Hi, all currently we have a requirement from our customer. They want to
capture the wrap up time for agent, they want the agents status become
wrap up state automatically after a successful call, and only change
back to available state when the agent send some indication to pabx
server( via
What is the maximum number of calls can a trunked iax2 line take ?
siqhamO
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Yes. I don't use my customer's names on the list, so I can't say anything.
Porier, Jeremy M. wrote:
They're not the only ones :-)
Jeremy Porier
Senior Director of Information Systems and Technology
Colorado Christian University
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL
Hi,
we were using asterisk 1.09 with bristuff 0.20 for quite a long time now.
With the release of Asterisk 1.2 we tried to update our servers.
Now, with Asterisk 1.2.5, patched with Bristuff 0.3.0-PRE-1k on a SuSE
10.1 (we use the official SuSe 10.1 RPM of asterisk) we get some strange
behaviour
I can take 30 calls in one trunk with good voice quality
more calls cause awesome sounds___
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In article [EMAIL PROTECTED],
Avi Miller [EMAIL PROTECTED] wrote:
Brent Franks wrote:
We ran into the same thing, and the only way I can get it to work
(which is goofy, but it does work) is modprobing the same device
multiple times.
Try waiting after modprobe zaptel for udev to create
Hi users;
i am new to use asterisk so i am facing some problems.
i have installed asterisk 1.2.4 and all the
requirements.
i have to implement the Method 3 of the link.
http://www.voip-info.org/wiki/view/Asterisk+at+large.
i am doing like that;
XLITE---OPENSERASTERISK
On 9/13/06, Forum [EMAIL PROTECTED] wrote:
Unfortunately they pointed me back to Polycom and I have not yet heard back
from them.
Can somebody post a link to download sip2.0.1?
If they point you back, report that to Polycom, they'll contact the
reseller (if it's an authorized reseller, that
Patricio Valarezo a écrit :
Hi, it's possible to implement a callback without agi?, i'm trying
this but * exits without dialing (if I hungup during s,3 wait) but if
it hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do it? just for learning
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Xue Liangliang wrote:
Hi, all currently we have a requirement from our customer. They want to
capture the wrap up time for agent, they want the agents status become
wrap up state automatically after a successful call, and only change
back to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Do you have queues/agents configured?
No, I don't have queues nor agents configured.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail:
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that
SER -- Asterisk A -- Asterisk B (chan_ss7) -- PSTN
What happens is, that after a while (uptime was a least two days) the
gateway starts to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You have to leave a message in the voicemail, then listen it and the error
will not apear again.
That's bad procedure. Because, all of my clients receive voicemails on e-mail
with delete option. So, they will newer listen voicemail and
Hi users;
i am new to use asterisk so facing some problems.
i have installed asterisk 1.2.4 and all the
requirements.
i have to implement Message waiting indicator (MWI) by
using METHOD 3 of the link:
http://www.voip-info.org/wiki/view/Asterisk+at+large
i am doing like that:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You probably already figured this out, but you use either Agent or SIP,
not both. Use Agent if they login through AgentLogin or SIP if it is
calling the SIP phone directly.
Yes, I have figured it out later.
It should go like this:
Hi,
I had exactly the same problem with trixbox and asterisk 1.2.11. That
looks to be a bug in this release of asterisk. When using one channel
with one license, I had the message Out of G.729 Encoder Licenses! (in
/var/log/asterisk/full).
And once the call finished, the command 'show g729'
Here is what I've answered to a similar thread, I guess that's the same
problem :
Hi,
I had exactly the same problem with trixbox and asterisk 1.2.11. That
looks to be a bug in this release of asterisk. When using one channel
with one license, I had the message Out of G.729 Encoder Licenses!
Douglas Garstang wrote:
Has anyone ever gotten the Polycom MyStat soft-key to do anything?
