Hi,
If several phones register to the same sip.conf section what will happen
with a Dial SIP/shared in asterisk?
All phones ringing and the first one to answer gets the call?
Undefined behavior?
Thanks,
--
Jesus is coming! Everyone look busy!
___
Hi,
When I Dial(SIP/1SIP/2SIP/3) if any of these phones has a forward to
another destination (302: moved response) then the simultaneous ring
stops immediately and the incoming call goes to wherever the forward
points to.
We are using simultaneous ringing as a fallback when the receptionist
On Fri, Apr 01, 2005 at 04:12:05PM +0200, Julius Vindex wrote:
When I Dial(SIP/1SIP/2SIP/3) if any of these phones has a forward to
another destination (302: moved response) then the simultaneous ring
stops immediately and the incoming call goes to wherever the forward
points to.
We are
Hi,
Is it possible to have distinctive ringing in a queue?
I've tried:
exten = s,2,SetVar(ALERT_INFO=Custom 1)
exten = s,3,Queue(standard|r)
without success.
However the SetVar(...) works fine when just dialing a SIP device.
Any ideas?
___
Please find attached a patch I made to app_queue.c to add distinctive
ringing support. So the following works:
exten = 2131,1,SetVar(ALERT_INFO=Internal)
exten = 2131,2,Queue(standard|r)
I took code in app_dial.c and lightly adapted it.
I hope this gets included in * as it is really useful. I
On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote:
If you have it, can I get a copy please, or possibly can you send it to the
keeper of http://www.freedomphones.net/polycom/files/
I am looking for the latest boot image too.
1) I have the 1.4.1 firmware. To whom should I send
On Tue, Feb 01, 2005 at 09:41:25AM -0500, Noah Miller wrote:
I faithfully followed the instructions from:
http://www.voip-info.org/wiki-
Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
but still the message waiting indicator doesn't flash when a message is
waiting. There is a brief
Hi,
I am mostly happy with my Polycom IP600 but it apparently needs to check
the FTP server every minute. I couldn't find any obvious setting related
to that behavior in the configuration files.
Any idea how to curb the IP600's spurious network activity?
Thanks,
--
Lord, protect me from your
On Tue, Feb 15, 2005 at 07:56:10PM +1100, Adam Goryachev wrote:
On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote:
Hi,
I am mostly happy with my Polycom IP600 but it apparently needs to check
the FTP server every minute. I couldn't find any obvious setting related
On Tue, Feb 15, 2005 at 08:26:42PM +1100, Adam Goryachev wrote:
On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote:
You are right, this activity is related to logging.
After consulting the admin manual I am unsure as to what settings
related to logging are safe to change
When I access voicemail remotely, from a gsm phone say, some extra
characters get inserted in my dtmf tones: when I type , *
understands 88f8f8 (it always seems to be 'f'):
-- Incorrect password '88f8f8' for user '2130' (context = any)
And the 'f' always starts after the second digit. Might
Hi,
I am in the process of setting up call forwarding through capiECT with
the 7960's CFwdAll button.
When the phone redirects the call to an outside number (through a 302
SIP redirect) then the CAPI[contr1/xxx] channel becomes Local/[EMAIL PROTECTED] as
the call is reinjected into the
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186
phones.
All sip entries have:
callgroup=1
pickupgroup=1
However I am unable to remotely pickup a ringing phone using *8#. I get
fast busy tone. Is there some flag to add in extensions.conf ?
Thanks in advance,
Hi, I am testing a 7960 in this context:
[SIP] --- VPN --- [*] --- [ANY]
(ANY == any type of phone: isdn, SIP, IAX, etc.)
the call goes through and is dropped after 5 seconds with this message
in the log:
File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on
call IP address
Hello,
We have been using a Diva 4BRI with our Asterisk PBX through the capi
interface for almost a year now with good results. However, recently we
started to hear heavy clicking sounds in our phones when two
simultaneous incoming calls are processed by the card. The clicking does
not originate
Hello,
As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco
7960 with *8 but the extension then keeps ringing indefinitely, even
though I picked up the call.
Is this a known issue? Thanks,
--
There are no Debian developers in any part of Hell, because the good
karma
On Thu, Sep 04, 2003 at 10:56:10PM -0700, Andrew Gillham wrote:
Unless you're hoping to load Linux or some pirate image in the future,
there is no
reason to stay with the old code.
