try with
type=peer
good luck
Edgar
On Tue, 2005-04-26 at 13:08 +0200, Alessio Focardi wrote:
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
This is my sip trunk configuration
canreinvite=yes
context=from-internal
disallow=g729
host=192.168.1.138
mask=255.255.255.255
qualify=yes
type=peer
maybe it could be the context!
HTH
Edgar
On Thu, 2005-04-21 at 12:57 +0800, Dinesh wrote:
Great:)
Just one question, I am trying to get the
Hello i need to make a central phone book, at this time we got a lot of
offices far away from here and i want to know if it possible to get a
central phone book from ldap or mysql to make calls just typeing the
name of the office, i saw the macros extensions using ldap but just to
get the caller
Hello, i want to be able to use my zultys softphone to make calls pc-tp-pc
and pc-to-phone, from my home, i want to install an asterisk server but at
this time i need to connect to a voip service provider, can anybody tell
my wich provider are the best and got good rates???
TIA
Edgar
Hello, yesterday when i wasnt in the office the asterisk server stop
working, it was registering the sip terminals but cant make calls, because
im not in the office i told the people to reboot the server to make the
server works again but today i found this lines in the full log, can
anybody tell
Hello Mark, i tried to get the spanish soun but get
You don't have permission to access /VoIP/AsteriskSounds_ES.tar.gz on this
server.
can you help me???
TIA
Edgar
Hi Derek,
Yes there is. Take a look at my web pages
http://www.g7ltt.com/VoIP/vmfiles.html. You'll see that I started a
be transfered with # transfer, even
though the queue command has the tT options...
Hope this helps
Guido
-Ursprüngliche Nachricht-
Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 3. Februar 2005 15:34
An: Asterisk Users Mailing List - Non-Commercial Discussion
Hello, anyone knows if exist the audio files in spanish??
or how can i record the voice in gsm extension???
can i play for some announce a random file??
TIA
Edgar
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Thanks for your help, here are my config for the queue,
agents.conf
[agents]
musiconhold = random
autologoff=15
wrapuptime=5000
ackcall=yes
group=1
agent = 1001,3101,Edgar de Leon
agent = 1002,,Jorge Cabrera
agent = 1003,,Nati del Pozo
agent = 1004,,Emilio Perez
agent = 1005,
with queues.
Hope this helps a bit more...
Guido Hecken
-Ursprüngliche Nachricht-
Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 3. Februar 2005 09:08
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: RE: [Asterisk-Users] howto answer a call
,Queue(esculapio|tT|||300)
exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody
answers the call
exten = 76522,4,Hangup
I had similiar problem in working with queues.
Hope this helps a bit more...
Guido Hecken
-Ursprüngliche Nachricht-
Von: Edgar de Leon [mailto
Hecken
-Ursprüngliche Nachricht-
Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 3. Februar 2005 09:08
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: RE: [Asterisk-Users] howto answer a call in a queue
Thanks for your help, here
similiar problem in working with queues.
Hope this helps a bit more...
Guido Hecken
-Ursprüngliche Nachricht-
Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 3. Februar 2005 09:08
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: RE
hello i need to know how to enable the feature in the agents.conf to make
the users got to press # to answer the call when is in the queue and the
agent is logged in.
at this time the call enters the queue and the agents who is logged in
only beeps once and then the call enters automatically.
Thanks for your answer, i got ackcall=yes but the call when enters only
ring once in the agent phone and connect directly,
agents.conf
[agents]
autologoff=15
wrapuptime=5000
ackcall=yes
group=1
agent = 1001,3101,Edgar de Leon
agent = 1002,,Jorge Cabrera
agent = 1003,,Nati del Pozo
Hello, i got configured the queues.conf and agents.conf and works well in
the first configuration for testing purposes i used
[agents]
autologoff=15
wrapuptime=5000
ackcall=no
group=1
agent = 1001,3101,Edgar de Leon
agent = 1002,,Jorge Cabrera
and when i loged in, plays a musiconhold
Hello, i got a question,
i need to create a group extension, to make calls to 6 sw phones, but i
need to know if asterisk can do help me to get a unique number and check
what extension has received less calls than the others, and pass the new
call. We got a call center and want to know if we can
Hello everybody, i got and asterisk and a CCM configured thru SIP, and in
the sip show peers appear
Name/usernameHostDyn Nat ACL Mask Port Status
CCM 10.60.27.138255.255.255.255 5060 OK
(1 ms)
but when i enabled sip debug in the
Hello Phil im from Guatemala, im living in Madrid but im thinking in came
back in july, if its helps to you, im thinking in make an installation of
asterisk to make calls, if you found something now to make calls please
inform me!
TIA
Edgar
I'm looking for a company that offers Guatemala
Hello im triying to config xlite on wine for linux, but got problems with
the mic test, can anybody tell me how to get the mic config to work with
wine or x-lite?
TIA
Edgar
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Thanks, its a better solution!!!
Edgar
Hi Edgar,
Don't use XLite under wine, use native code for Linux:
http://sipthat.com/archives/000187.html
Cheers
Edgar de Leon escreveu:
Hello im triying to config xlite on wine for linux, but got problems
with
the mic test, can anybody tell me
i apreciatte if u can send me the conf files, and the screenshots about
the CM config, its really easy as you said, i like asterisk very much,
after that we are planning to make test on echo and relay calls, but think
it would work great, thanx for your help,
Edgar
You need to create a SIP
Anyone???
Mensaje original
Asunto: [Asterisk-Users] Callmanager 4.1 and asterisk
De: Edgar de Leon [EMAIL PROTECTED]
Fecha: Mar, 28 de Diciembre de 2004, 8:21 am
Para: asterisk-users@lists.digium.com
Hello everybody,
im newbie in VoIP, but find this project asterisk very interesting, i
tried to install and its a great sw, i really get sorprised about all of
its functions, we need to use the asterisk server in conjunction with
cisco callmanager.
We have a Cisco Callmanager 4.1 and the clients
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