On 2006-11-06 13:16:50 -0800, Christian [EMAIL PROTECTED] said:
Hi all,
DO my messages come through to the list? I have had some problems wiht
my email client here.
Looks like your spell checker has issues also...
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Hello all,
just curious if anyone's successfully compiled (*) with the latest
FreeTDS code/driver. The Makefile in (*) seems to only take care of
0.63 or older. I tried to muck around with it a bit into tricking to
compile for not just 0.63 but anything later than 0.62 but it seems to
crap out
On Monday 06 November 2006 16:41, Matt wrote:
This should work.. please make sure you have qualify=yes on in
your sip.conf file for each of your sip entries.
Now it works. Thank you!
On 11/6/06, Dmitry Ivanov [EMAIL PROTECTED] wrote:
Hello!
I have this in my dialplan:
Hi guys. I just bought and configured a Snom 360 and have noticed that the
LCD is constantly flickering at a rate of ~10-15Hz (that's a guess).
Either way, it's very distracting. Has anyone else encountered this
before? Any solutions?
Cheers,
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F:
Hi All,
I have installed Asterisk Successfully and configure a out bound trunk for
another SIP server so that if Ill dial 777123 from an asterisk-registered-phone
then it will dial to the phone extension(123)-registered in the third party
server.
But my problem is that the reverse is
Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
I am trying to do something that I see describe in a book and it is not
working
In my sip.conf, I have in my [fxo] context=from-pstn
I then have in extensions.conf
[from-pstn]
exten s,1,answer()
exten
Hi,
I'm currently running an * version 1.2.13 and sox version 12.17.5. I want to
upgrade sox to the newest release ( 12.18.2 ); need mp3 support.
But how do I make the upgrade.
Do I need to recompile asterisk afterwards?
If I make a sox -h after a reboot I can see the new version is running
Hi All
I am not sure what I wish to do it possible but I would like
to see if you guys know any better.
I have a site who has the extensions: 1231, 1232. 1233, 1234
Each of these users can dial each other on the extension
number an also has an external CLI mapped to them.
On all
Your offnet calls will be more than 4 digits, so use that to ur advantage.
so, for internal calls,
exten = _,1,Set(CALLERID(all)=Name ${CALLERID(num)})
or if u dont want to change the CLID at all.. dont do anything..
exten = _,1,NoOp(nothing)
else, for all external calls(4 digits)
exten
On Tue, Nov 07, 2006 at 11:14:43AM +0100, René Christensen wrote:
Hi,
I'm currently running an * version 1.2.13 and sox version 12.17.5. I want
to upgrade sox to the newest release ( 12.18.2 ); need mp3 support.
But how do I make the upgrade.
Do I need to recompile asterisk afterwards?
Hi!Then please just tell me if it's even possible, as i cannot find any configuration to allow unknown codecs to be used in reinvited calls
.My question is that is it possible or impossible to handle this with asterisk?Thanks!AndrásOn 11/5/06, Szabó András
[EMAIL PROTECTED] wrote:Hi!I want to
Hello
I have set the time out to 30s in queue.conf, but
my agent has called 2 times, and the next extension(the Hangup)
is called after 60s.
do you think that it is normal?
On asterisk console i have these messages:
-- Executing Queue(SIP/1-0cf8, queue|tn) in new stack
-- Started
Yes, Telefonica is able to do PRI but just in a very restrictive area.
RR Libera
Ilan Rabinovitch escribió:
We briefly used it with iPlan, but found that there were some problems
with the stock asterisk implementation and Argentina variation of R2.
