Hi Al,
That was it, Thank you!!!
Al lists wrote:
check tz option in your voicemail.conf
On 10/5/07, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I have a really oddball time problem. When I check the server time
using
'date' it is correct. When I
it is correct. When I hit *98 and then my extension the CID says
a time that is some 6 hours off (early)??? I am really confused where
could CID be getting this bogus info???
I am using Centos 4.5, Asterisk 1.2.7.1 and Freepbx version 2.3.0.3
Thanks
Chuck Bunn
Hi,
Can anyone tell me the pros and cons of Proximity Detection using
bluetooth versus using GSM cell phone with receivers. I like the idea of
calls be transferred to my cell phone when I am away from the office
and I would like to implement such a system.
Thanks
Chuck Bunn
to be on and
this eats power.
Thanks
Chuck Bunn
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Lacy Moore - Aspendora wrote:
On 1/12/07, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell
phone
to the line we get a busy signal...
There was something similar to this posted a few months ago. What
Hi,
I am having a weird problem with one of my incoming lines. After a
reboot everything is fine if I disconnect the line from the wall and
reconnect it. After an hour or so the lies goes busy but no indication
of this shows up on the Flash Operator panel. I also do not see anything
in the
Hi,
I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP
phones. The phone can no longer receive calls. DND is not turned on and
the phone has the exact same configuration as the other 2 phones (they
each have unique extensions and such but all other settings are the
same.)
Hi,
I am having a weird problem with one of my incoming lines. After a
reboot everything is fine if I disconnect the line from the wall and
reconnect it. After an hour or so the lies goes busy but no indication
of this shows up on the Flash Operator panel. I also do not see anything
in the the
Hi,
Sorry I forgot to mention that the phone is showing registered and 'sip
show peers' shows that it is registered. Also the user can make outgoing
calls without a problem.
thanks
Eric ManxPower Wieling wrote:
Chuck Bunn wrote:
Hi,
I am having an odd problem with one of our Zyxel P
Hi,
I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone
to the line we get a busy signal...
Thanks
Lacy Moore - Aspendora wrote:
On 1/12/07, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I am having a weird problem with one of my incoming
10.0.0.35D 5060 OK (312 ms)
410(Unspecified)D 0UNKNOWN
31 sip peers [30 online , 1 offline]
turnip*CLI
Thanks
Eric ManxPower Wieling wrote:
Chuck Bunn wrote:
Hi,
Sorry I forgot to mention that the phone is showing registered
Hi,
I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP
phones. The phone can no longer receive calls. DND is not turned on and
the phone has the exact same configuration as the other 2 phones (they
each have unique extensions and such but all other settings are the
Hi,
Can anyone recommend a large button/type sip phone (VOIP) that an older
person could use. I have a client that needs to have large button phones
for elderly residents in her facility.
Thanks
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Hi Cory,
Do you know if any of the ATA to SIP converters that can handle visual
indicators (flashing light in addition to ring sound) and for that
matter can Asterisk handle visual indicators for ringing?
Thanks
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Hi,
Does anyone know if there is a blue-tooth wireless headset that works
with asterisk and/or a SIP software phone on the PC?
Thanks
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Hi,
So am I to understand that the visual indicator responds the same way a
ring would and thus if Asterisk tells a phone to ring the visual
indicator uses that signal and does not require a separate signal? I
guess I am use to seeing visual indicators in hotels that blink when
there is a
Hi,
Can anyone direct me to where I might find examples of handling
interactive input from a phone using PHP and AGI. I want to have someone
dial an extension and then have the system request input from the user,
take that input and put it into a database.
Thanks
Hi,
What does the following error mean:
Apr 5 12:39:40 NOTICE[22755] app.c: Huh? no dial for indications?
Here is the 'full' log around the error:
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to
agent '3002', on 'Local/[EMAIL PROTECTED],1'
Apr 5 12:38:24
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Here is the portion of the log:
Apr 5 14:02:42 NOTICE[23363] chan_zap.c:
Hi,
I have debug off (debug level 0) why are the following lines showing up
in '/var/log/asterisk/full'
Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on
Hi,
Thank you that was it, I had 'debug' listed under 'full' in logger.conf.
Not sure how I missed that...
Thanks Again
Filip Drągowski wrote:
Look what you have in /etc/asterisk/logger.conf
find:
console =
message =
full =
Hi,
I have debug off (debug level 0) why are the following lines
Hi,
I am not sure if this is a bug in FOP (Flash Operator Panel), a
configuration error on my part or a bug in Asterisk. I am using Asterisk
1.2.5 and Zaptel 1.2.4 on a Centos 4.2 server with Linux version
2.6.9-22-EL-i686. Kernel updates are excluded and the server has been
updated using
Hi,
What does your SIP config look like for the SJPhone? Also what operating
system does this PC have and is it up to date with security and bug patches.