At least I have not experienced any new issues I can attribute to the
update to 5.3 code.
Hello,
I bought
On Tue, Sep 09, 2003 at 08:23:49AM +, WipeOut . wrote:
OK you are correct..
*8 picks up the call..I wonder why *8# does not work??
I also had the same problem that the phone that I collected the call
from did not stop ringing..
I have the same problem. Mark Spencer is working on the
On Tue, Sep 09, 2003 at 07:57:20PM -0400, Jeremy McNamara wrote:
Travis Johnson wrote:
I've called NuFone and was not impressed by their voicemail answering
system (choppy) and was unable to even leave a message before the phone
call was disconnected (in the middle of the
recording).
If like me you run * on a VPN (or multihomed) gateway and want to serve
remote SIP clients, make sure you have
bindaddr = 192.168.0.1 ; or whatever is your box's private IP
otherwise * might bind to its public IP and send it as return address in
the SIP call setup, which will (should) be
On Mon, Sep 15, 2003 at 10:48:00PM -0400, [EMAIL PROTECTED] wrote:
I upgraded my 7960 firmware to ver 4.4. I now can't make any calls and
I get errors (retrans_packet) on call on the console maximum retries
exceeded. And ideas?
Check that the bindaddr in sip.conf is set to a reachable
On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote:
On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote:
Anyone have a good source for BT-101 phones?
Yes.
But it may not work for you because I've no idea on which of the 5
continents you are.
I am looking for Grandstream
On Tue, Sep 16, 2003 at 11:10:33AM +0200, Jean-Marc V. Liotier wrote:
On Mon, 2003-09-15 at 11:52, Klaus-Peter Junghanns wrote:
i dont think that the Eicon Diva Server 4BRI's NT mode feature will
work with linux/capi. I think the feature in the driver is for their
PRI cards (where
Suddenly after recompiling my 2.4.22 kernel I can no longer load
chan_zap:
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span
status: Inappropriate ioctl for device
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to
register
On Mon, Sep 22, 2003 at 03:41:43PM -0400, Jeremy McNamara wrote:
cvs update the zaptel source and make clean install it.
That did it, thanks a lot.
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[EMAIL PROTECTED]
On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote:
My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of
service. They were deployed for about 6 months. These include the AC power
adapter and station license. We also have some other related equipment. If
someone is
On Mon, Oct 20, 2003 at 09:21:45AM +0200, Michael Devenijn wrote:
Justy to let you all know
that i tested 7905G phone with a SIP image (latest download) and it
works great ! for a reasonable price but with a good quality and a
brand ... which inspires trust and helps selling better
The
Hi,
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with Transfer?.
Is there a way to escape the pound key, short of disabling transfers?
Cheers,
--
Make it idiot proof, and
On Mon, Oct 20, 2003 at 02:55:18PM +0100, WipeOut wrote:
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with Transfer?.
Is there a way to escape the pound key, short of
On Mon, Oct 20, 2003 at 08:49:42AM -0700, John Todd wrote:
At 2:54 PM +0200 10/20/03, Louis-David Mitterrand wrote:
Missing a microphone to work handsfree (or i didn't find it.) but
strange enough their is a speaker ...
Yeah, that's a real bummer. Cisco calls that feature Monitor mode, ie
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
Also trunking requires that some sort of timing device (digium card or
ztdummy) be in place for trunking. Otherwise trunking is disabled.
What does ztdummy require to work? kernel compile options? Does it work
on SMP systems?
On Thu, Oct 23, 2003 at 07:55:09PM +0200, Olle E. Johansson wrote:
Louis-David Mitterrand wrote:
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
Also trunking requires that some sort of timing device (digium card or
ztdummy) be in place for trunking. Otherwise trunking
On Mon, Nov 03, 2003 at 11:32:11AM +0100, Roy Sigurd Karlsbakk wrote:
If your DSL link is the bottleneck, rather than earlier hops back
through the providers network, the provider could also prioritize VOIP
packets going up the DSL line. That requires a cooperating provider,
of course.
Hi,
At the * console I periodically get these messages:
Dec 9 10:58:11 WARNING[-1248765008]: chan_iax2.c:5021 socket_read: midget
packet received (1 of 4 min)
Which seem pretty inocuous.
Google say (almost) nothing about that subjet.
What does it mean?
--
Field Artillery lends dignity to
Hello,
I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42
firmware to the latest version (1.56) through tftp.
