First, I'm not sure is this Asterisk or ooh323 channel problem.
It seams that I have solved (I do hope so!) deadlock problem with ooh323
(thanks to Sean and his patch). Now I have another one. It seams that some
channels stay open even they should not. This is what I see from CLI:
pbx*CLI show
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Everyone is free to implement products based on what their heart tells them
and not their head. I don't know a lot of people that manage to stay in
business very long doing things that way though.
I agreed with you. And I must add
is there a difference between 1.2.1 and 1.2.2 ?
Yes there are. Read the Changes file.
--
Tomislav Parcina
tparcina#lama.hr
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi Folks
I have had a look through the Features list, and I see that the system does
support an auto attendant, however is it possible to have say 5 telephone
numbers that a person would dial and have 5 different messages
Yes, it's
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi all,
I'm experiencing problems using H323, with 60 calls, * crashes...!
My server is a PIV with 2 Gb, on CentOS 3.6, Asterisk 1.2.5,
openh323_v1_17_1, asterisk-addons-1.2.2.
Errors messages, are:
chan_h323.c:1483 cleanup_connection:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hardcoding IP in /etc/hosts works but it's not reliable if voip
provider's IP change.
Does someone already as a similar problem and resolved it?
I head similar problem, but I didn't solve it. If you find workable solution,
please mail
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
For now, I use Via EPIA PD6000. Pretty small and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm and [EMAIL PROTECTED] user - been so now for almost a year.
Lately, I've upgraded to the latest greatest.. (which is built on 1.2.5)
and am unable to install oh323.
I don't like to bring bad news, but I'm unable to install h323
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Guyz!
Is there any known issue with ooh323 and g729? I am experiencing one side
voice okay from ooh323 extension to sip ext, but on reverse side voice
quality is very poor. Any thoughts?
I have one question. Are you using ooh323 from
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I never so this error.
I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
You are using ooh323 from asterisk-addons-1.2.1?
If so, you are experiencing the same but that I have. There is patch (which I
can't apply) or
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Above products will work with any commercially
available motherboard in the market. This we have
proven true because we have at least 5 different
servers that we have installed them to. And take note
of their on-board echo cancellers,
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Its also very important to recognize that some of us have problems with
some of the digium products where the equivalent sangoma product
resolves that problem. If it were not for the sangoma products, asterisk
would not be installed in
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You just add the same agent to both queues (don't use groups), like in
queues.conf:
[queue1]
member=Agent/101
[queue2]
...
member=Agent/101
Now Agent 101 is a member of both queues, and will not be called while
s/he
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi all,
I would like to hear from you, SjPhone has the option to Do not Send
silence (audio options, advanced), should i use this or remove this
option?
Everything ran well until now, but there was few people on my server,
i'm
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
The h323 channels doesn't have any support for NAT. You'd need to
register with a properly configured gnugk for that.
E didn't mention NAT, he only spoke about firewall.
--
Tomislav Parcina
tparcina#lama.hr
How should I upgrade addons form ver. 1.2.1. to 1.2.2.?
I'm particularly interested how to upgrade ooh323 channel driver.
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Tomislav Parcina
tparcina#lama.hr
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Hi group!
I'm having problems with ooh323 (ver 0.3?!? - the one that comes with asterisk
addons 1.2.1) and I need to know what h323 channel driver you use in production?
Have a nice day!
--
Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Just add the agent to both queues, * will take care of the rest.
l.
I have tried to put agents in groups and then join groups to specific queue. It
doesn't work. I don't know is the problem because one agent can't be in more
groups or
What is maximum length of name in caller ID? How much charters can I put and be
sure it will work fine?
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Please stop send me email
Best Regards,
Mr.Peeramate Rochanasmita
Project Manager/General Manager
This message was sent to me?
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Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and
I can't transfer call which was picked up with feature - group pick up. I'm
running * 1.2.5.
The problem is that asterisk doesn't hear that I have pressed #1 and doesn't
play transfer sound for me.
Regular phone calls I can transfer without problem. Can anybody check is this
a BUG?