Setting the status to something like 'Away', does not generate any outgoing
SIP traffic from the phone. Calling into the phone either from a watched
buddy, or other number, acts as if the status was
Hi Kai,
we had a similar problem with a PBX which had PSTN lines and SIP phones:
sometimes some phones had one way calls...the caller couldn't hear. We
hadn't tried to restart but we reduced the number of RTP ports (rtp.conf
if memory helps!) to a range of 200 (it depends from the number of
On 9/14/06, Steve Underwood [EMAIL PROTECTED] wrote:
Steve Davies wrote:
[snip]
This looks pretty good I have to say - The ECM seems as if it may be a
little intolerant... On a fax machine where I got 100% success in the
past with 0.0.2, I am now getting result (60) Disconnected after
I want to change that ASTCC will ask for pin.
1. Where to set it? Pin length and number?
2. Can I set the pin only for a few people? E.g. Would deleting the
pin number not ask for the pin or needs than still the #
3. How to change the pin? Can the user change the pin?
bye
Kevin Kiely wrote:
I am considering a Polycom expansion module for the IP601 for a DSS/BLF
application. I had read that there was a limitation as to the number of
lines that could be monitored with the ‘hint’ command.
Can anyone tell me if they are using this with multiple lines, I need to
Hi users;
i am new to use asterisk so facing some problems.
i have installed asterisk 1.2.4 and all the
requirements.
i have to implement Message waiting indicator (MWI) by
using METHOD 3 of the link:
http://www.voip-info.org/wiki/view/Asterisk+at+large
i am doing like that:
Hi list,
I've got following Problem:
i have severel phones on my asterisk. and externel lines connected (POTS
sip, does not matter)
a externel caller A (CID(num)=0815) calls me ( 4711) .
4711 can be distributet to severel internal extensions for example 23
and 42.
23 is on ZAP/1 and 42
Is there a document somewhere giving the correct TDM400P FXO settings for
use on a BT PSTN line in the UK? All I can find is
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
but it doesn't give the complete settings, such as loopstart vs kewlstart.
Anyway, the problem I'm having is that an
Hi,
I had been struggling with this, and I thought I will post the solution.
I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and
I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips
PABX. The
Hi all,
Does Thomson 2030 hardphone has the feature of supporting more than one
user registered at the same time? I heard not... But I think that's
weird because it has 4 profiles...
Thanks,
Ricardo.
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Hi all,I was running my asterisk from one year. today i upgrade with 1.2.12.1 once the caller is dialing the destination number caller can hear well real RBT from telecom and once called party pickup the phone the call became silence no voice both side.Please try to help me ASAP 150 calls waiting
On Wed, 2006-09-13 at 21:27 -0700, Crazy Boy wrote:
Hi,
I am using Trixbox on CentOS. I bought BT speedway ISDN PCI Card.
But, I dont know how to configure this card with Trixbox. I searched a
lot in Internet and forums. But, I didn't get any tutorial or any
response. You are using this
On Thu, 2006-09-14 at 11:05 +0200, Benoît Mérouze wrote:
Hi,
I had exactly the same problem with trixbox and asterisk 1.2.11. That
looks to be a bug in this release of asterisk. When using one channel
with one license, I had the message Out of G.729 Encoder Licenses! (in
I'm getting a strange situation with the first digit being doubled on
outbound dialling, and other oddities. I think something strange is going on
in my dialplan, rather than a DTMF decoding issue, but see what you think.
The platform is CentOS 4.4 plus Asterisk SVN trunk as of yesterday, and a
On Thu, 2006-09-14 at 04:21 -0700, Abdul wrote:
Hi all,
I was running my asterisk from one year. today i upgrade with 1.2.12.1
once the caller is dialing the destination number caller can hear
well real RBT from telecom and once called party pickup the phone the
call became silence no
Hi Abdul,More information about your setup would be helpfultoresolveyourproblem. RBT could be a locally generated tone at the endpoint as a result of SIP provisonal messages. Are you sure that this RBT is played from the remotes Telecom
Interesting problem - small server with a TDM400P card, one FXO and one
FXS module.
