I'm not too pleased with the phones, I have about 40 of them, some of the
displays tend to die and the dial pad feels to 'mushy' IMHO, just like the
keys on a good old ZX80 computer
Also I'm having some issues with sound quality on some phones, but I still
need to switch some phones to see if
I would like, that incomming calls displays (clid) on all ip phones in
group (helpdesk), but only on user specific extension ringing...
e.g. helpdesk employee have number 310, 311, 312 etc.,
when incomming call to 311 line, only 311 line ringing, but on all other
phones (310, 312) only display
We have a need to allow the caller who is in the middle of a call (i.e.
who is already bridged between a PRI channel and a SIP channel as the
result of entering a Dial cmd in the current context) to type something
like ## to cause the called party to be disconnected and to return
from the Dial
Kevin P. Fleming wrote:
'lspci -vb' does not understand IO-APIC mode, as best I can tell, so
the interrupt number that it reports is totally useless.
On my desktop machine, with an nVidia graphics card in a PCI-Express
slot, /proc/interrupts shows it using interrupt 185, and 'lspci -vb'
Hi,
Do you think there would be any interest in a softphone that supports
LDAP ?
why not? You can use ldap commands to connect to Domino and MS ADS, so a
softphone with ldap capabilities sounds like quite a good idea to me.
Stefan
--
in-put GbR
Hi Guys
Im having a problem getting CAPI to work on my
Traverse NetJet card.
CAPI is enabled in the kernel and Im using the mISDN
drivers with the NetJet patch. I cant seem to get astcapi to load
Heres the output im getting
Nov 12 21:18:36 VERBOSE[4011]:
== Registered
On Nov 12, 2005, at 12:11 AM, Francois Meehan wrote:
Hi all,
I have upgrade my kernel and asterisk to their latest release on a
Centos
4.1 box, now it won't start anymore.
Have you rebuilt Zaptel against your new kernel? If you upgrade the
kernel, you need to rebuild zaptel.
Tom
Cleaner looking, 12 line apperances, affordable sidecar, runs on
linux, developing XML services, very programable, buttons are firm in
a good way, simple layout for users
installing 30 right now...
On 11/12/05, Remco Barende [EMAIL PROTECTED] wrote:
I'm not too pleased with the phones, I have
Using fc3 with cvs-head from Nov 1st and TE205P (dual T1 port) with a
T1 cross-over cable between span 1 and 2. (udev properly defined.)
In /etc/zaptel.conf I have:
span=1,0,0,esf,b8zs,yellow
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs,yellow
bchan=25-47
dchan=48
And in zapata.conf I have:
George Pajari wrote:
I am aware of all of the above but does the fact you are posting this
mean that Digium is now aware of this?
It means I am :-) I have been trying to teach the tech support team
about this, but I don't think it has quite sunk in yet. I believe that
means it's time for
Teliax looks good - not comfortable with the soft limits but love the
free setup!
[EMAIL PROTECTED] wrote:
Have you looked into teliax? 4 simultaneous calls on a bus plan is
pretty good for less than $50/mo. And I cannot complain about the
quality or the support.
Greg
-Original
Do [EMAIL PROTECTED] have KDE or GNOME?
How to start GUI?
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No it does not. You should not run X with asterisk. If you
want answers to questions about AAH you should ask those questions on
that forum. http://sourceforge.net/forum/?group_id=123387On 11/12/05, Goran
[EMAIL PROTECTED] wrote:Do [EMAIL PROTECTED] have KDE or GNOME?How to start GUI?
Hi:
Does Trunk=yes in IAX save bandwidth as it should in
case the other server (voip provider) has SIP only?
Regards;
Chawki
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Hi,
chick*CLI show version files chan_sip.c
File Revision
chan_sip.cRevision: 1.907
chick*CLI show version files pbx_realtime.c
File Revision
pbx_realtime.c
Thanks that's the type of info I'm looking for...
I've heard some early grumblings but wanted to see if anything else has come
up...