--
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Absolutely right :)
\ escapes the next character, so if you wants *69 to go through
immediately, you'd put \*69 so that the * gets recognized as a digit.
, returns the dialtone sound. When my users hit 9, they like to hear
the
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
There is no such thing as a 'free' G.729 - The DSP Group has claimed and
defended the Patents they hold against the algorithm and process.
Please do not use Asterisk/Digium related resources to exchange this
information - They are
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/
1.) setup your /etc/asterisk/sccp.conf with something like:
2.) setup lines 30/31 as a custom extension in astersik (i used amp)
and had it dial SCCP/30 and SCCP/31
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Yes, my mistake in /tftpboot/SEPMAC.cnf.xml. Having said that, Please
double check that you have set the line:
permit=192.168.1.90/255.255.255.255 ; This device can register only
using this ip address
or in your case:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Marty,
But with the same 128 bit upstream circuit, directly connecting the SJPhone
the Stun server and using ulaw, everything is perfect. The problem comes
when i am putting Asterisk in the picture.
I have used SJ Phone softphone. His
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
i can confirm that this exist on 1.2.5, and the last time i said this, the
original poster was supposed to file a bug on bugs.digium.com.
OK. Can anybody else confirm this? I don't wona report it if it isn't bug.
--
Tomislav Parcina
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
It's a toll free number. You can call it from anywhere and the costs of the
call go on the callee not the caller.
Thank you.
--
Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and
I have running Asterisk 1.2.5 with addons 1.2.1. on Fedora Core 4. I have
installed ooh323 from 1.2.1 addons.
How to upgrade addons to 1.2.2 version and install new ooh323 driver? Do I need
to install addons 1.2.2 if I only need new ooh323 driver?
Can I just untar addons, and run make clean;
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
That's good to know... this only affects 8.2, right?
As far as I know, yes. I have been using 7.5 and now I use 7.4 on 7940 and 7960
and I didn't have those issues.
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have setup several queues for a customer. Their periodic announcement says
please wait for the next available agent, or press * to leave a voicemail.
This does not work when the message is playing. The message stops, but the
user is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Sorry for being a bit of a newbie here but I find the
docs or README for downloading the G.729 codec from Digium
are not as detailed as I would like or just don't really
break down the different versions to a point that I am clear
on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
now, i just edited the Makefile that comes in zaptel directory to disable
any usb, as i am not going to use any usb device in my asterisk, and it
compiles and work ok.
Hi Raul!
Please send us what lines did you comment. Does it work with
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323
module and g729/g723 free codecs too.
Can you send us more information about this free g729 codecs?
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Can someone tell me how I can configure ooh323.conf to accept call
from h323 gateway (only the authorized h323 gateway) to my asterisk.
Sorry, this is not answer to your question, but I need to ask you something.
Are you using ooh323
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought
I'd pick a few brains first.
I'm not looking to configure the Asterisk server itself, VI works adequately
for that. But I want to give Web access to as
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi !
What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting
from new)
sip reload - (cli command sip reload). Is sip reload part of the
reload
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You have to set up a dialplan.xml file in your tftpboot directory for the
phone to pull:
DIALTEMPLATE
TEMPLATE MATCH=9,59. Timeout=0/
TEMPLATE MATCH=9,29. Timeout=0/
TEMPLATE MATCH=9,832... Timeout=0/
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if
I
hang up during Playing 'conf-onlyperson' my machine freezes. It's not a GSM
Enconding problem as I suspected first, this happens with every encoding.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
conf-onlyperson. This has been seen
I have search wiki, asteriskguru, chan_sccp and some other site's for
information's how to upgrade, and make Cisco 7970 IP phone to work with
asterisk on SCCP firmware.
I'm sure that there are users on this group that have working Cisco 7970 phone.
Please send me some information's how to do
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Exchange 12 will support OVA Outlook Voice Access. It will do this using
VOIP. I was wondering if anyone has given any thought as to how Asterisk
might interface with Exchange 12. It will do this using a VOIP gateway.