Thanks
Marco Mouta wrote:
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i
sjphone?
On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
What does your SIP config look like for the SJPhone? Also what operating
system does this PC have and is it up to date with security and bug patches.
Thanks
Marco Mouta wrote:
Hi all,
I've my Server running well
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
Hi,
This is interesting you used the ctp profile. Did you document how you
changed it?
Thanks
wendell hamilton wrote:
I am able to pickup, hangup, and flash, using the buttons on the phone
with all of the soft clients and both of the headsets I mentioned below.
I don't believe I had to
Hi,
Anyone know where I can download or purchase a Bluetooth stack that
supports the CTP (Cordless Telephone Profile). Apparently this in the
only way to get the answer/hangup button on a wireless headset to work
with SIP soft phones. I have looked at the Widcomm site and they are not
Hi,
After much searching I have found that it might be possible to get a
bluetooth headset to answer/hangup with SJPhone or Xlite if the headset
supports handsfree mode. My Toshiba bluetooth stack supports this but I
have not been able to figure out how to enable it. Also Windows XP
desktop
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 27, 2006 6:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with
SJPhoneor X-lite
Hi,
After much searching I have found
not having both of these in the Bluetooth service
selection, you won't end up with the results you're looking for.
HTH
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 27, 2006 9:46 AM
To: Asterisk Users Mailing List - Non
Hi,
Is it possible to get a blue tooth headset to work with SJPhone or any
other SIP phone for that matter. Adapting the single button
(answer/hangup) on the blue tooth headset to answer/accept a call has me
stupefied. I must be missing something, it can not be that hard?
Thanks
Hi,
Without separate incoming and outgoing context you could not secure your
system from an outside caller using your system to dial a long distance
number.
Here is an example outgoing context that restricts who can call long
distance. If a SIP phone does not belong to the 'longdistance'
Hi,
Check your context you need to specify voicemail as [EMAIL PROTECTED]
(context seems to have been more tightly enforced since version 1.2 came
out). Below is an example of one of the macro I use for extensions...
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten =
Hi,
I disagree that contexts are not for outgoing calls, how else do you
restrict certain user to local calls only without using contexts?? On
the subject of grouping extensions I use pickup groups so that any
person can answer any phone in their immediate area by using a '*8' (as
long as
Hi,
I use the Zyxel P-2000W v2 wireless VOIP phones with Zyxel G-1000
access points and the hand off calls fairly smoothly using a port for
the hand off and using WEP security (the Zyxel is not capable of WPA
security yet). I understand that people have problems with some
manufactures
Hi,
Without separate incoming and outgoing context you could not secure your
system from an outside caller using your system to dial a long distance
number.
Here is an example outgoing context that restricts who can call long
distance. If a SIP phone does not belong to the 'longdistance'
Hi,
Is it possible to turn off the request for a security code when
transferring in FOP (Flash Operator Panel)? If not can the security code
be set to use the SIP or voicemail passwords? I know there is a forum
for FOP but no one seems to be answering there... so I thought I would
see if anyone
Hi,
Is there some way to set the priority for the poup window that SJPone
displays when the phone is ringing (the popup that asks you to accept or
ignore the ringing phone) so that it is always on top. We have users
thas are using applications that will pop over the SJPhone pop up window
and
with your .config and try
again.
on Monday 03/13/2006 Chuck Bunn([EMAIL PROTECTED]) wrote
Hi,
I am having the same exact problem. I am assuming that it was a problem
with a kernel update I did. I am in the process of rolling back to an
older kernel... I will let you let know if this works
Hi,
When an agent is logged in and his phone goes off the network (using an
SJPhone on a portable PC) and the user had forgotten to log out of the
queue all calls go to this agent that is no longer connected to the
system. I have tried training and retraining but I need some way to fix
this.
from the mirror and it worked
like a charm!
HTH
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 13, 2006 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Failed
] On Behalf Of Chuck Bunn
Sent: Monday, March 13, 2006 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Failed installing zaptel
Hi,
I am having the same exact problem. I am assuming that it was a problem
with a kernel update I did. I am
Hi,
I made a big mistake on a Centos 4.2 box - I forgot to exclude the
kernel from updating. Now zaptel will not do a make linux26 see below.