We ended up convincing iPlan to switch us to
I have the following
setup in my test lab (which reflects very much my production installation, just
on a smaller scale)
Asterisk server
- Internet -- Home router (Linksys) ---Hub
Polycom 501 (Phone A)
|---
Polycom 501
What settings are you using when you call the queue? Ie in the
extensions.conf I have the below, this keeps calls in the queue for 30mins
(1800 secs). If you adjust it to 30, the call will come out of the queue in
30secs and move onto the next dialplan.
exten = 8000,3,Queue(fservices1800)
I was wondering whether you have canreinvite=yes on those
phones, and that the audio between the phones is working, but not between the
Asterisk server and the phones - perhaps an Ethereal trace from your Hub might
help?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I have always been happy with Snapgear units. Most also have hardware based Encryption acceleration for IPSec and PPTP VPNs. As for setting up with SIP, use a VPN, or call their tech support line and they'll help you. Each unit comes with support for setup built-in last I
I _had_ canreinvite=yes, before I read your post. My
production environement though cannot handle reinvites (all phones are behind
different NATs, too messy). So I've set those to canreinvite=no.
Unfortunately, it's not making a difference. I still
get the 1-2 seconds silence at the
Hello *,
I recently started playing around with the SMS application. Several of
my SIP clients are FritzBoxes, with SMS capable DECT phones connected
via ISDN or analog line.
So far, SMSing works great: I defined extension 0193010[01] to receive
SMSes, which works well with the default settings
*bump* Anyone?
On 11/6/06, Curt Shaffer [EMAIL PROTECTED] wrote:
I wanted to add what we have both seen on traffic captures.
You see Caller 1's RTP stream. Call 2 comes in and you see the creation of
its RTP stream. After Call 2 is put on hold the RTP stream from Caller 1
disappears without a
My company has had a few screens go out on us, but all of those were
completely blank. I'm not sure if we just got a bad batch or what, but
the Snom phones are usually a solid piece of hardware. I'd try to RMA it.
Nick Hoffman wrote:
Hi guys. I just bought and configured a Snom 360 and have
Matt,
I haven't heard of this happening elsewhere, but I don't hear of every
issue with the phones.
It sounds to me that you've got more than enough information to raise a
case with our support team.
Gareth
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Today we appear to have discovered our first bug. We have an extension
setup to followme by ringing that extension + an external cell #
(ringall). If nobody answers after 20 seconds the destination if no
answer is set to go to the extensions voicemail in the followme module.
The problem is it
Honestly they are not that bad. When they first came out they were very buggy, but about 6 months ago I pulled one out and updated the firmware and actually over a few days was reliable.
On 11/7/06, RR [EMAIL PROTECTED] wrote:
On 11/2/06, Eddie Johnson Jr [EMAIL PROTECTED] wrote: Hello Matthew,
Thanks, that set off a light bulb In my spa3K my incoming dialplan was
set to (S0:405)
Since this is a one FXO unit and my [from-pstn] will always be that line
can I make it generic and use the 's' extension as I described? If so what
would that spa3k dialplan be? just s0 ?
Doug
On Tue, 7
I asked on freenode #asterisk a while ago about the followme and someone
was nice enough to share a macro with me that I'm using (although I use
voip outbound instead of zap, but it may be worth trying):
MYPHONE=IAX2/666
MYOUT=SIP/[EMAIL PROTECTED]
MYEXT=666
exten =
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten = 1,1,Goto(handle_fax,s,1)
exten =
Hi
Thanks for reply but that wasnt quite what i was trying to explain :-)
Bascially a users callerID would be their extension. On offnet calls i needed
to have the callerID reset as their DDI rather than their internal extension.
I endup using a mysql command in the dialplan to pull out a ddi
On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21
I want everything to stay in the VoIP server rather then briding. I
have notransfer=yes on, but it still seems to bridge the call
natively.. can I keep the RTP stream on the asterisk server some how?
On Tue, 7 Nov 2006, Pedro Silva wrote:
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
This patch is not added to chan-capi.org, but receivefax and sendfax is
available via capicommand(). Please see README of chan-capi 0.7.x package.
Armin
I tried to
I have 2 asterisk boxes connected via SIP
box 1 sip peer connected
to box 2 (ip addresses intentionally removed)
[ast20]
type=friend
host=x.x.x.20
insecure=very
context=subscriber
dtmfmode=inband
qualify=no
canreinvite=no
disallow=all
allow=ulaw
box 2 sip peer connected
I had a problem with snom where the screen went completly blank. Snom told
me there was an issue where that the cable going from the phone board to the
screen would fall out. I opend the phone and sliped it back in.