The phone loads the .inf file, then the correct firmware file (as stated
in the ST2030S.inf), then it reboots and loops doing these same things
again and again. The
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot figure out what is wrong with my IAX2 trunk.
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote:
I second that, and I think I remember hearing Mark talking about it too. But.
What type of encryption can you do that does not introduce latency?
That said, I would like it to support hardware encryption cards.
I
On Sat, Nov 29, 2003 at 11:28:56PM -0600, Mark Spencer wrote:
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in Paris sometime near there. Any one out
there interested? Anyone in Paris who could help organize something like
that? :)
On Mon, Dec 15, 2003 at 09:25:05AM -0500, John Todd wrote:
Paul -
Yes, your description is correct.
- moving the phone (no ethernet passthrough) results in no symptoms
You might have a virus on that XP box that totally saturates the poor
7960 switch with bogus IP packets.
--
May the
Hello,
Has anyone used Polycom's VSX line of videoconferencing equipment with
Asterisk?
It seems some of their models, namely the newer VSX 5000, supports SIP.
--
The Internet used to be a lot of smart people sitting at dumb terminals,
but now its a lot of dumb people sitting at smart
Hello,
I just acquired a used Cisco 7920 wi-phone and it mostly works with
the newest asterisk and chan_sccp, but it reboots after most calls.
Would a kind soul send me the latest firmware for that phone?
Thanks in advance,
___
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Hi,
I am trying to build zaptel 1.4.8 with kernel 2.6.24 on debian/sid:
zenon:~# module-assistant -t build zaptel
make[3]: Entering directory `/usr/src/linux-2.6.24.3'
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/modules/zaptel/Makefile. Fix it to use
On Thu, Feb 28, 2008 at 11:10:37AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
zenon:~# module-assistant -t build zaptel
make[3]: Entering directory `/usr/src/linux-2.6.24.3'
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/modules/zaptel
Hi,
Returning to my office I find two missed calls (from autodialers) that
my IP601 displays as originating from 011. However the CDR
database recorded the call this way:
calldate: 2008-04-04 14:18:16+02
clid: 0172752780
src:
Hello and sorry for the OT,
Is it possible for a wireless headset of which the base is connected to
a Polycom IP601 to remotely answer a call? In the same way as a
bluetooth headset.
thanks,
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Hi,
Using asterisk 1.4.21.2.
For some calls (usally telemarketers) entering through a BRI zap channel
I somtimes notice the callerid on my polycom 601 phone and the CDR's
'src' field don't match. They are even totally different. And the
displayed callerid is nowhere to be seen in the CDR record.
On Wed, Oct 15, 2008 at 11:30:49AM -0500, Tilghman Lesher wrote:
On Wednesday 15 October 2008 10:26:50 Louis-David Mitterrand wrote:
For some calls (usally telemarketers) entering through a BRI zap channel
I somtimes notice the callerid on my polycom 601 phone and the CDR's
'src' field
On Mon, Apr 16, 2007 at 12:25:55PM +0200, Bas van der Veen wrote:
Hello,
Did you find anything while testing the LAN? Also, can you confirm that
switching the switch, cabling, etc. did NOT solve the problem?
It did not.
We finally changed the server itself and reinstalled from a
On Wed, Apr 18, 2007 at 01:04:31PM +0200, Tim Koehler wrote:
Hi,
are you using PoE or power supplies?
As power supllies usually are not grounded it could be that it's comming
from the power source.
We are using PoE
You could try using a grounded PoE switch or probably a power backup to
Hi,
a block of my extensions.conf no longer works after upgrading from
1.2.17 to 1.4.4. I have:
[macro-dialout]
exten = s,1,Gosub(s-${ARG1},1)
exten = s,n,Congestion
;; default
exten = _s-!,1,Gosub(s-NET,1)
When calling that macro whith no argument
Hello,
I'm using the classic [stdexten-macro] in extensions.conf whereby a call
is picked up by voicemail after a certain ringing time.
When programming a SIP phone to redirect calls (SIP 302 redirect) to
another extension I'd like to avoid that voicemail pickup so that the
call goes into the
On Tue, Mar 06, 2007 at 07:18:08AM -0600, Eric ManxPower Wieling wrote:
How can I detect that a call has been redirected and should no longer be
intercepted by vm?
That should happen by default. The call should get sent to the new
place and it should act like the call was directly
Hi,
At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.
