Hi,
I have a client (xlite) connected to my server, on the server I have
type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the
server is listening on port 5060.
However, xlite is connect to a router where the port 5060 is blocked,
therefore, I am using 5065 and I have an
Dear all
I have '*' 1.4.11 and 2E1 port hardware installed on it now
i have single lan not nat anywhere ( 10.20.1.x ) all phone in single network
domain without NAT now i have configured canreinvite=no so that asterisk work
in meddle path of RTP so what is suggestable option
I have had a 941 for a couple of years. It works great for daily use
at the office and I'm quite pleased with it.
On 9/23/07, Robert Webb [EMAIL PROTECTED] wrote:
Is anyone out there using any of the newer linksys phones since Cisco
took over? I am more specifically looking at the spa-941
On 9/21/07, Tim Panton [EMAIL PROTECTED] wrote:
I don't think IRC is the natural habitat of people who like NOW,
NOW is for people who like web based GUIs. You are much more likely
to find them over in the web based digium forums.
Since we're talking about this, I have been on the #asterisk
I've installed them in a number of sites. The phones are good and easy
to provision. If you need a good speakerphone then choose another
phone.
If there is something specific you need to know let me know.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi all,
I wonder why my call was transferred when I pressed '#' in a conversation.
How can I disable this kind of call transfer?
Thanks.
David
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On Monday 24 September 2007 10:21:44 VoIP Newbie wrote:
I wonder why my call was transferred when I pressed '#' in a conversation.
How can I disable this kind of call transfer?
Thanks.
David
Take a look at features.conf - probably there is blind transfer enabled on #
key.
Regards,
Atis
--
Hi all,
We need to get better echo cancellation on an Asterisk gateway.
Currently it has two TE410P (1st gen) cards. So would it be possible to
just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ?
In that case a Sangoma A108d card would be nice as well ?
What configuration
Hello all,
I'm looking for a solution to offer Virtual PBX, to my clients. I just saw
software with multi-tenant support and I tested it, but no one likes me
enought.
Finally, I want to offer this service like a kind of hosting.
Has you experience with multi-tenant software? Which has you
Is anyone out there using any of the newer linksys phones since Cisco
took over? I am more specifically looking at the spa-941 942's. Just
curious about call quality, programability, and functionality with asterisk.
We've used the SPA-942s in most of our recent installs and been very
Dear Tzafrir,
I just try Destar, but one thing I dislike was, that there are no
posibilities to login the manager of each virtual PBX.
Then customers cannot manage their owns PBX.
VoiPCrazy
2007/9/24, Tzafrir Cohen [EMAIL PROTECTED]:
On Mon, Sep 24, 2007 at 11:38:38AM +0200, voip crazy wrote:
My experience is that both T1 and TDM cards from Sangoma which come with
HWEC (hardware echo cancellation) give you excellent sound quality, better
than those without HWEC. They do something which not only removes echo, but
improves sound quality as well. Digium I haven't tried because of
On 9/24/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote:
Is there any way to setup the asterisk cli to use such keybindings ?
...
Set in your environment:
AST_EDITOR=vi
before starting Asterisk.
(See main/asterisk.c)
Great !
Linksys are great phones. I like them but there only problem is limited line
appearances. I prefer Aastra over them because Aastra has more lines
appearances. They both are good. If you are not planning to have more than 4
lines, then Linksys is a great phone.
On 9/24/07, Chris Bagnall [EMAIL
Hello
each 15 days my Asterisk crashes.
Every time it happens I try to change something in its configuration to
avoid the next crash.
I already checked the logs but I don't know what to do.
Can someone tell me whats the problem?
These are my Asterisk logs:
http://vox.fccn.pt/crash
Thanks
On Mon, 2007-09-24 at 16:45 +1000, Klaverstyn, David C wrote:
I've installed them in a number of sites. The phones are good and easy
to provision. If you need a good speakerphone then choose another
phone.
SNIP
I'd like to echo this. The SPA-942 'looks' the part, it's good value
and bulk
Hi List!
does anyone played around with the OCS and Asterisk?