Is there a way to roll this back or is there a patch to get Zaptel to
compile? I have a link to the modules using 'ln -s /lib/modules/uname
-r/build
Hi,
I am having the same exact problem. I am assuming that it was a problem
with a kernel update I did. I am in the process of rolling back to an
older kernel... I will let you let know if this works. There is also a
patch for zaptel but I believe this is for going from 1.3 to 1.4?
Thanks
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I
Hi,
I saw this problem mentioned before but the user appeared to be using
the MP3 software with asterisk. I am using the native music on hold
player in asterisk 1.2 and I too have a volume problem with music on
hold. Is this controllable through the 'indications.conf'? I know this
file
the transferer
TRANSFER_CONTEXT. That context is used to match the extension to dial.
So you can set this var to any context you want.
Regards
On 2/23/06, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
Is setting the variable _TRANSFER_CONTEXT required in
features.conf
to transfer a caller from the spa to the resturant and vise
versa. There are seperate phone lines comming in for the spa and
resturant as well.
Thanks
Moises Silva wrote:
it seems im not undestanding your question then. Could you provide a
practical example?
On 2/24/06, *Chuck Bunn* [EMAIL
Hi,
I recently moved all of my conf files over to a new Asterisk 1.2.4
server and every works except the features enabled in features.conf. Was
there a syntax chnage in 1.2.4? Or is there something else... Here is my
features.conf:
[general]
parkext = 880;
Hi,
I am getting repeated messages in my logs with the following:
*
Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default'
Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be
handled, bad request: [EMAIL PROTECTED]
Feb 23 07:56:12
Hi,
Is setting the variable _TRANSFER_CONTEXT required in features.conf for
Asterisk 1.2.4? It appears from the Wiki that transfers across contexts
are not possible when this is set. If it is not set can one trasfer
across contexts??
Thanks
___
Hi,
I got rid of the messages I was getting in the CDR (pbx.c: Cannot find
extension context 'default') by adding a blank 'default' context at the
front of my extensions.conf (I use the context 'extensions-home') this
also (well sort of ) fixed my problem with blind transfers. I can blind
Incantalupo
Chuck Bunn wrote:
Hi,
I thought I had this problem licked but there still is a rights
problem with ARI and Asterisk when using a non-root user (Following
the wiki at
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25).
When I issue the following:
chmod
Hi,
Just so I am clear this patch will work with 1.2.4 and requires manual
updating to files and then a recomplie of Asterisk source correct??
Thanks
Ben Klang wrote:
Hello,
I found the same problem very frustrating, mostly because it causes Asterisk
to ignore ACLs and umask settings.
Hi,
Could you post the updated patch for 1.2.4
Thanks
Ben Klang wrote:
On Thursday 16 February 2006 11:47, you wrote:
Just so I am clear this patch will work with 1.2.4 and requires manual
updating to files and then a recomplie of Asterisk source correct??
This patch was written
Hi,
Anyone got AFOP working. The install instructions tell you to copy all
of the files extracted under the 'html' directory to a subdirectory
under your main web directory (in my case this is /var/www/html/panel/)
and then the instructions talk about modifying the 'op_server.cfg' file
but
more...
Thanks
Giorgio Incantalupo wrote:
Hi Chuck.
I had the same problem.
I solved it using the externnotify parameter inside voicemail.conf.
Just launch a script which changes the /var/spool/asterisk permissions.
Giorgio Incantalupo
Chuck Bunn wrote:
Hi,
I thought I had this problem
Hi,
I thought I had this problem licked but there still is a rights problem
with ARI and Asterisk when using a non-root user (Following the wiki at
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25).
When I issue the following:
chmod --recursive u=rwX,g=rX,o=
problem. If you have not tried the ARI interface you might look at it,
my clients love it!! Great job DAN
Thanks
Chuck Bunn
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in as Admin with ari_password I see all of the call
detail but still no voice mail. Any ideas where I might look for my
problem. Voicemail is working since I can call the voicemail extension
and retrieve messages. I am not using AMP and I have set the standalone
flag to true.
Thanks
Chuck
Hi,
Does anyone know if it is possible to setup an SJPhone with an external
ringer of some sort. One of the operators may not always be at her desk
and when she is not wearing a headset she cannot hear the phone ring -
is there some way to fix this?
Thanks
Hi,
I have not been able to find anything about persistent agents in any
wiki? Where does this command go and what is its syntax?