If this
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I believe the OP wants to use GSM handsets as extensions, like running
your own localized GSM network. That's not the same as using a GSM
terminal to connect Asterisk to the cellular network.
IP Access makes such products.
Hi group. I have huge problem. My pickup exten #8 isn't working.
This is what I have configured.
pbx*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 #8
In sip.conf I have
callgroup=2
pickupgroup=2
For
I use asterisk 1.2.5 and h323 that comes with addons 1.2.1.
Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and
tries to make attendant transfer to person B (local SIP phone). They speak.
Then A hangs up. Call form h323 trunk doesn't get to person B.
This is what I
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'd recommend the VIA EPIA PD series of boards. If you want to be safe
load-wise get the PD 1 which has a C3-2 (Nehamiah core) processor at
1Ghz. The board has two ethernet ports and one PCI slot. Combine this
with Astlinux and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You can do the same thing with DND. Turn the value on or off, then in
your dial string, check the database value and act accordingly.
Hi Doug.
Do you know how to, when leaving office, set all incoming calls to transfer do
my coworker?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Grretings to all,
I am having a problem with a customer's queue setup that I don't really
understand.
Has anyone come up with a solution for this, or know if Asterisk properly
treats Agents who are In use.
Hi Joe!
Asterisk treats
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
2. If you receive it because you have an 800 number, you are not
allowed to use it for anything else (read marketing) but billing.
Can you please tell me what is 800 number?
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
you are using the attended transfer feature..
ist it already possible to hang up before the other person lifts the handset
without loosing the caller when you are doing an attendet transfer?
(person A takes an incoming call, person A
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P
switch (with PoE functionality). I have tested three phone's, one is working
(7905) and two aren't (7905 and 7940). I have plugged all three phones on same
switch port with same cable!
Do I need to change anything in
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P
switch (with PoE functionality). I have tested three phone's, one is working
(7905) and two aren't (7905 and 7940). I have plugged all three phones on
same
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Nothing for sure, and you may already know this, but some early Cisco
phones only knew how to speak Cisco PoE, not the 802 standard which was
defined a bit later. The Cisco web site should tell you which phone
talks which protocol though.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
7905/12, 7940/60 are NOT 802.3af compatible
ONLY 7911, 7941/61/70 (and their gigabiteth variants), are 802.3af
compliant
Totally not true!
I have 7905 phone that IS 802.3af compatible. Its on my table, right to my
laptop from which I'm
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Tomislav, please look at:
http://powerdsine.com/Support/Certification/company_All.asp?company=Cisco
also on ci$co site you can't find info, that old phones are 802.3af
compliant, but pre-standard...
old ci$co phones can work with some
HI Group! I have strange problem. Since I started to use H323 with my VoIP
provider when I dial the person that is currently busy, I can't hear busy tone
on my handset. What could be the problem? What should I look for? How is this
exactly called (because I even don't know what to look for).
Hi Joe!
Thank you for your mail. The thing is that I have never
program anything so it will take a lot of my time, which I don't have right now.
Hopefully, when I finish started projects I'll be able to play with this
stuff.
In the meantime if anybody solves this problem, please let
the
How to upgrade h323 from Asterisk add-ons (from version 1.2.1 to 1.2.2)?
In INSTALL they don't say anything about upgrade...
Thank you for your time!
--
Tomislav Parcina
tparcina#lama.hr
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Hi group!
Does anybody knows about any news server that works the same way that Gmane
www.gmane.com/ does it? I was satisfied with Gmane for few months, but now it
seams that it doesn't work any more (no new posts in past few days). Now I'm
looking for alternative.
--
Tomislav Parcina
How can I send recordings, that I have recorded with One Touch Record, to
e-mail address that is defined in voicemail.conf?
Thank you for your ideas.
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Tomislav Parcina
tparcina#lama.hr
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Hi group!
How to set language for queue?
I have several queue's. In every queue, agents speaks different language. I
need to announce queue-youarenext and similar on different languages.