I want to integrate OCS and Asterisk to enable Office Communicator 7.0
client to make and receive calls from PSTN
I know that I need patch Asterisk to support TCP. But I am a bit ( a lot)
lost
Which more things should I need to
Hello,
I am having an issue whereby calls are being dropped randomly. I have an
ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk
install is based on Trixbox 2.0. However, I have updated the source code
to the following. The Asterisk release is asterisk-1.2.20. Zaptel
release is
I am using an asterisk to call another asteisk (i.e
Dial([EMAIL PROTECTED]) in asteriskA). After that, the following error
message displayed and asterisk crashes at once. Anyone has such
experience and can help to fix it?
asterisk version: 1.4.11
zaptel 1.4.5.1
using RealTime
On Monday 24 September 2007 14:06:27 Joao Pereira wrote:
Hello
each 15 days my Asterisk crashes.
Every time it happens I try to change something in its configuration to
avoid the next crash.
I already checked the logs but I don't know what to do.
Can someone tell me whats the problem?
Hai,
Is the Asterisk supports PMC-Sierra Analogue Telephone Adapter?
Thanks Regards
Bincy K Philip
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Hi all,
Thanks everyone for the quick reply. I think I'll go for the Sangoma as
I just saw that the VPM450M is no option anyway:
http://store.voxilla.com/voip-products/digium-vpm450m.html
We have rev1 cards, while the module needs rev3 or greater.
Thanks again,
Leon de Rooij
On Mon,
All of my testing has shown it be be pretty clean.
We have it on our contact us page of our website and we also give that url to
overseas (India, Germany, Japan) contacts and some
have used it.
Some do not want to open up the iax2 port in their firewall, but that is their
issue.
I wanted to
Leon de Rooij wrote:
Hi all,
We need to get better echo cancellation on an Asterisk gateway.
Currently it has two TE410P (1st gen) cards. So would it be possible to
just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ?
In that case a Sangoma A108d card would be nice as
At this point I do not think the problem is the wiring. What else should I
try?
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asterisk-users
Is anyone out there using any of the newer linksys phones since Cisco
took over? I am more specifically looking at the spa-941 942's. Just
curious about call quality, programability, and functionality with
asterisk.
We like these phones, hundreds deployed to business customers.
Mass
If you are not planning to have more than 4
lines, then Linksys is a great phone.
Out of the hundreds of users I've spoken to, there are only 2 individuals I can
think of that routinely juggle more than 2 concurrent calls. The 4 line
limitation has never been a problem for the vast majority
Brian Alexander wrote:
At this point I do not think the problem is the wiring. What else
should I try?
Have you confirmed that the failing card is working correctly? Maybe
the card is at fault.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
On 9/24/07, Doug Lytle [EMAIL PROTECTED] wrote:
Have you confirmed that the failing card is working correctly? Maybe
the card is at fault.
All of the cards have been confirmed to work by themselves.
-Brian
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Brian Alexander wrote:
On 9/24/07, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Have you confirmed that the failing card is working correctly? Maybe
the card is at fault.
All of the cards have been confirmed to work by themselves.
The only other suggestion I
On 9/24/07, Doug Lytle [EMAIL PROTECTED] wrote:
The only other suggestion I have would have would be to use IAX instead
of PRI for inter-machine communications.
LOL Yeah, normally that is what I would use. Unfortunately it is not an
option for this...
Finally, press and hold all 4 arrow keys until the phone bleeps, then
capture the log files dumped to your provisioning server one last time.
If the problem's not obvious from reading the logs, escalate these logs
to your Polycom reseller and ask them to open a ticket with Polycom on
your
I would use SER or OpenSER as a middle man.
Set it up to receive via TCP and send it on to the asterisk server using UDP.