The dialplan (amongst other things) routes an incoming call from the PSTN
no the FXO port to to the FXS port, and a number of SIP phones.
There is a fairly standard DECT phone with handsets round the office
I have 2 asterisk servers. I am trying to connect them with SIP and
getting an error.
My first box I define sip.conf as:
[devcentos64_to_bt610tMM]
type=friend
username=devcentos64_to_bt610tMM
secret=password
disallow=all
allow=ulaw
allow=alaw
allow=gsm
host=192.168.1.159
context=default
my
Hello ppl,
Am trying to build a system, wherein users can access their profiles,
and hence voicemails thru a browser.
I am using Apache and am running it on another box and asterisk on
another. Am keeping them seperate to not have http traffic on the same
box as asterisk.
Now, my qs:
Is
I am a new member and I got this error message:
rmv aiu stoped data base error
tks
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, September 14, 2006 9:13 AM
Subject: [asterisk-users] asterisk server to server using sip
Luiz Miguel wrote:
I am a new member and I got this error message:
rmv aiu stoped data base error
this sounds like a rather big problem
tks
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Luiz Miguel wrote:
I am a new member and I got this error message:
rmv aiu stoped data base error
google for it... it found a few interesting references, among which:
http://www.textfiles.com/magazines/TOT/tot-o6.txt
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Yes . I know.
Do you have the ODBE forms related page 10.16 and 10.17. These pages are
about AIU ( EUAIU and APAIU)
Tks
- Original Message -
From: Raphaël Jacquot [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
have a look at Wiki for asterisk + odbc storage. The database for
storing entire voicemail messages can be stored on a local or a remote
database. Then you can do whatever you want with it. You will have to
recompile asterisk by turning on ODBC storage. It's all there on the
Wiki
After launching the FFasterisk project (video file converter for Asterisk),
i6net is launching VXIasterisk : the VoiceXML browser for Asterisk.
The VoiceXML browser supports Voice and Video with Asterisk.
VXIasterisk allows to create easily powerful VoiceXML portals for Asterisk.
For more
Hi,
does anybody currently use voipstunt from finarea? I place a call to
sip.voipstunt.com I get a 302 redirection. Unfortunately the second
server seems to support only a different set of codecs than the first:
-- Called [EMAIL PROTECTED]
-- Got SIP response 302 Moved temporarely
Brian Candler wrote:
I'm getting a strange situation with the first digit being doubled on
outbound dialling, and other oddities. I think something strange is going on
in my dialplan, rather than a DTMF decoding issue, but see what you think.
The platform is CentOS 4.4 plus Asterisk SVN trunk
Jean-Michel Hiver wrote:
Patricio Valarezo a écrit :
Hi, it's possible to implement a callback without agi?, i'm trying
this but * exits without dialing (if I hungup during s,3 wait) but if
it hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do
Why oh why do so many people do all this modprobe stuff manually or in
rc.local etc.?
If you are running a RedHat / Fedora / CentOS distribution, just do
make config in the zaptel directory, and it will create a proper
startup script in init.d and set up the rc.d links for invocation at
boot
Hi,
I have the following setup:
[ Voip Provider ] -- (XX) -
x.x.x.x (real world phone number)
|
{ The Internet }
|
200.x.x.x (Internet IP)
[linux router]
I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it.
Sep 12 18:52:36
WARNING
[4620]: chan_sip.c:1835 retrans_pkt: Maximum retries exceeded
on transmission [EMAIL PROTECTED] for seqno 1620
(Critical Response)
Sep 12 18:52:36
Im updating from 1.2.9.1 to 1.2.11 and im having a issue with attendad
transfer via SPA 941 that i did not have with 1.2.9.1. I get this
message on Cli log.