Has anyone seen that?
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On Wed, Mar 21, 2007 at 07:07:00AM -0400, joe a. wrote:
Did you swap the power module as well? If POE, did you swap the
patch cord?
If the power module plugs into a power strip did you change that? or
at least the position in the strip?
Thanks for the tought, but the IP430 has no
On Wed, Mar 21, 2007 at 04:07:34AM -0700, Henry Cobb wrote:
On 3/21/07, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:
Hi,
At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.
Has anyone seen
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just swicthed to it.
What kind of test tool would you suggest?
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just swicthed to it.
Was that phone using POE ?
Hello,
After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I
no longer match any extension. Apparently the dad is empty. However I
can see the number just before it (146472130):
P[ 4] I IND :SETUP oad:!?145201798p
¡146472130 dad:
¡146472130 pid:2
Hello,
We are at wit's end on this. One (and only one) of our five asterisk
installation is giving us real headaches. Buzzing and/or choppy sound
interfere with conversations. I recorded some conversations with
monitor() and no problem whatsoever appear in the recording, while the
local user
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
read(0, unfinished ...
+++ killed by SIGSEGV (core dumped) +++
Process 15755 detached
On a second
Hi,
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4 min)
This is triggered by the monitoring app sending a POKE to the iax port.
The warning appears
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
Your monitoring app is not sending valid IAX2 packets to the
server. If
it was sending a true IAX2 POKE, it would be a valid packet and
wouldn't
generate this warning.
Could asterisk at least _not_ report this harmless,
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get
On Thu, Nov 06, 2008 at 11:46:48AM +0100, Louis-David Mitterrand wrote:
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
read(0, unfinished
Hi,
I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by zap show channels. I tried adding
dahdichanname = no to asterisk.conf's [options] to no effect.
Going back to
On Tue, Nov 11, 2008 at 09:49:14AM +0100, Louis-David Mitterrand wrote:
Hi,
I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by zap show channels. I tried adding
Hi,
How hard is it to integrate asterisk with Microsoft CRM?
Thanks for any suggestions, pointers, etc.
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On Wed, Jan 21, 2009 at 09:02:51AM -0200, David fire wrote:
how hard is to integrate whit a virus?
sorry
ok i read MS CRM but... did you tried VTiger? www.vtiger.com the next
release (5.1) will be integrated whit asterisk not only click to dial and
popups on incoming calls a queue monitor
On Wed, Jan 21, 2009 at 12:58:51PM -, Andrew Thomas wrote:
Try http://forums.vtiger.com/viewtopic.php?t=14314
Thanks, this is a really interesting link.
--
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On Wed, Jan 21, 2009 at 09:00:41AM -0500, Jon Weisman wrote:
ok what about people that have no choice but to use MS CRM?
That's also my concern, as MS CRM is my customer's choice, not ours, and
I may or may not succeed in steering them toward an open-source solution
such as vTiger. They already
On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote:
I use them both; my legacy dialplan is all .conf and new stuff is .ael.
I find AEL to be the better option when jumping around, but that's
just my opinion.
But isn't AEL just converted into .conf language anyway? Or has this
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better option out there?
Thanks,
___
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On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote:
Louis-David Mitterrand wrote:
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Thanks,
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On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Digium's BRI cards are also based on Cologne Chip - thus you could try
Digiums BRI
On Tue, Jun 09, 2009 at 04:04:29PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
mISDN
Hi,
After upgrading to 1.6.x and hdvoice (g722) polycome phones I am
wondering how to optimize asterisk sounds and music on hold to take
advantage of that codec. I often listen to a special music extension on
my headset:
/usr/bin/wget -q -O - http://music.example.com | /usr/bin/madplay -Q -z -o
Hi,
I have a 2/3 second gap between the end of a welcome message played with
Playback and the start of the Queue music. Here is the dialplan:
exten = ${EXTEN},1,NoOp($EXTEN)
exten = ${EXTEN},n,SIPAddHeader(Alert-Info: Ring_CCC)
exten =
On Wed, Jun 17, 2009 at 01:08:33PM -0500, Danny Nicholas wrote:
If this is a recorded sound, you might want to truncate it with lame or
audacity. It is quite common in my shop as we record using the phones.
Thanks for this suggestion.
The problem was indeed a silence at the beginning of my
Hi,
Is there a way on Polycom phones to show an agent whether he is logged
in or not?