Kind Regards
Jon Leren Schøpzinsky
Solution Engineer
Dansk Erhvervs-Telefon A/S
tlf: +45 88200336
mob: +45 31206709
-Original Message-
From: [EMAIL PROTECTED]
Hi All,
I read rumors of a potential Asterisk 1.4.12 release for last week. I
would like to start testing Asterisk 1.4 on Solaris, but, the fix for
the segfault in res_features is only in the current development trunk. I
would much rather test a release version, and as such, need to wait for
the
Brian Alexander wrote:
On 9/24/07, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The only other suggestion I have would have would be to use IAX
instead
of PRI for inter-machine communications.
LOL Yeah, normally that is what I would use. Unfortunately it
Note that the newish SPA962 has 6 appearances and a color screen.
I've noticed that the bright color screen does impress people when
they first see it. PoE is also very nice and web provisioning was
quite easy. I've yet to try a more automated provisioning method on
it. I know that getting
Tzafrir Cohen wrote:
On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote:
This might sound lika a small issu, but here it goes: I'm a long time
unix user and my shell history usage and editing is configured to use
vi keybindings; it's something that's already built into my fingers
Hi,
In what direction should I start looking so that I can let Asterisk
forward (pass through) sip calls containing strange codecs without
transcoding? (Not just the G.729, which already seems to have some sort
of support in Asterisk.)
Are there some config files to fix, or should I
Interesting comment on the speakerphone. Have you found a reasonably priced
desk set with a good speakerphone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn,
David C
Sent: Monday, September 24, 2007 2:45 AM
To: Asterisk Users Mailing List -
Douglas Garstang wrote:
Wow. Polycom phones are STILL doing that? I haven't been involved with
Polycom phones since before January, and it was a problem back then too.
Jeez...
Doug -- he's using 1.6.7 firmware.
-Stephen-
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Hi, Greg:
I really can't recommend upgrading to a 2.x firmware highly enough. Many
people had the spontaneous reboot problems and I think they were all
solved by going to current 2.x firmware.
-Stephen-
Gregory Boehnlein wrote:
Finally, press and hold all 4 arrow keys until the phone bleeps,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: Monday, 24 September 2007 6:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Anyone use the Linksys phones?
Is anyone out there using any
On 9/24/07, Craig Guy [EMAIL PROTECTED] wrote:
The only nasty thing I've found is that whenever the handsets resync they
reboot even if no settings have changed. When this occurs anything
connected to the phones second Ethernet port will drop connection for a few
seconds.
The phones can
If you pay for the free calling via wifi, TMobile bases cost of call
where it initiates. So you call your long lost buddy from the house,
jump in the car, drive for an hour, the entire call is free. If your
buddy calls you as you're pulling in the driveway, you have the same
hour long call,
On 9/24/07, Steve Davies [EMAIL PROTECTED] wrote:
The phones can send a parameter to the provisioning server to indicate
that they want an Update if they do this, and you send no network or
other major config parameters, the phone does not reboot.
Look at the Linksys provisioning PDF for
On Mon, Sep 24, 2007 at 03:18:53PM +0100, Bruce McAlister wrote:
Hi All,
I read rumors of a potential Asterisk 1.4.12 release for last week. I
would like to start testing Asterisk 1.4 on Solaris, but, the fix for
the segfault in res_features is only in the current development trunk. I
would
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is ${CALLERIDNAME})
exten =
Hi,
I have an IBM server running latest asterisk 1.4.x connected to a
Siemens
hi-path user a TE120P single-span. Approx every 8 hours (although not every
8 hours and sometimes 2 in a row) at exactly the same time I see the
following errors
Does anyone have any suggestions / ideas ?
On 9/24/07, Brian Alexander [EMAIL PROTECTED] wrote:
At this point I do not think the problem is the wiring. What else should I
try?
Is this the latest Zaptel? Is it 1.4 or 1.2? I may have missed it in a
previous message if you mentioned it.
I've got a setup where I have a 2 port PRI on
try ${CALLERID(all)}
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten =
Those variables were deprecated in 1.2 and removed in 1.4. You should
read both the 1.2 and 1.4 UPGRADE.txt files. Also read README.variables.
