,Katarina Ivanisevic
agent = 404,404,Marija Bilic
agent = 405,405,Ana Kaliterna
Will this work? Will agents 401 and 402 be in both groups? If I join
every group to another queue, will one agent be in both queue's?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e
$
Thank you for any suggestions.
P.S.
If this is second time you see this message, then sorry for resending, but I
didn't see it on list...
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
)
Do I have to add one more line (311,n) which will define what will
happen with call if no agents are logged in that queue? Can I do that on
any other way?
Thank you for your time!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
with a call.
Thank you for your time.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
___
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Asterisk-Users mailing list
When I try to make att transfer (*2) of call that was in queue the call get's
disconnected. Blind transfer (#1) works fine. In dial plan I don't have any h
or H (hangup call with *). In features.conf I have this line disconnect = *0.
What could be the reason why call hang's up?
--
Tomislav
Hi list!
How to send a call directly to voicemail recording?
When I put this
exten = 313,n,VoiceMail,u221
Or this
exten = 313,n,VoiceMail,b221
In my dial plan, calling person first hears greeting message (busy or
unviable). I would like to avoid greeting message (I would play something with
When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why?
When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for blind.
In queue blind transfer works. For disconnect I have #0.
I guess that
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
Subject: RE: SIP Register
From: Tomislav Parcina [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
First impressions telling me you want to check your phone settings. What
phone are you using and what are the config settings?
Hi Mark, thank you for your reply.
I'm
Subject: RE: Queue - check agent
From: Tomislav Parcina [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Hello,
I might be wrong here, but I thought that in Queues.conf, if you defined a
queue with joinempty=no, or joinempty=strict then no calls will be placed in
Why do you think it's phone problem and not Asterisk? Asterisk is the
one that contents my provider. * is the one who should decide what
information's to send to my VoIP provider... Anyway, I'm inexperienced
with this and I'm just trying to understand what is happening and where
could be
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
How can this be done without AMP? Using personal queue's and agents? I need
information's to get better
You can do this with agents, no need for a queue.
Define agents in agents.conf
In your dialplan, instead of Dial(SIP/bedroom) use
Dial(Agent/200)
Let the phones login as agent :)
OK, I know I have to Dial(Agent/200), but how will I login agents if I don't
use queue? If phone log's in as
AMP doesn't do miracles! Look at its dialplan.
I believe he doesn't, but I don't have AMP installed. Next week I think I'll
have enough free time to try it. Will [EMAIL PROTECTED] do the trick?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
We have got some ATA for only $55 if you are interested?
Sam
Yes Sam, I'm interested. If they work with FAX I'll definitely buy one of them
for testing.
--
Tomislav
[EMAIL PROTECTED]
___
Since you have no Digium hardware (and thus no connection to POTS or
PRI)... are you routing your phone calls via VoIP? If so, it is not
recommended to run FAX via VoIP. The two don't mix. FAX is not able to
handle packet loss like VoIP. Also, any codec other than uLaw will not
even come
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
AFIK, fax is supported and installed with with app_txfax app_rxfax
If this proves to be true why would you need the ATA?
I'm working on this one. I have to install app_rxfax but I have failed. Soon,
I'll try again (hopefully next week).
Why don't you simply rotate the logs with logrotate ?
How to do that?
--
Tomislav Parcina
[EMAIL PROTECTED]
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
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?
--
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?
--
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
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have server with a total of 6 Analog ports...using TDM04B and TDM02B.
I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have
worked through getting the DIDs to work and route to the
extensions...now what I need to do is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Thank you, but this is how I see your mail. How can I see it right?
http://lists.digium.com/pipermail/asterisk-users/2006-February/146742.html
Thank you!
--
Tomislav Parcina
[EMAIL PROTECTED]
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...
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.
--
Tomislav Parcina
[EMAIL PROTECTED]
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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...
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
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...
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
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
___
--Bandwidth and Colocation provided by
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...
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
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]
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
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.
--
Tomislav Parcina
tparcina#lama.hr
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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
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).
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Try safe_asterisk , for an easy way to start asterisk in background,
a plain 'asterisk' is even better and safer.
asterisk -U asterisk . is better.
/etc/init.d/asterisk start
is similar.
Why is this better than safe_asterisk?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
i can see AOC messages on the asterisk console. Can i sendtext() them to the
caller or put them in cdr?
Regards, Andreas.
I'm also interested in this. If you find solution, please mail it to the list.
--
Tomislav Parcina
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
E.g: because you have a valid PID file of the controlling process. If
you actually want to kill it, you can.
And you don't need physical access to the system to get to the one and
only real console. OTOH, if you do have physical access,
Why there is no asterisk.conf.sample file?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
winmail.dat___
--Bandwidth
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I upgraded to the newest 1.2 Zaptel release and this is still occurring. I
checked and the digium card is not sharing an IRQ with any other devices.
I also changed busycount=8, and set callprogress=no.