This is what I have in my extensions.conf and it does set language, but when
calls enters queue, it doesn't
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does this belong to my dialplan or my sip registration settings?
To your SIP registration settings. You should limit that user/peer/friend to
only one line.
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
what are the file permissions/ownership and are they readable by the
asterisk process ?
The problem was that wav files where in stereo mode. I have encode them and now
it works fine.
--
Tomislav Parcina
[EMAIL PROTECTED]
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You need to use mpg123 to convert the mp3 files to wav files first.
mpg123 -w out.wav in.mp3
This one works. Thank you!
sox out.wav -r 8000 out.gsm
I have problem with this command. It runs fine, but when I play that file it is
twice
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql
Chris
This is what happens.
[EMAIL PROTECTED] mohmp3]# ls
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
[EMAIL PROTECTED] mohmp3]# sox -V fpm-calm-river.mp3 -t au
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/
I have some wav files (with 755 permission). In musiconhold.conf I have
[native]
mode=files
directory=/var/lib/asterisk/moh-native
And in sip.conf I have
musicclass=native
When I put call on hold this is what I get
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
But when a call enters queue_1 or queue_2 it allways rings everyone directly
without checking if Agent1 is available or not. It should distribute the
calls from queue_1 to the other agents only when agent/1 is unavailable and
agent/1
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello Tomislav,
I borrowed F1000 from my friend for testing. I am not sure if that is
different from F1000G, but I am experiencing the following issues:
1. As a user, it is not easy to get a firmware update as I need to have a
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
i can confirm that this bug exists in 1.2.4 as well. we've managed to fudge
it by dialplan tricks and Pickup().
Please report the bug.
In 1.2.1 it works fine.
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You need to install either libmad or libmp3lame.
Sox checks for this on startup.
This is what I get when I enter yum install libmp3lame or libmad
Parsing package install arguments
No Match for argument: libmp3lame
Nothing to do
Parsing
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
what are the file permissions/ownership and are they readable by the
asterisk process ?
Asterisk runs like root and permissions are 755. So, as far as I know, that
should be fine.
--
Tomislav Parcina
tparcina#lama.hr
Does ooh323 from asterisk-addons 1.2.1 support alaw codec?
This is what is written in h323.conf.sample that can be found in
asterisk-addons dir.
The codecs to be used for all clients.Only ulaw and gsm supported as of now.
Default - ulaw
ONLY ulaw, gsm, g729 and g7231 supported as of now
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
How can you check if transcoding is configured to work properly on a system?
Is there a way of knowing that transcoding is configured properly and is
giving
some output to indicate so?
CLI show translation
--
Tomislav Parcina
How to disable silence suppression (or Voice activity detection - VAD) on Cisco
7905 phone?
On Cisco 7940 I use enable_vad: 0, but I can't find anything similar for 7905.
--
Tomislav Parcina
tparcina#lama.hr
___
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Where can I find alaw, ulaw, gsm, g729 formats for native music on hold?
I have some mp3 files and I have tried to transcode them to above, but it seams
that SOX can't do that. Please, tell me where to download some MOH files (in
above formats) or how to transcode mp3?
Thank you for your
Does anybody use UTStarcom F1000G Wi-FI VoIP phone?
http://www.utstar.com/Solutions/Handsets/WiFi/
I'm planning to buy one and I need to know did you have any problems with
phone. What is the sound quality? How close you need to be to the access point?
Please, any information's are useful to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
the callgroup/pickupgroup settings are correct...
dialing *8 or *8# on any client (zap/sip/sccp) results in unknown
extension...
To pick-up with SIP phone, it has to be defined in sip.conf. Same goes for zap
and iax2.
--
Tomislav
Situation. I call out from SIP phone over h323 trunk and called person decides
not to pick up (on mobile phone they press red button - NO - hang-up). Until
the called person press the NO button, I can hear ringing. When called person
press the button, I don't hear anything. Asterisk waits until
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Edit logger.conf and uncomment full.
Start Asterisk with the the -d option.
View debugging information in the /var/log/asterisk/full
Is -d option necessary?