Sep 14 16:09:52 NOTICE[5780]: chan_sip.c:6897 get_refer_info: Supervised
transfer requested, but unable to find callid
'[EMAIL
We
are trying to set up a script that will test hundreds of toll free numbers to
ensure that they correctly terminate at our Nortel PBX. We have the perl
scripting written to dial the numbers and it works like a charm except for one
problem. We are not sure how to detect whether a
Hello,
In the process of finding my way around, I tried to get Asterisk to give a
recorded message if an invalid extension is dialled by a locally-attached
phone (FXS port on TDM400P)
Here's what I'm trying:
-- extensions.conf --
[internal]
exten = 611,1,Answer()
exten
Hi Raul,
Try canreinvite=no in your sip.conf file. Then all calls will go via
asterisk.
Marcus
Raul Dias skrev:
Hi,
I have the following setup:
[ Voip Provider ] -- (XX) -
x.x.x.x (real world phone number)
|
Hello guys,
Is there anyone who could explain me some stuff about sip show peers ?
108/10810.1.1.40 5060 OK (1 ms)
107/10710.1.1.246 D 51074OK (101 ms)
The port seems different here, and the main difference is
On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote:
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1}) NOTE HERE
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
Try replacing the first step above with:
exten = _9.,1,Dial(Zap/4/w${EXTEN:1})
Note the w in
Yeah there are some problems with the docs, and the product itself isn't very
impressive -- still bugs that existed for months that basically make it
worthless for me to use.
Anyway, since they didn't include ircd and the perl mods in the new package,
just download and install ircd-hybrid
Hey everybody,
I've been struggling with this for a while and figured I'd ask. I have
a facility that is connected via a PRI to a Definity G3R, it does not
pass caller id number, only name. I see the following when this call is
passed via IAX to one of our facilities:
-- Accepting call
2 asterisk boxes connected via SIP trunks.
Is there any way to subscribe to BLF on the other side? Using GXP2000 - not
wanting to have an account for each box if possible.
Thanks,
Sam
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Evnin...
Googled around for this strange error meesage with no
helpful results at all...
Does somebody has any idea what this means?
Forcing Marker bit, because SSRC has changed
At the same time I only get inbound audio but other
side can't hear me...sometimes I just hear my echo
and
I over the last 2 weeks have had voicemail start to disappear from the
system. The symptoms work something like this: The user logs into
Comedian Mail, and checks how many messages they have. They then opt to
call back in later to check the messages, but when they do so the
messages are gone,
That is not helpful in convincing my customers that there are many
companies using Asterisk.
Michael Welter wrote:
Yes. I don't use my customer's names on the list, so I can't say anything.
Porier, Jeremy M. wrote:
They're not the only ones :-)
Jeremy Porier
Senior Director of Information
Title: WAIT FOR DIGIT not working
Hello all,
I have been trying to solve this problem for days, with no luck.
When I run an AGI script from my extensions.conf, it seems no matter what I do, the WAIT FOR DIGIT command will not work. The system just flies past it without waiting a single
exten = _X.,1,Playback(pbx-invalid)
exten = _X.,2,Goto(s,1)
Brian Candler wrote:
Hello,
In the process of finding my way around, I tried to get Asterisk to give a
recorded message if an invalid extension is dialled by a locally-attached
phone (FXS port on TDM400P)
Here's what I'm trying:
Response below
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Rousse
Sent: Thursday, September 14, 2006 10:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip show peers
Hello guys,
Is there anyone who
Brian Candler wrote:
On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote:
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1}) NOTE HERE
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
Try replacing the first step above with:
exten = _9.,1,Dial(Zap/4/w${EXTEN:1})
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Thursday, September 14, 2006 8:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk server to server using sip question
I have 2 asterisk
-Original Message-
From: Norris, Sam [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 14, 2006 8:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BLF across asterisk trunks
2 asterisk boxes connected via SIP trunks.
Is there any way to subscribe to BLF on
I have a problem :
when i receive a
call in h323 and send on zap channell, there is no cdr..
if i receive in sip
is all ok .