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is
I'm wondering how the speakerphone on the 962 compares to the Cisco 7960?
I found that the SPA-941 did not work well in noisy situations, unlike
the Cisco 7960 which performs flawlessly. I find that the automatic
gain circuit with regards to how it functions with speakerphone VOX on
the Cisco
We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI
DNIS is received in-band DTMF in a format such as *7145551212*8002*
What happens when there are 30 or more calls, asterisk might see is DNIS =
802 or ANI = 4551212 for examples, where parts of the numbers are dropped.
All
Hello All
I have a TDM2400 card with 4 FXO, and the the following problem: This card
answered all the calls, but for the caller, the call is ringing and I don't
hear nothing when it has picked up. Here is piece of my log:
Thanks for any help.
== Starting post polarity CID detection on
Hi Guys,
I want to configure asterisk to act as media server on my network.
I have one specific situation descibed bellow.
Collect Calls
1. Subscriber A call to subscriber B
2. Gateway in A side, send this call to media server (asterisk) and
the asterisk send the call to subscriber B
3. When B
Hi, anybody can give me instruction on how to setup multi-line appearence in
Asterisk.
I have a Polycom soundpoint IP650 phone and want to provision multi-line for
it.
Thanks
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Hi all,
I am building out a new platform and I need help with a couple of
items. I need to have an extension 101 that is public (on business
cards, in the directory, etc...) however I want this extension to
exist as a hunt group with a ring all strategy so two phones (107
which is the private
I have an Asterisk 1.4 setup behind a TDM800P with 5 incoming lines,
which happen to be inside my university's PBX. I have a problem
wherein the university's PBX is apparently sending its CID spill after
the second ring for a certain class of incoming calls (those from
outside the PBX). I've
Sorry to drag up an old thread, but the backport of ringinuse is a
godsend for those of use stuck using asterisk 1.2 (trixbox 2.2). Many
thanks, Gavin
GTG
On 1/21/07, Gavin Hamill [EMAIL PROTECTED] wrote:
Nothing much to be said.. I backported ringinuse, autofill and the QueueLog
application
I was progressively upgrading this phone from 3.1.2
to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but
when I went to 4.0.0 (Even adding the more specific
2345-11500-040.bootrom.ld), it won't run, and
just keeps rebooting.
Now I've got a really nice doorstop unless someone
knows how to get out of
I read something the other day about 4.0.0 being not-quite-right.
PaulH
On Mon, 2007-09-24 at 19:49 -0500, Doug wrote:
I was progressively upgrading this phone from 3.1.2
to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but
when I went to 4.0.0 (Even adding the more specific
I have googled and can seem to find the answer to this one Does
anyone here have experience with externnotify in voicemail.conf?
The sample states that it will run when a message is delivered and
retrieved.
Does asterisk pass any arguments to the script?
Thanks.
Forrest Beck
At 20:48 9/24/2007, Paul Hales wrote:
I read something the other day about 4.0.0 being not-quite-right.
Hmmm. Should I downgrade? If so, what versions?
BootRom? sip.ld? etc?
PaulH
On Mon, 2007-09-24 at 19:49 -0500, Doug wrote:
I was progressively upgrading this phone from 3.1.2
Hai,
Is the Asterisk supports PMC-Sierra Analogue Telephone Adapter?
Thanks Regards
Bincy K Philip
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a2billing so far seems to be quite comprehensive compared to the other
freeware asterisk-based billing solutions available out there.
We are building our own billing solution(due to the very peculiar
requirements, one of which is to bill the callee, rather than the
caller). We are achieving
I am running sip 2.2 with an older bootrom, and the phone is running
fine.
PaulH
On Mon, 2007-09-24 at 22:32 -0500, Doug wrote:
At 20:48 9/24/2007, Paul Hales wrote:
I read something the other day about 4.0.0 being not-quite-right.
Hmmm. Should I downgrade? If so, what versions?
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
dialplan)
Calls from both sipgates make my hardware phones ring
But here
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