let's suppose I have this dialplan :
exten = _X.,1,Playtones(ring)
exten = _X.,2,Dial(CAPI/contr1/${EXTEN},,g)
exten = _X.,3,AGI(update)
where update updates some db tables we have based on the type of extension
Now, from the wiki :
If the /g/ option is specified, and the called party hangs
for the particular configuration of software/hardware that connects to
my asterisk pstn gateway I need to do something like the following :
[...]
exten = _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf))
[...]
[macro-senddtmf]
exten = s,1,SendDTMF(*)
but the DTMF must be sended to the caller
Kamran Ahmad ha scritto:
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
Look at the examples you find at www.onsip.org, they are really well
explained.
log every step taken with something like log(2,now I'm doing
I've the following setup :
sip phone - ser (auth and routing) - asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones(SIP/2.7.184.61-08152880, dial) in new
I've this setup :
CiscoAta186 - asterisk with oh323 chan - gsmgateway
dtmf doesn't work, tryed inband, with g711a and g729 codecs
CiscoAta186 - gsmgateway works, even with g729, so it seems the problem
is in *
oh323.conf has inBandDTMF=yes, what else may I need to tweak ?
we got this installation :
WinSip(demo version) - ser(radius accounting) - asterisk(from sip to
h323 channel) - gsm gateway(with 32 sims in it)
we configured winsip to make 28 calls like from 28 different sip
accounts, to 28 different cellular phones numbers
after the first ten :
--
ok, they let me know I'm an idiot, maybe
outboundMax=10
has something to do with it
after the first ten :
-- Executing Dial(SIP/5060-081925b0,
OH323/[EMAIL PROTECTED]) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec(s) alaw
-- Called [EMAIL PROTECTED]
we get :
/Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users
lists ?
Simone Cittadini
IT Manager
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servers do a lot of noise, consider it if you
aren't putting the server in a dedicated room, really ... I have two of
them in the corridor out of my office, they drive me insane ...
Simone Cittadini
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Tel
exten = 555,1,MusicOnHold(default)
i can hear the music, so far so good.
But when i hold an incoming call by pressing the HOLD-key on my snom
telephone - nothing happens.
No output at CLI that the MOH gets played.
When debugging SIP on asterisk, in the moment i press the HOLD-key i can
I have an asterisk box connected to two isdn lines via an AVM c2 card,
the ISDN boxes have the 0227006XXX and 0227007XXX numbers, and are
configured both p2p, with the first one as file-leader.
(I don't know if file-leader is the correct term, it's a literal
translation from the italian term
Il giorno mer, 22/06/2005 alle 07.39 -0400, Dean Collins ha scritto:
As an asterisk server it is more than fine but asterisk prefers to be a
standalone machine.
You would have a lot less issues if you had 2 machines, one handling
file serving, SMTP and one dedicate machine for asterisk.
I have an asterisk installation connected to 2 isdn lines via an AVM C2
card.
modules seems to load well, lsmod gives :
c4 19588 4
b1 24192 1 c4
capidrv28468 2
isdn 134604 9 capidrv
slhc7552 1 isdn
-
From: Simone [mailto:[EMAIL PROTECTED]
Sent: 10 June 2005 10:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity
I understand what you're saying, but I am not the one who makes the
decisions. That decision is made already, so since I
My fault. I understand my terminology was not accurate. Thanks for your
reply.
Simone
Steve Hanselman wrote:
With call manager V4 and above it's extremely easy, just connect a SIP trunk to
*.
BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX so your terminology
to implement Asterisk in these ones, but if it cannot be connected
to Cisco this won't be an option at all, they won't consider it.
So, back to the question, is it possible to connect Asterisk to Cisco
and have all the functionality expected, and is it hard?
Thanks, have a nice day
Simone
William
the
extensions and directly reach the other office).
Thanks, have a nice day
Simone
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Hi, just wondering if my question is just unusual or if it is a quite
stupid one. Thought there would be someone having this kind of scenario,
but maybe I'm wrong.
btw, have a nice day
Simone
Simone wrote:
Hi all, first post. My company's office in the UK is soon going to get
a Cisco VoIP
the
extensions and directly reach the other office).
Thanks, have a nice day
Simone
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mk111 wrote:
I was
told that the phone should be able to download the SIP... file once the
TFTP address was changed. So far nothing though. Any ideas?
have you rebooted the phone after changing the tftp address ?