The call drops are still
Is uniqueid globally unique? I have three Asterisk installations and I need to
store data from all of them in same database, in same table. Will this uniqueid
field be unique?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP
Hi list!
How to make outgoing call thru other mISDN channel group of all channels on
first group are busy?
I believe I'll need to
- Check of there is free channel on group1
- if there is free channel call thru group1
- if there are no free channels call thru group2
--
Tomislav Parčina
Lama
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Iirc you can put more than 1 interface in a group and it should just use
any free channel of whichever interface that has a free channel. Check
the sample config.
Hi Patrick!
Yes, I know that and I'm using that. But then I need to change
for Notify User 28
Why do I get == Spawn extension (sip2, **20, 1) exited non-zero on
'SIP/27-b65a1100'
I have to pickup either 2X, 8X, t or uevents extension (phone will ring on any
of those).
Have I done something wrong?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I worked with Cisco and HP and they should do what you are looking for.
I even worked with cheap unmanaged switches ~20 Euro and they work with
VoIP.
Do you know for switch that can tell me that on port 7 there are two active SIP
calls.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad
(gigaset.siemens.com).
C450IP costs less than 100 USD (in Italy at least), S450 is slightly
more expensive.
I have Siemens C450 IP for two days and it seams weary good.
I'm
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
What do you mean by mapping the 200 ?
In this example I can pickup any ringing extension:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
If phone with number 42 rings you can catch the call by dialing 742. You
don't need to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I just want out find out how to do bill recon's when you send calls to a
provider. They send me
their CDR's, and when I compare it to my * CDR's, some calls are 1 second
off, either way.
How in general is it done by others?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other sort of
faxing, the endpoint must support T.38 and you must
Is it possible to use Digium (or Sagnoma, or Beronet) cards with Asterisk on
Vmware?
Has anyone done it?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
CLI sip show registry
HostUsername Refresh State
iinettrunk:5060 [EMAIL PROTECTED] 3584 Request Sent
sip.pennytel.com:5060 N 280 Registered
Yes, I have
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You then ask the telco to include Advice of Charge (AOC) in your ISDN setup.
The AOC then is included somewhere in the Asterisk CDR, but I don't have
direct experience of this. You can then get appropriate software to issue
bills to
which supports only h264 codec. Right now I
can make only direct IP video phone calls, and I would like to make calls true
Asterisk.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http
tell me more about this? You mean when call from SIP goes to FXO port,
if phone attached on FXO port answers after the first ring (before second) ATA
will always stop to work?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
1. you need qualify set as the wifi radio on the phone sucks big oranges
What is qualify set?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Have you tried setting timeout, attempts and rotate in resolv.conf?
Can you please tell me more about this? How to do it and what would I achieve
with that?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Actually you need to use the SVN version of Asterisk to support H264
video. It should be part of the planned 1.4 release.
When can I expect 1.4 release? Will it be this year? First quarter of 2008?
--
Tomislav Parčina
Lama Computers
of the following:
- my screensaver
- picture of calling person
- External directory
- dialplan.xml
- How to setup hinting (Multiple Call Appearance)
- How to login true ssh?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e
Do Asterisk team care about this anymore?
Whole text can be found here:
http://www.asterisk.org/developers/releasecycle
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
qualify=yes
Put in in the sip.conf file in the configuration section for the
specific phones.
I don't think he thought on that.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
, thank you for info.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
___
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in
SVN trunk? Right now I know only for H264, where can I find the list of others?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Err, wasn't the patch for H.264 just changing one digit for another?
Hi Thomas,
I don't know. I should check BUG page for that.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Other cool things:
make menuconfig
Jingle/jabber support
IAX2 media transfers
new sound files
New answer machine detection (AMD)
and much much more!
Hi Matt, thank you for info!
Bye.
--
Tomislav Parčina
Lama Computers Split
to see hinting on 7940/7960.
Can you send me your's Phone Directory xml files? I can't manage to add
second page so I have only 32 numbers :(( Also, I can't manage to enable search
thru directory.
Other thing, can personal directory be in xml file?
--
Tomislav Parčina
Lama Computers Split
Stinice
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any
info?
What version should I download? Is this one all right?
cmterm-7940-7960-8.4.00-sip.cop.sgn
Signed SIP Firmware for CCM versions 5.0(4) and later
--
Tomislav
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my
phone and now it doesn't register with Asterisk. In full.log file I don't see
any reason why phone doesn't register.
Has anybody head problems like this one?
--
Tomislav Parčina
Lama Computers Split
Stinice 12
this one?
Now I have downgrade to 8.0.2 version and phone has registered fine.
Does anybody know what is the problem with SIP 8.0.4 firmware and how to solve
it?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail
hears dead air.
Hi Marty,
I can live with that. I don't have anything connected to FXS port :)
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
and I have that pages on http server and in log I see that phone
asks for them...
Have you done anything in Java for this phone?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http
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