- Original Message -
From: Nick Hoffman [EMAIL PROTECTED]
To:
Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.
2006/11/7, Michiel van Baak [EMAIL PROTECTED]:
On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
Hello,
Anyone knows if chan_capi-0.7.1
Hi all !
I have a question regarding flexible callerid setting
using the misdn
I want to acheive the following:
when starting a call with 0 I want to display CALLERID (which is setup
right now) but when I start the call with 9 I want the callerid to be
surpressed.
How can this be done?
nik
Hi,
I've recently bought
new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I
just noticed something, which I first blamed on Asterisk and NATs (a 2 second
silence at the beginning of a call). Something I'venoticed also on
my old phone (which is having the same problem now,
Does anybody have a good link how to connect Sipura with Asteriks, both
behind NAT? I'm using FWD but their connection is like a weather
(especially IAX), I need something more reliable.
I was thinking of using stun and/or proxy but can not find any good link
explaining how to setup Linux
HiI'm trying to generate a Recall/Flash on an FXO (TDM400) connected to a PABX and failing at the moment. I was using the Flash application, which seems to generate a hook flash as opposed to a Recall. Debugging the Zap channel output using a second FXS channel gives me the following
[ TYPE: Null
On 11/6/06, Mosiuoa Tsietsi [EMAIL PROTECTED] wrote:
Hi,
After some more searching I decided to try USING unix ODBC for the
connection. I have both the unixODBC and unixODBC-devel packages on my
fedora box:
[EMAIL PROTECTED] /]# rpm -qa | grep -i unixodbc
unixODBC-2.2.11-7.1
Jason,First, before you start reading, get to the latest firmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), there have been significant echo improvements in this version.After many days of working with Audiocodes on this problem and much time spent here by multiple technicians trying to
Andres,
The Bicom Systems Operator Panel is probably what you are looking for. OPCOM
http://www.bicomsystems.com/docs/opcom/1.0/html/
This is included with every copy of PBXware and is fully supported.
If you care to register you may order a trial of PBXware with our SOHO.
Regards
Steve
steve
Does digium have a g723 codec can work pass thru mode
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of
Dear
How can I charge the incoming call to the destination call
,using a2billing
I used to make setaccount but it didnt work such a
loopback detected am using
context=default for incoming calls
I
Please help
Thanks
*
No
Hi,
Aastra IP Phones have two configuration files on TFTP, aastra.cfg and mac.cfg. Both are in text format, which makes editing easy. And aastra.cfg has system wide settings and mac.cfg has settings for each indivifual phones. This makes it really easy to change the global parameters system wide
I had this EXACT same problem, and 2.0.x is the problem
according to Polycom Tech Support.
I had such a hard time explaining the problem, too
Downgraded to 1.6.7 and all worked well again. Polycom says if
youre using Asterisk, dont
go past 1.6.7 until they say to.
From:
Matt wrote:
-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21
I want everything to stay in the VoIP server rather then briding. I
have notransfer=yes on, but it still seems to bridge the call
natively.. can I keep the RTP stream on the asterisk server some how?
Asterisk
Khaled wrote:
Does digium have a g723 codec can work pass thru mode
You don't need a codec loaded to do passthru, Asterisk just needs to
know about the codec (which it does in the case of G723.1). If things
are properly configured and codec negotiation happens the right way, it
should
On Wed, Nov 08, 2006 at 08:06:21AM -0800, Khaled wrote:
Does digium have a g723 codec can work pass thru mode
Digium does not provide a g723 codec. However to work in pass-through
mode you don't need a codec. A codec is used when transcoding the
stream.