Thanks
Michiel van Baak wrote:
On 18:06, Tue 27 Dec 05, Bud Bach wrote:
But, if the agents don't log out for some reason, they will still be logged
in the
:
It is set in the queues.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chuck Bunn
Sent: Wednesday, December 28, 2005 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Automatic logoff of all agents
Hi,
Oh sorry I am using asterisk 1.2.1
Thanks
Kevin P. Fleming wrote:
Chuck Bunn wrote:
When I add 'persistentmembers=no' in queues.conf and reload I get a
message in the message log file saying unknown keyword
'persistentmembers'. I got the syntax from
http://www.voip-info.org/wiki/view
Hi,
Is there a way to force the logoff of all agents at a set time say
8:00PM or do I have to do an agentcallbacklogin with each agents
credentials? I am using Asterisk 1.2 The wiki shows an extension that
the agent calls to preform the logoff - I need something that is
completely automated
Hi,
I understand how GototIfTime works but that still leaves agents logged
in and if an agent is absent the next day calls will go to an agent that
is not there.
Thanks
Michiel van Baak wrote:
On 16:22, Tue 27 Dec 05, Chuck Bunn wrote:
Hi,
Is there a way to force the logoff of all
Hi,
What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings. If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work. Also since your
hear the phone sometimes you may be experiencing QOS issues on your
network. Doe
, Chuck Bunn [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings.
I use gsm.
If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work.
I changed
is used that packets could time out and you would never
know it since UDP is dumb and has no packet loss recovery mechanism.
What is the topology of your network. Is the Asterisk box and the client
on the same backbone and switch?
Thanks
Evil Skymarshal wrote:
Hi Chuck,
2005/12/17, Chuck
. Is the Asterisk box and the client
on the same backbone and switch?
Thanks
Evil Skymarshal wrote:
Hi Chuck,
2005/12/17, Chuck Bunn [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings.
I use gsm.
If you
to the problem. I will use your command to see if 'asterisk.pid'
inflates over time...
Thanks Again
Tzafrir Cohen wrote:
On Wed, Dec 14, 2005 at 08:49:06AM -0700, Chuck Bunn wrote:
Hi,
I am planning on restarting asterisk nightly as I seem to be
experiencing some sort of memory leak (Asterisk
Hi,
'#' (fast forward) and '*' (Rewind) not working in VoicemailMain with
Asterisk 1.2.1 Do I have to do something in the dialplan to make this
work? I have '##' set as a blind transfer and '*2' set as a attended
transfer in features.conf. Per the Wiki Voicemailmain has the following
Hi,
Please excuse the double post but I am about to report this as a bug and
I want to verify that others are having the same problem. Also I have
seen numerous bugs reported that are not bugs but just misconfiguration,
etc. and I do not want to burden the developers with a frivolus bug
Hi,
I am planning on restarting asterisk nightly as I seem to be
experiencing some sort of memory leak (Asterisk slows down over time). I
have reviewed the Asterisk suggestions for management and one item is
the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is
the recommend
OOPs I forgot to mention I am using Asterisk 1.2.1 and I had the same
problem with 1.0.9 and 1.2.0
Chuck Bunn wrote:
Hi,
Please excuse the double post but I am about to report this as a bug
and I want to verify that others are having the same problem. Also I
have seen numerous bugs
Hi,
Please excuse the cross post but these seems to be one of those issues
that may be answered by a developer or someone with direct
administrative knowledge of the deep workings of Asterisk. I have
deleted my log files expecting them to be recreated by Asterisk 1.2 but
nothing happens
Hi,
Two things does your codec set in X-lite match what is set in the sip
file and have you rebooted since setting up music on hold. I should also
ask if ran a make and make install in the asterisk-addons directory,
this installs a mp3 player (among other things) in Asterisk 1.2?
Vipul
Hi,
Push the '#' key followed by the extension for a blind transfer.
Thanks
Denny Schierz wrote:
hi,
my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls
from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)?
Or when the phone 401 rings,
Hi,
A while back I made the stupid mistake of deleting my log files 'full'
and 'messages' for asterisk. I recreated the files by 'touch' filename
and I have gone into the Asterisk CLI and tried both 'logger restart'
and 'logger rotate' but the logs still show nothing. I run 'logger show
Hi,
To pick up another persons phone that is ringing dial '*8' followed by
their extension. To do an attended transfer dial '*2' followed by the
extension...
Hope that helps
Denny Schierz wrote:
hi,
Quoting Chuck Bunn [EMAIL PROTECTED]:
Push the '#' key followed by the extension
them , and asterisk will recreate them as user asterisk,
or chown them, or change them to 777
best of all, delete them!
Marco.