Why
??
Thanks
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, September 14, 2006 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Controlling the channel
Hey everybody,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian Candler
Sent: Thursday, September 14, 2006 10:44 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] 9 becomes 99 ?
http://en.wikipedia.org/wiki/Top_post
On 9/13/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
James Jones wrote:
but in the register program it is staticly linked.
Maybe because you are top posting you didn't see my reply:
Best place to ask
Is there any documentation, maybe at voip-info.org, available for Asterisk 1.4?
Either in the form of _new_ docs, or docs that outline the differences and new
features that will be available in 1.4?
I'd like to avoid the months of trial-and-error that I went throo with 1.0, 1.2
if I can...
Andrew Kirch wrote:
I over the last 2 weeks have had voicemail start to disappear from the
system. The symptoms work something like this: The user logs into
Comedian Mail, and checks how many messages they have. They then opt to
call back in later to check the messages, but when they do so
Hello,
here Is my Problem:
I want asterisk to sent none local URI to SER
My config asterisk svn-trunk:
UA===SER=ASTERISK===SER===sip URI
---INVITEINVITE-INVITE---
491-491 req pending
I set a peer with outboundproxy so in extensions.conf
I forward to ser non local URI
I second this wish.
On 9/14/06, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message- From: Norris, Sam [mailto:[EMAIL PROTECTED]
] Sent: Thursday, September 14, 2006 8:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] BLF across asterisk trunks
2 asterisk boxes
ZoIP 0.2.0, the Zork/Asterisk bridge has finally been released. Now you
too can play 80's era text adventures over the phone using
text-to-speech, and speech recognition ;-)
What's a text adventure like you ask? Well, depending on your skill, a
typical dialog might go something like this:
Tell them Vonage uses asterisk for VM.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, September 14, 2006 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello...
We have a SIP provider for inbound dialtone that
periodically goes into Unregistered state (per sip
show registry) and doesn't seem to recover. Most
often it seems to happen after storms, when our office
DSL may have gone wacky for a little bit, but will
then stay down even days later.
Steven Totaro wrote:
So the definity is not sending the callerID?
In the trunk group there should be an option to send number and name.
Also I think there may be something in the system configuration.
This system has no D channel, the data comes across via DTS on a TN767.
Doug
--
Ben
There is no voicemail in the old folder, I've manually inspected the
folders where this is occurring via 'ls'. The mail is in fact gone.
Andrew
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, September 14,
or at least have a SIP extension check both systems without having to point
them at both ... can't asterisk have a hint for an extension across a trunk
?
Sam
-Original Message-
From: Norris, Sam [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 14, 2006 8:54 AM
To:
Hello,
here Is my Problem:
I want asterisk to sent none local URI to SER
My config asterisk svn-trunk:
UA===SER=ASTERISK===SER===sip URI
---INVITEINVITE-INVITE---
491-491 req pending
I set a peer with outboundproxy so in extensions.conf
I forward to ser non local URI
Joel Lansden wrote:
Hello all,
I have been trying to solve this problem for days, with no luck.
When I run an AGI script from my extensions.conf, it seems no matter
what I do, the WAIT FOR DIGIT command will not work. The system just
flies past it without waiting a single millisecond, and
Douglas Garstang wrote:
Is there any documentation, maybe at voip-info.org, available for Asterisk 1.4?
Either in the form of _new_ docs, or docs that outline the differences and new
features that will be available in 1.4?
I'd like to avoid the months of trial-and-error that I went throo with
Hi Andrew,
Thanks for the response. Interesting.
But one thing though, both extensions are softphones actually.
The one on 108, is actually VoiceGenie that I'm testing with Asterisk.
But I'm trying to explain why I'm getting some glitch with the systems
sometimes with my softphone,
and I
News stories, press releases, etc are helpful. Anecdotes are less so.
