ter 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.
--
_______
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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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 separa
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 of the
I don't really think so. They only can tag traffic they generate, AFAIK.
My ATA-186 from cisco has that functionality, but never tried myself.
Peter Svensson ha scritto:
I have not used any ip-phones with vlan support but several switches. They
all allow you to confgiure which tags are accepted o
I've got a problem with DTMF, again.
My asterisk box is connected with the outside world (PSTN) via a sip
proxy. The problem is that for some reason, I need to use rfc2833 for
signaling digits to the gateway and inband to accept digits from outside
(eg. when someone dials one of our DIDs). It's pos
Anyone knows where can I find spandsp? Official site seems permanently
down...
TIA,
Simone.
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Hello All,
Currently my setup uses Xlite and Asterisk and i found that all the RTP
voice packets are transfered via the asterisk server from one xlite to
another. Is there any possibility that we can make all the RTP Packets to be
transfered directly between the two clients once the connection is
e
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 with
asterisk.
This was my (very basic)config:
exten => 1,1,Answer
exten => 1,2,Wait(1)
exten => 1,3,SMS(test,as)
exten => 1,4,HangUp
This is supposed to answer th
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
FWI, I'm using it very successfully. Where's your problem?
My system is running Fedora Core 2, libtiff and libtiff-devel as
provided by distro (installed with apt-get, version 3.5.7). I've got to
add some headers in /usr/include to get it work: tiffiop.h, tif_dir.h
and ports.h
Hope this helps.
William Glynn ha scritto:
I'm seeing training errors that often stop fax machines from even
starting delivery, lots of errors from libtiff regarding unexpected
line lengths, and garbled data even if RxFax thinks it's okay.
Strange. Which version of libtiff are you using?
However, I've gotten TxFax
administrator tootai ha scritto:
>
> From where you got port.h? I install tiff-3.5.7 (I'm running an RH73
> with those rpms installed) and didn't manage to create this file.
> tiffiop.h and tif_dir.h are parts of tar.gz package. I had to modify
> tiffiop.h and replace port.h with tiffcomp.h but I'
Jon Bebeau ha scritto:
1) Where did you get the C headers from (what dir).
tiffiop.h and tif_dir.h somewhere in the libtiff's 3.5.7 source .tar.gz
port.h with google's help. I can send my version to you if you wish. It
worked for me.
2) What HW are you using (processor, motherboard)
Intel(R) Cele
William Glynn ha scritto:
I'm using Gentoo's tiff-3.5.7-r1 package, which appears to be built
from virgin (unpatched) 3.5.7 sources.
So it must be ok.
Strange, again. I've not tried loopback, but my incoming/outgoing faxing
works like a charm, via SIP. I don't have any pstn hardware, my pstn
inte
administrator tootai ha scritto:
I just finish installation and give it a try on reception: same problems
than related by William. Seems that around 20 first kb of data's are
approximatively ok and the garbage start (unexpected line lengths)
[ ... ]
I use an X100P. SMP (2 celeron 400) 440 Mb RAM
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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I've got an old HFC-S card to play with, and I would like to use it in
NT mode. I've a problem only: wiring. I can't fully understand the
instructions I was able to find online. Someone can point me to a site
which explains the whole procedure clearly (like with some schematics,
even in ASCII)?
Hi,
I'm experiencing a strange issue with static realtime. Seems that some
values belonging to 'general' category (like, for example, rtptimeout,
rtpholdtimeout, realm) are ignored. Running asterisk 1.2.1, I've tried
both res_odbc and res_mysql (that one from asterisk-addons tarball)
without luck.
Hi,
I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as
sip-to-pstn GW. The issue is that when a call comes in from the pstn,
asterisk correctly contacts the router, which in turns send a "183
Session progress". Obviously, asterisk thinks that the telephone is not
ringing (because it
Hi,
what's the current status of jb implementation in chan_sip? Are there
any patches out there available to be applied to the brand new 1.2-release?
Cheers,
Simone.
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Hi,
I recently purchased a brand new 7905g with his SIP firmware (licensed).
Now, I want to play a little with chan-sccp, but I'm unable to find the
appropriate firmware for my phone. I know that I must get it directly
from cisco, but before purchasing it will be very good for me to try a
bit; does
Adi Simon ha scritto:
> Hi,
>
> Did anyone actually manage setting up a single SER with multiple
> Asterisk boxes?
> I particulary have a problem of keeping the session alive and by that I
> mean directing
> all the following sip messages to the same asterisk box the first signal
> was sent (rand
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