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Viva,
Witch trunk are you able to establish ?
SIP trunk beetwen asterisk and PBX ?
Oh323 ?
João
Manuel Silva wrote:
| Hello!
|
| I have installed 2 servers, one with SER integrated with PostgreSQL
| (Fedora Core 3) and the other with Asteris
" and then dies, withou any error
message, nothing at all.
But, if i start asterisk with cli console (-vvc) theres is no
problem and i can create the conference rooms.
Zaptel and ztdummy modules are already loaded.
Anyone with the same problem ?
Asterisk: 1.0.8
Kernel: 2.6.9-5.0.3.ELsmp
CCM do you have ?
Is it CCM ou CCM Express ?
If it is CCM > 4.0 then i advise you to use a SIP trunk between CCM and *.
Create your mailbox(s) in voicemail.conf, make your dialplan and route
the calls to * and it's done ;)
Rgds
João Amaro
| __ Do you Yaho
s this
a CCM feature ?
All the phones used are cisco (skinny) attached to CCM.
The only thing that appears on logs is:
file.c:550 in ast_readaudio_callback: Failed to write frame.
Thanks in advance
Regards
João Amaro
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Hi.
If you show what's in the .call file, it would help.
Rgs
Joao
Gilbert Abboud wrote:
| hi
|
| I created a .call file as mentioned in the WiKi but when i place it
| in /var/spool/asterisk/outgoing, the Asterisk console shows
| "unknown keyw
r CCM without
problems till now.
João Amaro
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Asterisk-Users mailing
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Hi
How do you queue the incoming call ?
Do you queue the call with the t option (allow the called user to
transfer the calling user) ?
Regards
João Amaro
Anton Krall wrote:
| Guys.. Why is it that when a call comes to a call queue and in
way ?
"Press 1 for Susan",
"Press 2 for David",
"Press 3 for Sam the Dog",
"Press 4 for Nemo the Little Fish",
"Press 5 to leave a message",
"Press 6 to Hangup".
rgds
João Amaro
| thanks David
|
|
|
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Hi Richard,
Your webserver:
~ Server Error: 403 Forbidden
-
--
~ Access denied
-
--
~ /doc/index.htm
S
y (via ccm).
Thanks in advance
João Amaro
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Asterisk-Use
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Ben Merrills wrote:
| There's a few (open source/free) ones in development. I myself am
| developing one of them.
|
| Ben
|
Hi.
Why not join all the project in just one ?
Actually which queue log analyzers projects are beeing developed ?
Check the mail
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Hello
You can use H323 to connect to Cisco CallManager.
Add asterisk as an h323 gateway on cisco callmanager.
Then you can send & receive call from asterisk.
TIP: Use OH323 instead off asterisk h323 native driver.
Regards
João Amaro
W
k-oh323-0.6.5/openh323_1.13.5-make.patch
#./configure && make clean && make opt &&
make install && ldconfig
Asterisk
#cd asterisk-1.0.3
#make && make install && make samples
Asterisk-oh323
#cd asterisk-oh323-0.6.5
Edit the Makefile
#make &
openh323 before compiling.
Hope it helps
Regards
João Amaro
Kanuri, Seshu (Company IT) wrote:
Joao wrote:
Meanwhile i've downloaded ,again, the 0.6.5 version.
I'm using pwlib and openh323 versions from sourceforge.
It compiled without errors, but the error
eed to do downgrade asterisk & chan_oh323 versions.
João Amaro
Tenorio, Leandro wrote:
I got it, but email it to the list is not a good option.
Who 're interested just email me, I'll send it asap.
But AFAIK, you still need the wrapper.
LTenorio
-Original Message-
From: [EM
ssnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?
Regards,
Silviu
____
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de João Amaro
Envoyé : mercredi 5 janvier 2005 16:38
À : Asterisk Users Mailing List
ssnetworks.com). What is the error message you get when you try to
compile asterisk-oh 0.6.5?
Regards,
Silviu
____
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de João Amaro
Envoyé : mercredi 5 janvier 2005 16:38
À : Asterisk Users Mailing List
failed!
I've tried to upgrade to version 0.6.5, but i got a compile error.
Anyone know how to solve this error ?
Thanks in advance, and have a GOOD 2005
Regardz,
João Amaro
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htt
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Hi
I'm using Fedora RC1 without problems (kernel 2.4.22)
Anyone here have tried Whitebox Respin 1 with asterisk ?
Maybe nest week i'll install a small Asterisk Server on it.
Asterisk + Apache only.
Bill Bradford wrote:
| I'm in the process of buildin
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Kevin P. Fleming wrote:
| João Amaro wrote:
|
|> I'm using 14 cisco 7940 as Dynamic queue agents.
|>
|> They use the pixel based screen, to login/logout from queues.
|> They can also see the queues stats.
|
|
| Now that's real
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Paul Crick wrote:
| Hey Jay
|
| All the stuff you've described is possible. I've done some playing
| with the XML services on a Cisco 7960 to give ACD queue stats and
| system uptime info. The phone has a mini web browser built in so
| it's pretty easy
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Hi List,
I'm using asterisk (1.0-RC2) as a callcenter, with IVR, for cisco
callmanager (via H323).
When the caller calls the callcenter, he will be prompted to press 1
or 2. For QUEUE1 and QUEU2.
While waiting i've musiconhold playing . All the agents
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Hi All,
I putting two * boxes into production. It's a callcenter + voicemail
to Cisco callmanager.
My problem is that mpeg123 sometimes doesn't terminate.
What should i do ? Don't use MusincOnHold, and use a single MP3 file
with a high length ?
-B
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I Vlasis,
I'm using those versions (Fedora COre 1) and it compiled without
problems, but when i try to initialize asterisk i get the folowwing error:
ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener
creation failed.
Hope someone can hel
driver when i start the asterisk.
Thanks ind advance,
Regards,
João Amaro
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Hi List,
I'm using the apllication AgentCallBackLogin so agents can login to a
queue. They just need to
enter the password, the CallBack Extensions is the ${CALLERIDNUM}
Is there a way to AgentsLogOff withou using the AgentCallBackLogin
application. I d
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