runaway ringing too, can we make this a class action
against someone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
Brower
Sent: Wednesday, March 10, 2010 10:20 PM
To: Chris Owen
Cc:
Anyone have a customer sending/receiving multi-page faxes over Verizon
Business SIP trunk/g711 ?
Verizon Business indicates they don't support it, and I have 2 recent
customers that it doesn't work for, and 1 current large customer telling
me he's going to make it work grin.
The issues is
BellSouth (now ATT) has a number you can dial and it will play back
voice prompts with your calling number? It's used by their techs with a
buttset in identifying analog 1FB lines...
Eg Dial 704-210-3233, it answers
Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two if your dialing from
To clarify the question is what is the number for ATT Calling Number
Verification?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason
Aarons (US)
Sent: Monday, June 29, 2009 7:24 PM
To: Asterisk Users Mailing List - Non-Commercial
Is this Project Eagle Eye ? Call every phone at once to tell them about
H1N1 in their neighborhood
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel
Business
Sent: Monday, June 22, 2009 3:46 PM
To: 'Asterisk Users
What is to stop anyone from dialing international at any time,
regardless if he bridges someone else on?
Usually we implement Force Authorization Codes (When dialing out after
dialing you have to enter a code) to track all Long
Distance/International calls. You can then generate bill back reports
No divx hd? just kidding
OT: Odd how many video/audio standards there are, and the growing issue with
them? I recall when you had two choice Windows Media or RealPlayer. Now I have
to make 3-4 for everything from DivX to iPod to Walkman. For example my cell
phone can't play a H264/AAC due the
- Non-Commercial Discussion
Subject: Re: [asterisk-users] AmooCon video recordings online
Jason Aarons (US) schrieb:
OT: Odd how many video/audio standards there are, and the growing issue with
them? I recall when you had two choice Windows Media or RealPlayer.
There is only one format[1
Are conference bridges and other resources going to work with SRTP ?
I'm wondering what enabling SRTP will break in Asterisk. It breaks
several things in Cisco CallManager. Also wondering what make/model SIP
phone you are using for SRTP and what experience other having using that
make/model for
Customer has a Verizon Business SIP trunk, I'm still used to PRI T1
myself for local service. The fax machines are having some issues (I
can use analog phone to call out fine) and I'm checking on modem
passthrough with Verizon, but wonder if any else is using Verizon
Business for SIP trunk and
Is this inbound calls to your automated attendant? Or Outbound calls to
say a bank ivr out in the pstn? What direction?
What is your interface/carrier? T1, SIP, H32? And what method are you
using for DTMF? Eg inband, out of band, what rfc, etc?
From: asterisk-users-boun...@lists.digium.com
Then if it's a IP interface (SIP, etc) have you tried a sniffer trace
(wireshark, etc) to verify the packets are being sent correctly to
carrier?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason
Aarons (US)
Sent: Friday, May 22
Can anyone recommend a codec pack with G723 for use under Vista? I have
G723 prompts (about 70 prompts totaling 1MB) needing to be converted to
G711 uLaw.
I tried Audacity but it doesn't have G723 codecs. I tired some google
found adware free tools and websites with no success in
If you set the system clock ahead does problem follow the clock?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Tuesday, May 19, 2009 6:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hang at 5:34
I wonder if using the Internet Low Bit Rate Codec or iLBC would work
better. G711/G279/GSM all suffer when too many packets are lost. You
would then need to transcode to G711, etc -jason
http://en.wikipedia.org/wiki/Ilbc
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I don't think a off the shelf modem has the necessary DSPs to convert
voice to codecthat is why a Voice Gateway/Analog Telephony Adapter
or FXO/FXS cards exist instead of modem having a second life.
I do recall a few that worked as a answering machine allowing your home
computer to answer
Nextel does that, pickups up after x rings and says 'The Nextel subscriber you
are trying to reach is unavailable, please try your call again later.
I'm not sure what Verizon or Nextel called this feature or what advantage is
it for the carrier to play it versus just letting it ring forever...
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn
Sent: Monday, March 16, 2009 8:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with Verizon Wireless
On Mon, Mar 16, 2009 at 5:43 PM, Jason Aarons
Any idea what legal statues setting caller-id fraudulently falls under?
Is there a federal law you can reference?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber
Sent: Wednesday, February 25, 2009 4:13
After helping out it seems I've been determined a female(wrongly). It
was disappointing and I'm considering a visit to the Dr Phil Show to
work out my anger
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Asterisk
Sent:
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650
;p=1
I can't see the Dept Transportation running copper to all the motorist
aid boxes along the highway. I thought most of your alarm panels have
moved to GSM/CDMA backup communications. I'd like to see a fire
marshall
.
j
On Tue, 17 Feb 2009, Jason Aarons (US) wrote:
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650
;p=1
I can't see
The Intel 80386 used 32-bit architecture in 1987...might want to specify
make/modelI'm not sure you want to run * on a old Tandy or Packard
Bell -jason
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire
Sent: Thursday,
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really
is a great sounding phone. I have several customers with them as SCCP.