From: Remco Barende [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sat, 12 Nov 2005
Answering self... this is a remote box and the guy that was suppose
to install the T1 crossover cable installed a loopback plug instead.
Sorry for the noise.
Using fc3 with cvs-head from Nov 1st and TE205P (dual T1 port) with a
T1 cross-over cable between span 1 and
Has anybody sucessfuly compilied spandsp-0.0.2pre21c under 1.2rc2?
I keep getting this:
[Nov 12 10:14:16] [app_rxfax.so][Nov 12 10:14:16] WARNING[12188]:
loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so:
undefined symbol: fax_set_phase_d_handler
[Nov 12 10:14:16]
Hiya
I've had Asterisk
working great with an X100P, but now I've got up to a TDM400P, with an FXO and
FXS card. I've changed what I thought I needed to, and upgraded the Zaptel stuff
from 1.0.9 to current CVS. Asterisk is 1.0.9.
I'm getting the
following when taking the phone off hook:
Nov
As far as I know, if the second server has only sip, it is going to be
difficult to connect each other. If that is the case, yes it is going to
save bandwith. Your total bandwith always it is going to be 0 kbps.
Regards,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
Thanks Tom,
That was it, after upgrading the kernel with Yum, it didn't change the
link for the modules. Fixed it manually, recompile everything and we are
up again.
Best regards,
Francois
On Nov 12, 2005, at 12:11 AM, Francois Meehan wrote:
Hi all,
I have upgrade my kernel and asterisk
I've had Asterisk working great with an X100P, but now I've got up to a
TDM400P, with
an FXO and FXS card. I've changed
what I thought I needed to, and upgraded the Zaptel stuff from 1.0.9 to
current CVS.
Asterisk is 1.0.9.
I'm getting the following when taking the phone off hook:
Make sure you have a /etc/asterisk/indications.conf Not every method of
playing tones requires this, but some do and it's a good idea to have it
anyway.
Rich Adamson wrote:
I've had Asterisk working great with an X100P, but now I've got up to a TDM400P, with
an FXO and FXS card. I've
Don't know from the limited amount of data that you provided.
Doh :)
Ensure you are loading wctdm driver (use 'lsmod' to see if
its loaded).
run 'ztcfg -vv'
That produces:
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
2 channels
Hi, I have an analog phone connected to a VigorTalk adapter. When I have
an active call and press R the call seems to be parked, the other end
hears MOH, but how do I transfer the call to another extension. All
options in features.conf is commented out, do I need configure something
here or is
Hello there, i have a swissvoice MGCP ip20 as the subject suggests. I
have it connecting to asterisk and it shows in `mgcp show endpoints`:
*CLI mgcp show endpoints
Gateway '192.168.0.10' at 192.168.0.10 (Static)
-- 'aaln/[EMAIL PROTECTED] in 'desk-phone' is idle
however when trying to dial
[EMAIL PROTECTED] wrote:
Thanks Armin, this version is working, but I still have an undefined symbol
in another module:
[pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:
ast_pthread_create
Nov 5 18:51:12
On Sat, 12 Nov 2005, Faris Raouf wrote:
[EMAIL PROTECTED] wrote:
Thanks Armin, this version is working, but I still have an undefined
symbol
in another module:
[pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so:
Paul,I have been using a SPA-3000 here in São Paulo, and after some tweaking it is working ok.I do recall that I had to set some line parameters due to the differences, but I need to open the configs of the ATA again Meanwhile, if you have any additional info let me know, I might be able to
I've got a customer on an IAXy and another with their own Asterisk box as a PBX
with an array of Cisco, GrandStream, ATCOM, and xten hard\soft phones.
Same LEC, same Asterisk box on our end, same broadband provider on the client
ends
With no packet loss, 15 ms pings, 13 hops, the IAXy sometimes
I'm trying to get this to work, but it always goes to step 4 - there
something I don't understand about LEN with GotoIf:
exten = _,1,NoOp,${CALLERIDNUM} ; CID as
received
exten = _,2,GotoIf([LEN(${CALLERIDNUM}) = 10]?4:3) ; if CID length =
10 then do
I think you have to swap the 4 and 3 around in your gotoif - it's true then
false...