How do we know that Vonage uses Asterisk for VM? Allison does other
voice work in addition to Digium stuff.
Steven Totaro wrote:
Tell them Vonage uses asterisk for VM.
-Original Message-
From: [EMAIL PROTECTED]
How can DTMF be sent down a channel?
I am thinking of method where say a channel id can be grabbed from
Asterisk Manager events and a DTMF signal sent down that channel,
through AGI, Asterisk Manager Interface or whatever?
Is it possible to have a command in extensions.conf which can take
both
Do they really? Wow. Cool. =) The more you know...-brandonOn 9/14/06, Steven Totaro [EMAIL PROTECTED]
wrote:Tell them Vonage uses asterisk for VM. -Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]] On Behalf Of Eric ManxPower Wieling Sent: Thursday,
Hi,
I was having a conversation with some friends and one of them brought up the
question about find out information about one sip number. Is there any way
to find out that a number, for instance 1(646)- is actually
[EMAIL PROTECTED]
In other words, I will know that the number belongs
I've got a curious situation. I have a client that is adding a new
site. At the moment they have a single 841 at the remote locaiton. The
Asterisk PBX is on a public IP simple iptables firewall.
On the remote location where the SPA841 is located they're running a
Win2003 ISA server as a
Michael wrote:
extreme echo. After 4 seconds, however, the audio transmission stops.
Even though the audio stops, the MeetMe is still in progress until the
user who initiated the page hangs up.
Maybe the 4 second time limit is within the AGI itself?
Doug
--
Ben Franklin quote:
Those who
In article [EMAIL PROTECTED],
Moises Silva [EMAIL PROTECTED] wrote:
Why oh why do so many people do all this modprobe stuff manually or in
rc.local etc.?
If you are running a RedHat / Fedora / CentOS distribution, just do
make config in the zaptel directory, and it will create a
Doug,
Do you think that you could fix my Definity problem if I paid for time
and gave you dialup?
Thanks,
Steve Totaro
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, September 13, 2006 9:56 AM
To:
Steve,
Yes Box1 does have multiple other peers. Do you know of something to try?
Jerry
/ -Original Message-
// From: asterisk-users-bounces at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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Jamin W. Collins wrote:
periodically, I've been getting reports from users of being
disconnected in mid-conversation. I've checked the system's logs for
any indication of problems and they all appear clean. Eventually, I
enabled both PRI and SIP debugging in an effort to track down the
Eric, contact me off list and I will give you a nce exemple with a worldwide
Asterisk network ;-)
Francois BERGERET,
France.
f6hqz-m_at_hamwlan.net
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Eric
ManxPower Wieling
Envoyé : jeudi 14 septembre 2006
Hi,
I would like to give a caller the chance to leave a queue after an agent has
already accepted the call.
The caller enters the queue by dialing 333:
[from-sip]
exten = 300,1,Answer()
exten = 300,2,Queue(q1|tT)
When the caller presses # and e.g. 1, asterisk is looking for this extension
in
On Thu, 2006-09-14 at 10:45 -0600, [EMAIL PROTECTED] wrote:
Any Idea
Yes I have an idea. How about you comply with list etiquette? Did you
not read Steven Critchfield's reply to your previous post? Or did you
think that you are the sole person in the universe to which some simple
etiquette does
I changed things so that the dialplan would answer, THEN launch the script, but
this made no difference. The script still won't wait for DTMF tones from the
caller.
~Joel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pato Valarezo
Sent: Thursday,
On 13 Sep 2006, at 22:56, Chris Bagnall wrote:
Greetings list,
Has anyone done any research into call routing and transcoding
performance
using a Via Epia based platform?
Be a bit careful on the transcoding issue.
We have a
VIA Nehemiah
1Ghz acting as our office PBX, it has a single
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Brian Candler wrote:
Is there a document somewhere giving the correct TDM400P FXO settings for
use on a BT PSTN line in the UK? All I can find is
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
but it doesn't give the complete settings,
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