--
Tzafrir Cohen
Say I have agents using a softphone like eyebeam that has 6 lines. They
log in to the queue. Say there are 3 agents in my queue. 3 calls come in
and all three agents are on a call. Now a fourth call comes in. Is it
possible to have it setup so that the 4 call rings on line 2 of one of
my
On 15:34, Tue 07 Nov 06, Pedro Silva wrote:
Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.
Hi,
Glad it worked. The generated file is a 'Structured Fax
File'
The faxreceive.php is
Any hints on downgrading? I placed the old SIP 1.6.7
on the right folder, but my phone wont pick it up and install it. It must
be thinking "this is an old version, ignore" or
something
I`ve never downgraded a phone, I tend to like upgrading
more :-)
Mike
From: [EMAIL
Try this site:
http://forum.voxilla.com/asterisk-users-group/sipura-asterisk-setup-14252-3.html
Juan Manuel
On 11/7/06, Joseph [EMAIL PROTECTED] wrote:
Does anybody have a good link how to connect Sipura with Asteriks, both
behind NAT? I'm using FWD but their connection is like a weather
Disregard my previous message, I succeeded in downgrading
my phones. And it worked, thanks Rick for the info. Is there any
Polycom-specific mailing list I should be on to be aware of stuff like
that?
Also, would you know how to check the version of sip.ld
remotely? I know how to reboot
HI folks,
I figured out where in the source code to hack the .wav file
permissions which were set too restrictive for me, but I cant figure out
how to do the same for the .txt file.
Looks like the voicemail.c file sets it nicely for
asterisk1.4beta3 using a #define statement early
Hi all !
I have a question regarding flexible callerid setting
using the misdn
I want to acheive the following:
when starting a call with 0 I want to display CALLERID (which is setup
right now) but when I start the call with 9 I want the callerid to be
surpressed.
How can this be done?
nik
I am getting ANI over an ISDN PRI T1 and they are sending *ANI*DNIS* as
the EXTEN I am under the impression this is an old way of doing it
on inband circuits such as EM wink but not used on a PRI.
Previously I had a T1 PRI from a different carrier and there was a
special field for ANI that
Also, change the port range to 1-10xxx in /etc/asterisk/rtp.conf and
match it up in your port forward on your firewall instead of 1-2
which is far more anyone is likely to need.
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Monday, November
I think in the features you can completely
wipe the sip image
Menus
Settings
Advanced
Admin Settings
Reset to default
Then format the file system
Bill
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006
1:49 PM
To:
Hi All,
I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:
asterisk sip sip TNT pri pri asterisk
The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other
If your Asterisk is behind a NAT you can use externip=x.x.x.x (sip.conf)
If your Sipura is behind a NAT you can use nat= yes (sip.conf)
Btu I'm really afraid that unless you use a SIP
proxy (e.g. Portaone) you won't be able to succesfully connect both
elements if they both are behind
I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN
Cards, if i press in a call the * Asterisk, Asterisk destroys the call
not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell
me in the warnings-log that the bridging was not successfull ?!
If have disabled
The default queue configuration would achieve this. Based on your queue
calling method (ringall, roundrobin, etc), all the agents would be able to
receive the 2nd call, and whoever answers it first gets it.
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Steve,
I am running EXACTLY the same software as you. SMP kernel and all. Are you
using the IDENTICAL non-SMP kernel verision# ? No crashes at all here for
the 2 weeks it has been up but we don't use meetme.
-Original Message-
From: RR [mailto:[EMAIL PROTECTED]
Sent: Monday,
Running several Aastra 9133i and 480CT phones with v1.4 firmware CentOS 4.4,
Asterisk 1.2.13, Zaptel 1.2.10, Freepbx2.1.3. Using all default settings
I have not seen that problem. I am not exactly sure we are creating those
exact same conditions but it sounds like standard extension use to
Asterisk is still going to try to native bridge the two channels. Once
this occurs chan_iax2 is going to notice that you don't want a native
transfer to happen and not do it.
