Hi all,
I have a PRI, and when the Internet connection goes out so do my
phones. I suspect it is some type of DNS issue. I do have a SIP
trunk, and it appears that if I lose DNS to the SIP trunk, the entire
PBX is offline. I have no actual proof of any of this, and have not
done any
Hi,
I have 1 zap channel in my house shared among couple people.
If someone dials 911, I want that zap channel to be disconnected
right away to make way for the 911 call.
I dug through voip-info.org and didn't find much.
Any hints?
try looking into SoftHangup()
Hi,
We have an issue where Polycom's lose BLF functionality after a reboot. The
only way to fix it is to reboot the Polycoms.
Anyone else have this issue? We are using 1.4.18.
If I run 'sip show subscriptions' all the subscriptions come back after the
restart but the lights on the phones
Hi
Hi Everybody, I'm having a little problem with asterisk CLI, after the
version 1.4.19 I'm not been able to see the CLI with colors anymore. I
have a ubuntu box with asterisk 1.4.21 installed and I don't know how to
enable the colors again. Of course I have the variable $TERM set to
Are you of that ?
I'm not 100% certain, but I think Thomson phones wouldn't query centralized
directory for outbound calls.
I think centralized directory is only queried when using Directory key.
well, our customers use it by opening phonebook and selecting
number to call ... so don't know
Hi,
[snip[
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
you can set callerid per-peer in sip.conf
like:
callerid='Jhon Doe' 1234
this should work
[snip]
To make myself clear, here is what I'm trying to do : when Alice is calling
Bob (Alice --- Asterisk---Bob), I would like Bob's phone to
display Alice's name (no problem, for that) but I would also like Alice's
phone screen to display Bob's name (instead of Bob's number)
mmm
(Even with the missing '}' at the end that I fixed
oops, always forget about those :'(
btw, in our installations we provide a centralized xml phonebook
(and phones that support it, mostly grandstream and thomson), and
they (the phones) autmatically set the called name on the display,
both for
Hi all,
after a little bit of googleing, seems that correct syntax is:
exten = _123X, 1,
SIPAddHeader(P-Asserted-Identity:${CALLERID(name)}
sip:${CALLERID(num)})
(notice the sip:${CALLERID(num)})
but, IIUC, this sets the header for *outbound* call to 123X number, so
don't know if CALLER
can see
Maybe a bit silly question, but why doesn't Asterisk accept if you set
both a usernamepassword as well as an ip address for a phone?
but it does accept!
in a peer definition:
[user]
type=user (or better friend)
username=user
secret=secret
host=10.0.0.1
[snip]
It's obvious that the more
Hi,
[snip]
For example I tried to block registrations from other subnets as
follows:
[general]
...
deny=0.0.0.0/0.0.0.0 ;deny all by default?
permit=10.1.0.0/255.255.0.0 ;allow registrations from local
subnet?
you should put deny/permit PER peer as
[200]
Hi,
There is a question about the fxo of the zaptel card which is set a
number to use as common analog phone. When I use ${CALLERID(num)}to get it's
number, it could'n be done. But ${CALLERID(num)} could get the other number
of the SIP or IAX . Could you tell me the reason, and how I could
Hi,
I have following settings done on my Fedora8:
Downloaded
openh323-v1_19_0_1-src-tar.gz
pwlib-v1_11_1-src.tar.gz
to my knowledfe chan_h323 should be compiled against
openh323-v1_18_0-src.tar.gz
and
pwlib-v1_10_3-src-tar.gz
cheers
--
P.s. I tried to use an Openvox B800P instead of the Digium B410P and it
worked immediately, without any problem or modification to the default
configuration files. Maybe Digium card is fisically damaged?
mmm... i have an installation with two OpenVOX B400P and misdn-1.1.7 connected
to a
Hi all,
[snip]
For this reason I set all the 4 ports of Digium's card in NT mode
(Philips can not do this). Then i opportunely
edited /etc/misdn-init.conf and /etc/asterisk/misdn.conf. In fact, when
I run the command misdn shows stacks in * CLI, I can see all ports in
NT (PTP) mode.
have
Hello all!
Hello!
I'm having problem with the calls that come through my asterisk box
and back out to our legacy pbx, it seems to be that even if the call
is ringing and not picked up yet, zap reports the line as answered,
why is it doing that?
could be that the PBX *answers* the line
Hi,
[2008-06-20 15:38:19] WARNING[9772]: channel.c:4347 ast_get_group: Ignoring
invalid group 5863165 (maximum group is 63)
we had a similar error, found somewhere (voip-info.org?) this solution:
exten = _**2XX,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _**2XX,n,PickUp(${EXTEN:2})
where
On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote:
I don't know if you intended to be rude with the normal
human comment but it sure seems like it when reading your
reply. Also how many users know they can dial ** to get a +?
Especially when so many cannot as said earlier be relied upon
to
On 8/7/07, Vieri [EMAIL PROTECTED] wrote:
--- Mr Shunz [EMAIL PROTECTED] wrote:
Caller ID Scheme as
ETSI-FSK Prior to Ringing with DTAS...
Thank you Daniele.
That seems to work.
I tested it on analog phones without a display.
Yeah, we had them without display too ...
But, if i
On 8/6/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
I have an 8-port Grandstream GXW-4008 V1.2A ATA
converter with analog phones connected to it.
Hi,
we hava a GXW-4004 but i think it has the same sw ...
They work fine except for just one feature I would
like to modify. Somehow, each time the
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