Displaying 20 results from an estimated 10000 matches similar to: "No subject"
2009 Jul 20
0
No subject
=20
arp | grep "192.168.0.1"
=20
substituting the IP address of the SIP device.
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
=20
hello,
is
2007 Jul 12
0
No subject
to use the table asterisk.cdr but I can't find it anywhere.
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Content-Type: text/html;
charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
2009 Jul 20
0
No subject
=20
arp | grep "192.168.0.1"
=20
substituting the IP address of the SIP device.
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
=20
hello,
is
2010 Jul 03
0
No subject
eeds to be added to the database or coding of the program. Is this right? I=
sthere something else I need to do??
Ray Garza
System Analyst II
Materials Management
(956) 381-2163
--_000_60EAA3B47B099544A16557729563A1D608920F4FCCINFDBMBX4dsut_
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<html
2007 Jul 12
0
No subject
"We have created an easy and cost effective way to have customized
recordings done quickly and with no hassle."
I thought this was rather amusing, as:
1. If you want multiple prompts recorded, you need to submit a new order for
each, which means that even prompts of a couple of words are still charged
at $12. That is NOT cost effective. You could record all your prompts as a
single
2009 Jan 16
0
No subject
Telco, location, ect?)
At X times of day?
=20
Ect, ect.
=20
It sounds like bleed over, which can be causes by some many things the
best place to start is to find a pattern if there is one.
=20
James Shigley
Monroe Telephone Answering Service
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC
Sent: Tuesday, May
2007 Jul 12
0
No subject
What is the problem with SIP retransmits?
-----------------------------------------
Sometimes you get messages in the console like these:
- "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet."
- "retrans_pkt: Cancelling retransmit of OPTIONs"
The SIP protocol is based on requests and replies. Both sides send
requests and wait for replies.
2007 Jul 12
0
No subject
=20
Thanks!=20
=20
=20
Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
=20
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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2007 Jul 12
0
No subject
Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
=20
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charset="iso-8859-1"
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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2007 Jul 12
0
No subject
community there is a real possibility this may come off so if you have
an interest in this space and want to contribute to the discussion then
this is your opportunity to do so.
=20
I look forward to all opnions on this topic.
=20
The slide deck for the agenda of this call is located here
http://voipusersconference.org/2008-05-09-Slides=20
Cheers,
Dean=20
________________________________
2007 Jun 15
0
No subject
using Asterisk.
=20
Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?
=20
As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to
2007 Jul 12
0
No subject
event - which it rejects. The voicemail notifications ARE working on the
device.
Any way to get rid of this message (while keeping the MWI on the phone)?
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2009 Jan 16
0
No subject
"What is CentOS?
CentOS is an Enterprise Linux distribution based on the freely available
<ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat
Enterprise Linux. Each CentOS version is supported for 7 years (by means of
security updates). A new CentOS version is released every 2 years and each
CentOS version is regularly updated (every 6 months) to support newer
2011 Apr 12
0
No subject
a phone system, and plug it into a SIP Adapter like the PAP2T.
Never done it myself, so I can't recommend a suitable intercom. Hopefully s=
omeone else can.
Dan Journo
Kesher Communications (UK)
Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h=
ttp://www.keshercommunications.com/hostedpbx.html>
2007 Jun 15
0
No subject
using Asterisk.
=20
Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?
=20
As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
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2007 Jul 12
0
No subject
display, accelerometer/motion sensor being the first 4 for release
(though 81 have been mocked up so far).
The long term concept is if you want a 'weather station with live video
feeds and gps location control you can add various modules together to
deliver what you are looking to achieve.
I have high hopes for the concepts, and wish the guys well as it seems
their hearts are in the right
2009 Jul 20
0
No subject
-- SIP/ vaso -e26c answered Zap/14-1
-- Executing DumpChan("SIP/ vaso -e26c", "") in new stack
-- Executing DumpChan("SIP/vaso-e26c", "") in new stack
Dumping Info For Channel: SIP/vaso-e26c:
============================================================================
====
Info:
Name= SIP/vaso-e26c
Type=
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2009 Jul 20
0
No subject
might be your best bet to get the information you want. I'd look at
voip-info.org for information.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 16, 2009 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to list ongoing calls