PaulH
- Original Message -
From: Bart Fisher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 8:32 AM
At least the soft limit is explicitly published (X Minutes) as
opposed to most companies' policy of There is a soft limit, and we
will not tell you what it is, but if you reach or exceed it we will
[charge you $100/day | terminate your service | switch you to a more
expensive plan without notice].
No. IAX and SIP are two completely different protocols for sending
voice across IP networks. IAX-Trunking is a feature of IAX, and the SIP
protocol does not have any such method for conserving bandwidth by
combining data from multiple calls into one packet.
-Rusty
On 11/12/05, chawki hammoud
Hi
Upgraded to Gentoo 2.6.14-r2. When compiling zaptel, warning appears.
Zaptel module loads fine.
Cannot remember seeing this on 2.6.13. Is there another Kernel switch
that needs to set. CRC and RTC is set in kernel.
make[1]: Entering directory `/usr/src/linux-2.6.14-gentoo-r2'
CC [M]
In addition to the mail below,
It is not the realtime! ARA is great.
Moving the peers to sip.conf, and ignoring extconfig.conf for a test,
discovered that, when left empty (secret=blank_space) is ignored as
commented (;secret=whatever). Obviously the sip channel was actually
prepared for the
FYI,
I was able to get my NEC NEAX 2400 and my * box to talk to
each other. For those who want to know more Ive added a wiki page:
http://www.voip-info.org/wiki/view/Asterisk+NEAX2400
-MC
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I am having the same problems. The polycom
phonesthe 501 or 601 or 301 will list more more than 7 buddies neither
will the 601 with an expansion module monitor more than 7 other
phones.
Is there anyone out there who can explain waht is
happening. My reseller can not help. I am surprised no
Hello All,
Can anyone give me an example of how can I configure Asterisk to Pass-
Thru G729 and G723.1 codec?
Thanks,
Neal
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I just recently upgraded to the latest HEAD, and am now getting the
following warning:
-- Including context 'fromcnet' in context 'pots'
Nov 12 18:45:17 WARNING[3035]: pbx_config.c:1697 pbx_load_module:
Invalid priority/label '' at line 17
-- Including context 'longdistance' in context
This might help you:
${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority
after a blind transfer (use ^ characters in place of
| to separate context/extension/priority when setting
this variable from
Curren,
Can you tell us a little more about the environment you are deploying these phones in, how many phones and what kind of setup?
Omar A. SabekOn 11/12/05, Curren C. Calhoun [EMAIL PROTECTED] wrote:
Thanks that's the type of info I'm looking for...I've heard some early grumblings but wanted
From the dump that I have attached It looks like the first attempt
at register does not work then followd by a second register which
then works.
This is happening on all the SIP phone attach to asterisk. The version
of asterisk here is 1.2.0b2.
Here is sip.conf for ext 204
[204]
username=204
Carlos Alperin wrote:
As far as I know, if the second server has only sip, it is going to be
difficult to connect each other. If that is the case, yes it is going to
save bandwith. Your total bandwith always it is going to be 0 kbps.
Hehe funny!!!
--
Cheers,
Matt Riddell
[EMAIL PROTECTED] wrote:
In addition to the mail below,
It is not the realtime! ARA is great.
Moving the peers to sip.conf, and ignoring extconfig.conf for a test,
discovered that, when left empty (secret=blank_space) is ignored as
commented (;secret=whatever). Obviously the sip channel was
Amund Nygaard wrote:
Do you know anywhere to find information about this?
set the option canreinvite=yes in the sip.conf section for that user and make
sure you don't have anything in the dial line that would keep asterisk in the
communication (i.e. t or T etc)
--
Cheers,
Matt Riddell
PLEASE DO NOT POST IN HTML! :)
Gervais de Montbrun wrote:
YPE HTML PUBLIC =22-//W3C//DTD HTML 4.0 Transitional//EN=22
htmlheadmeta http-equiv=3D=22Content-Type=22 content=3D=22text/html; c=
harset=3DISO-8859-1=22
style type=3D=22text/css=22body=7Bmargin-left:10px;margin-right:10px;marg=
Waldo Rubinstein wrote:
I upgraded one of our gateways connected to the PSTN with a TE410P to
1.2rc1.