Anyway, done that. Just thought that you think about something else.
Thank you!
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
callgroup and pickupgoup is configured in the config-files (zap/sip/sccp)
- is anything else needed ?
Sorry, I'm not up to this.
--
Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
When I try to call from asttapi one number, I get message No one is
available to answer at this time (1:0/0/0). Immediately after that I try to
call the same number from SIP phone (the same one that is used with asttapi)
and call goes
This is third time today that my Asterisk hangs up. It seams that I have
problems with h323. I'm using ooh323 from Asterisk add-ons. I have the
following configuration
Asterisk 1.2.1
Asterisk-addons 1.2.1
Fedora Core 4
I'm using SIP phones and
h323 trunk to my VoIP provider
Like I said this is
When I try to call from asttapi one number, I get message No one is available
to answer at this time (1:0/0/0). Immediately after that I try to call the
same number from SIP phone (the same one that is used with asttapi) and call
goes true.
What have I done wrong?
This is how it looks on CLI.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Asterisk's debug facilities need to be enabled before you'll get
debugging information.
And how do you turn on Asterisk's debug facilities?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
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
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello Asterisk community.
We have a small User-group in Melbourne Australia.
Recently I brought up the issue of STANDARDS for dialing Applications on
a PBX.
This generated some interest but also the fact little has been done on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
CDW and other large resellers like them have a difficult time selling
service contracts. The issue is they _must_ provide Cisco with a serial
number (of the phone) which is checked by Cisco to see if the company
...
First they are
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I installed FC4, ran command, # yum install asterisk. A bunch of stuff
happened, but can't locate .conf files. I have a list of files:
/usr/share/doc/asterisk-1.2.4/configs/features.conf.sample
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I don't know if this works for you, but I use the following mechanism. I
don't use the agent call back stuff, just the (Add|Remove)QueueMember stuff.
For each queue, dialing the extension (), puts the caller into the queue
(ie, a
I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I
need to buy firmware for them. I have contacted http://www.cdw.com and
http://www.insight.com/ but they didn't respond.
Can anybody tell me where can I buy SCCP and SIP firmware for my phones?
BTW, I'm in Croatia
One easy question for experienced users. Should I use Cisco VoIP phones with
SIP or SCCP?
What are the (dis)advantages of one or another? Please tell me your stories.
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Tomislav Parcina
[EMAIL PROTECTED]
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
But using the native transfer on the phone causes the system to think the
agent is still on the call
Yes, and I have desabled that options on my phones. Sometimes I have delay if I
use transfer or three way calling on Cisco phones.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does anyone know how to setup a linear type of queue strategy? By that
I mean that agents will be tried in a particular order and the call will
be routed to them unless they are on the phone or not logged in.
I want a 3rd party app to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi. I'm having trouble controlling the user info when sending e-mails
from asterisk via sendmail to a Microsoft exchange server.
When I receive the email the sender is always
[EMAIL PROTECTED] and the name of the sender is always
Added
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You can not define groups in sip.conf
But there are, as you hint, other ways to solve the problem, like using
queues or implementing it in dialplan logic.
Do you have any example how to do that?
--
Tomislav Parcina
[EMAIL
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k
K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp
bmEgPHRwYXJjaW5hQGxhbWEuaHI+IHdyb3RlOgo+Cj4gSGkgbGlzdCEKPgo+IEhvdyB0byBzZW5k
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Hi Patrick,
I'm new to Linux, so can you please tell me how
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I think it's a bit of a known fault - the attended transfer function
does not work from the queue system. It would be nice if it did, though.
Hi Paul!
Is there any explanation about this? Is that something that will change?
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You'll have to use uattended transfers for CCs.
l.
I have read Paul's mail. Is this bug or feature?
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Tomislav Parcina
[EMAIL PROTECTED]
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You have to use AgentCallbackLogin for that.
If a phone logs in that way, it's reachable as Agent/200
You can also use AgentCallbackLogin to logout the agent.
You don't have to worry about an agent that forgets to
logout on phone X
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