Ok should it be giving me any indication that it has NOT done a
native transfer? Or does it just say 'attempting
Hi, I am a newbie putting together my first Asterisk system
and having a problem with the IVR handling incoming calls.
I installed the Asterisk Trixbox version 1.2.2 with a X100P
FXO PCI card. I have a PSTN line connected to the card. I set up
two extensions: 200 and 201. I created a
On Tue, 7 Nov 2006, RR wrote:
On 11/2/06, Steve Edwards [EMAIL PROTECTED] wrote:
I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is
mostly meetme conferences being created and closed all day long. Peak load
is around 200 SIP calls.
I was crashing 7 to 10 times a day
Hi all!!
I've made some changes to the applications that Astertest was using to
monitor the performance of the server. Now is also possible to track the
bandwidth usage of the server, this has nothing to do with the executable
(astertest.exe) itself but with the events that the Asterisk Manager
Update,
I loaded asterisk 1.0.10 and it worked straight away. I can send
unauthenticated calls to asterisk. Something in 1.2.9.1 and 1.2.13
are not allowing unauthenticated calls when insecure=very is set in
sip.conf, either in the global or peer context.
Are there any switches in the Asterisk
http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm
Theres not much in the article so only click through
if super interested but Im curious and looking for peoples
opinions.
What application integration would you like to see between
MS (either
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hi,
it turns out that the iksemel library (which i installed using an rpm) was
returning 0 when the function iks_has_tls() was called. it should return 1
otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by
running a test program i wrote, that calls iks_has_tls . it
Hi,
My messages to the list don't get through. This must be the tenth message i am
trying to send!
Please ignore this test message.
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I was planning on
using a TDM400P with 3 FXO 1 FXS, with the 1 FXS being used for a fax
machine. It now appears that Digium doesn't support this, are there other
manufacturers anyone can recommend that will support it? Has anyone used a
TDM400P in this setup and had it work without much
Hi all,
My Asterisk server is working fine, although every time that in the middle of
any call there is a reinvite, the user hears a glitch. Why is this happening?
How can I solve this problem?
Thanks in advance,
Ricardo Carvalho.
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Using SIP:
Just create another user account
say the softphones user's name is bob:
create [bob] (bob's main line on his softphone)
create [bob1] (same configuration options, then you can do
all your other configurations for this user )
hope this helps
anyone is open to correcting me :]
my 2
I think you could enable call waiting (*70) on those stations and they would
have the second line ring in. This is what I have done in the past. The
second call would continue to use the ring strategy configured in the queue.
You can also enable call waiting from the Asterisk command line by
Hi all,
First, I really hope that this message gets through since i had to change email
address on this list. Only one message from me with my previous address got
through.
I am running latest test version of Debian and I have done the following:
First, i did apt-get build-dep asterisk
Then I
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hmm, Id like to know that. How do you reboot remotely ? J
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006 2:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Sticky Polycom 501 keys
what is the sip.conf for 1239
which I'm going to assume is a extension on the TNT
Barry
JR Richardson wrote:
Hi All,
I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:
asterisk sip sip TNT pri pri asterisk
The TNT is running 11.0.6 and the asterisk
Title: Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue
When all else fails I resort to adding this in the sip.conf peer config:
Insecure=invite,port
It took me a while to figure out they can be used together.
Regards,
In Asterisk enter 'sip show peer name' and you can see this in
the Useragent field.
Example (for 2.0.1):
Useragent :
PolycomSoundPointIP-SPIP_501-UA/2.0.1.0313
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
MikeSent: Tuesday, November 07, 2006 11:13To:
'Asterisk Users
Answering my own question. If you want to connect an spa3K with
generic pstn inbound do the following...
for the pstn to voip dialplan in the pstn tab - (S0:ip-address-of-*)
in sip.conf
[sipurafxo]
context=from-pstn
etc.
Then in * extensions.conf use the s extension.