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Simone Cittadini wrote:
I have asterisk 1.0.6 with cisco 7912/7960 phones (sip) and a isdn card
with capi drivers, everything works fine, except for music on hold, even
when you transfer a call (which is the most annoying part, since the
caller thinks the line is down and hangups
that's all
--
Simone Cittadini
IT Manager
==
COMVERT S.R.L.
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20126 Milano - ITALY
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Asterisk-Users
)?
TIA,
Simone.
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/lib/libz.so.1 (0x00ee8000)
libm.so.6 = /lib/tls/libm.so.6 (0x0028a000)
libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x00c88000)
Cheers,
Simone.
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,
Simone.
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Anyone knows if there's a way to select the bitrate of those codecs
supporting multiple bitrates (eg. g.726)? I've tried searching and
googling a lot, but without useful results...
Cheers,
Simone.
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.
Cheers,
Simone.
Jon Bebeau ha scritto:
Seth,
Are you using SpanDSP to receive faxes ??? I'm unable to get SpanDSP to
receive correctly and others I'm talking to (off list) are too. Can you
list the set of sources your using (libtiff and audio) for us?
Thanks
Jon
wish...tell me which files you need exactly. But beware, worst things
may happen (like strange segmentation faults).
Cheers,
Simone.
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not sure if it
will work.
I really can't remember, just googled a bit. I'll send you. If someone
else needs it, drop me a line. I'll upload the whole kit somewhere,
eventually.
Cheers,
Simone
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of which were successful.
Hope this helps.
Cheers,
Simone.
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Anyone knows where can I find spandsp? Official site seems permanently
down...
TIA,
Simone.
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. Please read it carefully before posting in
mailing list, thanks.
Cheers,
Simone.
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), sms_app starts but
after a while phone says transmission failed (or some other error
message). I've tried with different phones brand/type, all claiming to
be ETSI compliant (and they are, since sending SMSes through my telco's
gw works like a charm).
Cheers,
Simone.
Roland Zagler ha scritto
Tried, doesn't work. And onestly, I've not catched this. Where's the
difference?
Cheers,
Simone.
administrator tootai ha scritto:
Simone Ricci a écrit :
I've found app_sms which is supposed to do that. However, I never
managed to get it work. Every phone I tried refuses to communicate
Surely, 802.1Q wasn't designed with security in mind...change tagging,
change vlan...
Cheers,
Simone.
Steve Szmidt ha scritto:
Thus the Virtual part of VLAN. Though it's still a very good idea, from a
security standpoint, to keep them apart. You do not want to have your LAN
owned because
on which ports. Can the
phones be confifured to only allow untagged traffic on the port towards
the pc?
Cheers,
Simone.
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possible to do this?
I've ever tried splitting 'peer' and 'user' part in sip.conf, but that
seems to give weird results...
Someone has ever done something similar (or maybe knows how to?)
TIA,
Simone.
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That's not true: with some equipment you can use VLAN tagging to
separate VLANs. This allows to have multiple vlan's running on the same
wire.
Cheers,
Simone.
Richard Cook ha scritto:
I think the concept behind that is to have your voice on a separate VLAN
then your data. In this case
a Playtones(), waiting for user entering an
extension.
I've tried many solutions, played around with all dtmf options...but no
luck...
Note that after the first digit (which get 'lost'), others get processed
successfully.
Cheers,
Simone.
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You could use D's option of the dial() command.
Something like Dial(SIP/xx,,D(C)). But check the documentation.
Cheers,
Simone.
Antonio Rabena ha scritto:
Hi,
How can i send dtmf tone upon connection?
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Resolved with the help of a gentleman in chat :)
BTW, i was lacking an Answer() in the dialplan.
Cheers,
Simone.
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Hi,
when I make a call, chan_capi always uses controller 2, and never uses
controller 1 (so I have 4 lines for incoming calls, but only 2 lines
instead of 4 for outgoing calls).
this is with 2 AVM Fritz cards PCI.
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BK s.r.l. - Brain and Knowledge
first controller?
Thank you...
Il gio, 2003-09-04 alle 11:40, Jamie Neil ha scritto:
Simone Vasoli (BK s.r.l.) wrote:
Hi,
when I make a call, chan_capi always uses controller 2, and never uses
controller 1 (so I have 4 lines for incoming calls, but only 2 lines
instead of 4 for outgoing
, Jamie Neil napsal:
Simone Vasoli (BK s.r.l.) wrote:
Hi,
when I make a call, chan_capi always uses controller 2, and never uses
controller 1 (so I have 4 lines for incoming calls, but only 2 lines
instead of 4 for outgoing calls).
this is with 2 AVM Fritz cards PCI.
You can
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