Displaying 20 results from an estimated 10000 matches similar to: "No subject"
2009 Jan 16
0
No subject
Telco, location, ect?)
At X times of day?
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Ect, ect.
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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.
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James Shigley
Monroe Telephone Answering Service
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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
"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 Jul 20
0
No subject
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arp | grep "192.168.0.1"
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substituting the IP address of the SIP device.
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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.
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hello,
is
2007 Jul 12
0
No subject
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Thanks!=20
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Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
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2009 Jul 20
0
No subject
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arp | grep "192.168.0.1"
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substituting the IP address of the SIP device.
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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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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.
2009 Jul 20
0
No subject
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Error loading module
'chan_dahdi.so': libpri.so.1.4: cannot open shared object file: No such
file or directory
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Module
'chan_dahdi.so' could not be loaded.
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I am using on CentOS 5.4 64 bit.
Asterisk 1.6.0.25
Asterisk-addons
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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2011 Jan 10
0
No subject
Class: default
File: /var/lib/asterisk/moh//reno_project-system
File: /var/lib/asterisk/moh//macroform-robot_dity
File: /var/lib/asterisk/moh//manolo_camp-morning_coffee
File: /var/lib/asterisk/moh//macroform-cold_day
File: /var/lib/asterisk/moh//macroform-the_simplicity
Class: none
File: /var/lib/asterisk/moh/.nomusic_reserved/silence
2011 Sep 02
0
No subject
1. Does "Wrap-Up-Time" apply to all queue agents/extensions that just rang,=
or only the one who actually answered the call (I assume the latter)?
2. Does the "Member Delay" delay the ringing of new calls to agents, or onl=
y come into play AFTER the agent answers the ringing call?
Any other suggestions for how I can resolve this issue? I am wondering whet=
her "Agent
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 Sep 02
0
No subject
built-in; This doesn=92t matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web =
interface.
Let=92s say Yves=92 =93special conference=94 is 5555. The moderator =
would start
using this command
Exten =3D> s,1,meetme(5555)
The participants would do
Exten =3D>
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
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run
Apple doesn't accept (for the moment) an application runs in the background=
. So, when Siphon doesn't run, the SIP server of your provider doesn't know=
your iPhone."
--_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_
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2011 Jan 10
0
No subject
and Asterisk is plugging in pseudo ID. Is that correct?
It seems to me that Asterisk should simply say "no caller ID" or "No ID" or
something besides "Asterisk".
In any case, we are trying to filter them with little success.
When we do a LEN(CALLERID(num) we get "13", when we expect "10"
The call pattern is 1 call followed by a
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
major undertaking. But since you are using an AGI to control the Queue
command instead of using it from the dialplan, you have more control over
this problem than you realize. For simplicity of illustration, let's say
your AGI simply wants to take a call and send it to the next agent in the
queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the
Polycom transfer from
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
2011 Jan 10
0
No subject
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