[from-pstn]
On Wednesday 08 November 2006 13:15, Ken Williams wrote:
I was planning on using a TDM400P with 3 FXO 1 FXS, with the 1 FXS
being used for a fax machine. It now appears that Digium doesn't
support this, are there other manufacturers anyone can recommend that
will support it? Has anyone used
They do get through. Messages you send to the list won't get sent back to you, because you sent them. On 11/7/06, Christian
[EMAIL PROTECTED] wrote:Hi,My messages to the list don't get through. This must be the tenth message i am trying to send!
Please ignore this test
Unified messaging would be nice. Not just having my VM's e-mailed to me, but to be able to manage them from with Outlook (or any other mail client for that matter) would be nice. I picture it sort of like an IMAP mailbox, and the mail client just has some kind of functionality to recognize that
On 11/7/06, Christian [EMAIL PROTECTED] wrote:
Hi,
My messages to the list don't get through. This must be the tenth
message i am trying to send!
Please ignore this test message.
On Wed November 8 2006 13:08, Alex Robar [EMAIL PROTECTED] wrote:
They do get through. Messages you send to
I _was_ sure until mention it just now... I certainly don't get a copy of any messages I sent to the list, whether I send from my personal or office accounts. Maybe the way my mail clients are handling it? If so, my apologies to Christian.
AlexOn 11/7/06, Nick Hoffman [EMAIL PROTECTED] wrote:
On
On Wed, Nov 08, 2006 at 02:24:23AM +0100, Christian wrote:
Hi all,
I am running latest test version of Debian and I have done the following:
First, i did apt-get build-dep asterisk
Then I downloaded the latest version of Zaptel, Libpri and Asterisk,
something with 1.4.
I am using Kernel
- Original Message -
From: Hadley Rich [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 08, 2006 5:07 AM
Subject: Re: [asterisk-users] Fax Digium
On Wednesday 08 November 2006 13:15, Ken Williams
Take a look at OVA.
mms://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Tuesday, November 07, 2006
9:13 PM
To: Asterisk
Users Mailing List -
- Original Message -
From: Nick Hoffman [EMAIL PROTECTED]
To: asterisk-users Mailing List asterisk-users@lists.digium.com
Sent: Tuesday, November 07, 2006 10:53 AM
Subject: [asterisk-users] Snom 360 flickering screen
Hi guys. I just bought and configured a Snom 360 and have noticed
I have seen this mainly with gmail. the logic is
why do you need your own postings. Fish around to see if there is a setting in
Gmail where it will keep the email. I know for myself I want the email's that I
sent. It lets me know that they went out as well as it helps for sorting the
On 11/8/06, Nick Hoffman [EMAIL PROTECTED] wrote:
They do get through. Messages you send to the list won't get sent back
to you, because you sent them.
Hi Alex. Are you sure about that? I receive a copy of every email I send to
the list.
I think it's just Gmail that hides them, especially
To hide the caller ID, do this:
exten = _9NXXNXX.,1,Set(CALLERID(all)=Unknown00)
exten = _9NXXNXX.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
bp
On 11/7/06, Nik Engel [EMAIL PROTECTED] wrote:
Hi all !I have a question regarding flexible callerid settingusing the misdnI want to acheive
AFAIK, in 1.2.x the insecure=very change in favor to insecure=port,invite,
also you can try with allowguest=yes
Regards
JR Richardson wrote:
Update,
I loaded asterisk 1.0.10 and it worked straight away. I can send
unauthenticated calls to asterisk. Something in 1.2.9.1 and 1.2.13
are not
This should
be in your Asterisk sip_notify.conf file by default I believe (if not, add it
with an appropriate name):
[polycom-check-cfg]Event=check-syncContent-Length=0
Then in the
Asterisk run this (assuming the phone is registered
properly):
sip notify
polycom-check-cfg user
If the
On 11/8/06, Steve Edwards [EMAIL PROTECTED] wrote:
All calls come in from a Tekelec 7000 via SIP.
Out of a peak of 200 calls, probably around 100 are in meetme, others are
listening to recorded messages or bouncing around in the menus.
Sounds exactly like what people in my system would be
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