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/p
s8759/product_data_sheet0900aecd806e021a.html
From:
Google Multitech CallFinder GSM out of Minnesota if you want a common
off the shelf product. GSMA.org was using their product with FXO/FXS
for backup purposes.
I recall they have a GSM to FXO/FXS, and I thought they had GSM to H323.
I also found a European company that made high end (24+)
Microsoft doesn't make a native SIP client in Windows Mobile you can use
for a phone call. Do you mean Windows Live Messenger?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG
Technical Support
Sent: Wednesday, December 03, 2008 9:15 PM
To:
Just switching from Nortel to something else may not eliminate
hardware/software failures, or prevent those without experience from
pushing the enter key at the wrong time. You have to consider the two
professionals actually cost considerably more than just salary, due to
taxes, 401k, benefits
I would stick with 1.4 in production, how mad would you be if I gave you a cell
phone with new code and it didn’t work? Would you throw your cell phone at me
if it cut us off during phone calls from a bug? Some people are ok with trying
new stuff, others it costs money when they lose business
A lot of places you still can't get GSM in the US.it has
improved...but GSM 3G coverage is lacking compared to EVDO/CDMA.
Another option is a World Phone that can do all bands.
My story;
Visiting American lands in Kuala Lumpur and checks phone for messages...
Puzzled look appears as it
Jitter is what your describing, it's a bad thing.
http://en.wikipedia.org/wiki/Jitter
While VoIP may work (third party 128ms echo cancellers, etc) most
support organization won't go outside ITU-T G.114 recommendations.
I've done Cisco 7940 phones deployed in the Gulf of Mexico on a oil
platform
I haven't used Asterisk Voicemail but here are Unity Unified Messaging (for
Exchange) 5.x/7.x features, in short I think you need to be a
Callmanager/Exchange Server shop with heavy integration with ActiveSync/Direct
Push/Outlook 2007/OCS2007. The company that created Unity (Active Voice) was a
Was it like watching a 106' plasma at 1080p for the first time? grin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, July 17, 2008 7:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
My understanding is Skype's secret is using the iLBC codec, which Cisco
has also licensed for their 79X2 models as well. I travel and lot and
in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator
will fail the Skype client will work. The iLBC codec can really handle
packet loss.
Digital ISDN used Q931 messages. You should get a disconnect message
from telco on the d-channel 23.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Hazelbaker
Sent: Monday, July 07, 2008 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Google multitech CallFinder 100, has both FXO and FXS interface you can
connect.
Problem is you call out and don't get simulated ring back while the GSM
call is being setup.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Saturday,
While not on the FXS port itself other things to look for;
1) set the fax machine to disable Super G3 (33.3k), try to force it to
use G3 (14.4k).
2) set the fax machine to Disable ECM (Error Correction Mode)
3) If PSTN is T1 check span for errors, etc.
4) Some protocols have fax-passthrough
Will this work backwards? When I'm at home instead of my cell ringing
have the home phone ring? Why would anyone give up revenue from minutes?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Tuesday, October 09, 2007 12:03 PM
To: [EMAIL
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN BRIs for voice or data since early 90s.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent:
Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia?
What signaling did they provide, framing, formatting?
primary-4essLucent 4ESS switch type for the U.S.
primary-5essLucent 5ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type
Glad to hear you had a workaround.
I would suggest re-queing your TAC case, perhaps you got a outsourced or less
experienced engineer at Cisco. Their support has varied depending on which
city/group you get. Some have more experience then others.
While your 2600 from 2001 timeframe it should
I have a telco providing DTMF inband, they say they can't provide it any
other way. This is creating headaches for me.
What is the common method for SIP DTMF? Kpml, or 2833 or inband?
My handsets don't support inband so I'm tying up some expensive
resources to convert the inband DTMF to
I thought XO was reselling Level 3s (old Genuity assets) network/voip
just like Qwest ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 3:38 PM
To: asterisk-users@lists.digium.com
Subject:
For DND press Call Forward All (CFwdAll softkey)
then Messages button on the SCCP version. I havent seen the SIP version
of 7961G.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
Sent: Wednesday, August 30, 2006
Is there a FCC or other North America
requirement that I provide 911 versus 9.911. I want to require users to dial
9.911 in our office, and remove 911. Are there any statutory requirements or
laws about this? User accidentially dial 9 then 1 then another 1 and hangup.
Weve
XMedius is a great T.38 fax product, integrate with LDAP/AD/Exchange.
Integrates with the PRI card in our Cisco Routers using H.323.
-
Disclaimer:
This e-mail communication and any attachments may contain
confidential and privileged information and is
Is anyone familiar with the Telco in Prague?
We have an issue with the connection that will be made from the Telco
demark when we do an IPT installation next week.
-jason
-
Disclaimer:
This e-mail communication and any attachments may contain
52 matches
Mail list logo