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Hello,
Has anyone noticed that the Polycom IP430 has a low incoming/outgoing
sound level?
Is it a firmware issue or should I adjust my zap's tx/rxgain?
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To
Hello,
Since our telco messed with our PRI in some way, we get corrupt faxes
like these:
http://zenon.apartia.fr/stuff/corrupt_fax.pdf
We use the lastest asterisk with a TE410P and spandsp.
(for some strange reason, our neighbour company has a traditional pbx
fed by 7 BRI's and sees the same
Hello,
We have a TE410P connected to an EuroISDN E1 with these span
definitions:
span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3
span=3,1,0,ccs,hdb3
span=4,1,0,ccs,hdb3
Why should we add crc4 to these definitions? What does it do?
Thanks,
Hi Kape,
With latest asterisk 1.2.12.1, zaptel 1.2.9.1 and bristuff 0.3.1s after
a while calls become stuck: either the caller or callee can't hear the
other party, or heavy static is heard. An asterisk restart fixes it for
a short while only.
This doesn't happen with our older installs
Hello,
When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to
re-register themselves with asterisk, even though I put
timer_register_expires: 60 in SIPDefault.cnf
Is there a way to have these phones register themselves every 60
seconds?
Alternatively, can asterisk be made
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained).
Hi Kape,
Life is generally good with bristuff and the quadBRI cards. However I've
got a concern: how does one return a busy signal to the telco when all B
channels are busy? Right now, when all channels are in use, the remote
caller is kept waiting until the telco times out and finally get a busy
Hello,
I have the exact same question as you. Did you find an answer?
We are using asterisk at the office and the incoming line is an ISDN
(HFC-PCI card with zap_hfc driver from bristuff 0.2.0 RC3a).
And I have a problem, when both ISDN B channels are in use (i.e. 2
calls in progress) it
Hello,
With the 1.5.2 firmware, have you managed to get one-touch message access when
pressing the Messages button? It worked for me with 1.4.1 but no longer with
1.5.2: I have to go through the message count screen first.
In phone.cfg I have:
msg msg.bypassInstantMessage=1
and in
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote:
With the 1.5.2 firmware, have you managed to get one-touch message access
when
pressing the Messages button? It worked for me with 1.4.1 but no longer
with
1.5.2: I have to go through the message count screen first.
In
Hello,
I'd like to find a way to probe a SIP phone for forwarding information
before I actually Dial() it. For instance, if an absent user entered a
forwarding number in his (Cisco or Polycom) phone, it will anwser a
Dial() with a REDIRECT and asterisk will comply if the context allows.
However
On Thu, Aug 11, 2005 at 07:12:32PM +0100, Mark Thorpe wrote:
I have been trying to solve a problem wherby when I boot a cisco 7920 my
7940 seeks a new IP and the dhcpd log shows it released its existing IP. In
searching for the solution I notice there were 2 messages on this list in
Aug Sep
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Paul Hales wrote:
I battled with chan_capi during the week, and it was not fun.
Since I'm working on chan_capi, I would like to know what problems exist.
Can you please be more specific on what problems
On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Louis-David Mitterrand wrote:
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote:
On Fri, 13 May 2005, Paul Hales wrote:
I battled with chan_capi during the week, and it was not fun
On Fri, Jun 03, 2005 at 02:39:48PM +0100, Gavin Hamill wrote:
On Friday 03 June 2005 14:28, Nardis Dome wrote:
--- Brett, Gary [EMAIL PROTECTED] wrote:
Is the Eicon that much better ?
sorry, i have only experience with Eicon... maybe
someone else is able to give a feedback...
Aside
Hello,
I looked everywhere in the docs and in google but couldn't find an
answer.
Is it possible to localize the output of ${VM_DATE} (say, in french) ?
--
Only half the people in the world are above average intelligence.
___
Asterisk-Users mailing
On Fri, Jun 17, 2005 at 10:34:25PM +0200, Conrad Beckert wrote:
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference FM is :-) )
Has anyone an idea on how to disable the console sound driver. My problem is
that a running asterisk is muting
Hi,
After upgrading to 1.5.2 I no longer can directly access to my voicemail
by pressing the Message button, I have to go through the
urgent,new,old report first. The oneTouchVoicemail parameter is set to
1 but not taken into account apparently.
Anyone noticed that problem?
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