Chuck Bunn wrote:
Hi,
A while back I made the stupid mistake of deleting my log files
'full' and 'messages' for asterisk. I recreated the files by 'touch'
filename
Hi,
Does anyone have any details about the Linksys one product that was just
announced? Does it use Asterisk?
Thanks
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Hi,
Why do changes to musiconhold.conf require a reboot. Also if I put mp3's
into the /var/lib/asterisk/mohmp3 directory will the be played if I use
the -r option? Using Asterisk 1.2 and have run the make config in the
/usr/src/asterisk-addons directory.
Thanks
Hi,
Can I escape a call queue by pressing a '*' or do I have to use a digit??
Thanks
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- Original Message -
From: Chuck Bunn [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, December 03, 2005 2:06 PM
Subject: [Asterisk-Users] Can I escape queue with a '*'?
Hi,
Can I escape a call queue by pressing a '*' or do I have to use a digit??
Thanks
Hi,
I setup music on hold as directed for Asterisk 1.2 but still no music on
hold. Any ideas what I did wrong. I see it start in the CLI but then it
immediately stops?? I also see the Hangup occur 20 seconds later as it
should according to WitMusicOnHold(20). I used a test setup suggested in
Hi,
Is it possible to get to a menu system while in a call queue. I want
users to be able to hit the '*' and be able to goto a menu system from a
queue if they so desire. I thought the following would do this but no
dice...
*
extension.conf
[general]
#include macros.incl
Hi,
If I have an extension in a context and I have another context with the
same extension and I include the second context in the first does this
cause a conflict or does Asterisk know that there is a 600 extension in
each context
[big-business]
exten = 600,1,Dial(ZAP/1,20)
include =
Hi,
In Asterisk 1.2 according to the wiki and I quote:
It is now possible to use multi-digit extensions in the exit context
for a queue (although you should not have overlapping extensions, as
there is no digit timeout). This means that the EXITWITHKEY event in
queue_log can now contain a
has lower priority.
Hope this helps.
Andy
On 12/2/05, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
If I have an extension in a context and I have another context
with the
same extension and I include the second context in the first does
Hi,
Not sure if this is by design or my error... Several agents are logged in. One
of the phones was turned off without the agent logging off first. After the
phone was powered down all calls routed to the powered off agent and no other
phones rang. Is there a way to turn this behavior off. (I
H,
Not sure if this is normal but I thought the coma ',' was replaceable by the
pipe command '|' and vice versa? When I used comas instead of the pipe command
in AgentCallbackLogin certain SIP phones do not here the operator prompts when
calling the agent extension. Is this normal - I thought the
Hi,
Is it possible to paste phrases together and if so how do I separate each
phrase.
exten = s,4,BackGround(to-compose-a-message,press-1)
and
exten = s,4,BackGround(to-compose-a-message|press-1)
do not work...
Thanks
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Hi,
Yes that does work but in some cases it just seems it would be clearer
(also less code) to be able to have them on one line...
Thanks
Anthony Rodgers wrote:
exten = s,4,BackGround(to-compose-a-message)
exten = s,5,BackGround(press-1)
doesn't work?
On Nov 29, 2005, at 3:41 PM, [EMAIL
Hi,
I understand that a fax machine cannot connect through a Digium TDM400p
card (FXS connected to fax and FXO connected to a pots line) but can
spandsp send and receive faxes as an intermediary between the pots line
and the fax machine.
Thanks
the
echo and jitter control did not work as well, but at least now the
phones can be used to ack as an agent.
Thanks
Chuck Bunn wrote:
Hi,
Okay we have agents logging in to receive calls from a queue. Agents
logging in from a SJPhone (SIP Phone) can dial the login extension and
are asked
Hi,
I have now tried other strategies including random and round robin. I am
beginning to think there is some sort of bug with Agent groups? I will
try assigning members to a queue not by their group but individually.
Thanks
Chuck Bunn wrote:
Hi,
In the queue.conf I have set the strategy
member = Agent/511
member = Agent/512
***
Thanks
Chuck Bunn wrote:
Hi,
I have now tried other strategies including random and round robin. I
am beginning to think there is some sort of bug with Agent groups? I
will try assigning members to a queue not by their group
Hi,
Does anyone know how to implement the tranfer feature (button) on the SJPhone in
extension.conf
Thanks
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Hi,
Does anyone have a sample config for phones (like the Zyxel P2000wv2) that
cannot handle more than one line. I have tried using # followed by the
extension and nothing happens??? I have parking setup but for some reason we
cannot retrieve the parked call. I call the user who the call is
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