Any ideas?
Maybe change DEFAULT_CID_RINGS in /usr/src/asterisk/channels/chan_zap.c
--
Cheers,
Matt Riddell
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Please post in PlainText not HTML to this list.
Thanks!
:)
--
Cheers,
Matt Riddell
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Jason Walker wrote:
The statement of zaptel being required is strange...I use IX trunking
exclusively for my servers. Two of them have no zaptel/Digium hardware and
the trunk calls are fine.
IAX trunks require a zaptel timing source, be it hardware or ztdummy.
--
Cheers,
Matt Riddell
Hugh Jackman wrote:
How that could possibly be done with a special class of MOH as the
file will continue to play from wherever it stops last time? Can we
force * to spawn a MOH process for every incoming call?
Raw music on hold will do this.
--
Cheers,
Matt Riddell
I am getting the same with cvs head as of today.
on Thu, 10 Nov 2005 15:29:05 -0600 Anton Krall [EMAIL PROTECTED] wrote:
Hi Steve!
I tried compiling the recent spandsp 0.0.2pre21c with 1.2rc1 and I also
compiled unicall for r2mfc support.
R2mfc seems to be working great! But when I load
Hi
I was using a2billing with mysql-4.1.12 and php-5.0.4 very successfully
(thanks to areski for this great project and its invaluable assistance
to solve some issues in my last installation...) now I´ve upgraded
mysql to last release 5.0.15 and, without changes in 'mya2billing'
database I am
It would have been better to send such a long log directly to me, rather
than to the mailing list.
Ok .
That said, the log did the job. I found the problem. I just posted
another update to the MFC/R2 software - 0.0.2e and 0.0.3pre8
Regards,
Steve
I installed the new libmfcr2
Matt Riddell on November 12, 2005 at 9:53 PM -0400 wrote:
PLEASE DO NOT POST IN HTML! :)
Sorry Matt, this is controlled server side for me. The server should be sending in html and plain text and displaying what your email client should be able to read... Isn't this what is happening?
Any
I hope someone can help me with this
I have a cisco gateway trying to dial my box via H.323.
The call comes through ok and gets routed properly.. only
thing is NO AUDIO
I am confident that I have narrowed down the problem to a
codec issue.
I have the relevant G729 licences which
On Sat, Nov 12, 2005 at 03:35:43PM +0800, Dulmandakh Sukhbaatar wrote:
I run debian sarge with kernel-image-2.6.8-2-686, compiled and installed
zaptel 1.2rc2 without any problem. Modules can be loaded without
problem. Also I have TDM04M or TDM card with 4 FXO modules.
/etc/zaptel.conf
On Fri, Nov 11, 2005 at 11:08:25AM +0800, Ryan Pagquil wrote:
Hi,
I successfully installed asterisk in SPARC64/Linux as the
voicemail for my SER installation. No problem when I run it, but the
problem is when I forward the voicemail traffic to it, on the user agent
side I heard the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Saturday, 12 November 2005 7:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet
At least use
Rafael R. GV wrote:
Hi
I was using a2billing with mysql-4.1.12 and php-5.0.4 very successfully
(thanks to areski for this great project and its invaluable assistance
to solve some issues in my last installation...) now I´ve upgraded mysql
to last release 5.0.15 and, without changes in
Mr. Fleming:
Thank you so much for your email -- that's the best news I've heard all
week.
Kevin P. Fleming wrote:
George Pajari wrote:
I am aware of all of the above but does the fact you are posting this
mean that Digium is now aware of this?
It means I am :-) I have been trying to
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