Call ServiceMaster :)
Depends on how much charge was left in the circuit as to what will
happened. If it was saltwater, probably not. Freshwater, there might
be a chance that after it dries completely that it will come back
online. Won't know until you can test it.
Glad you and your family is
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in
Mr Richardson,
I sympathize with american people after this disaster.
However If i was God I would feel remorse for all
people in the world in destitution because of
diseases, wars, starvation, ...
God should really feel remorse .
Thinks to all people in destitution in the world .
Harry
---
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used
I am rewriting IPSwitchBoard at the moment. I want to
make IPSwitchBoard Skinable meaning that you can design your own
skins with company logo etc. You will also be able to add graphical extension
buttons, and leds that will light up ex. DND, busy/free, message waiting
and much more.
If
I know almost nothing linux, and don't really know that
much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an
[EMAIL PROTECTED]
installation CD (version 1.3) and just went with it. Newbie doesn't quite
describe it, I'm a banker.. this simply goes to show how easy
It is good to see JR's faith is not shaken by then even and that he will
rebuild better than ever.
I hope JR's Asterisk application development was backed up somewhere so he
doesnt have to re-write it all again.
I have seen drives recovered from being soaked in water and it wasnt a
major
A new release of the open source G.729 patch has been issued.
The new URL is:
http://www.readytechnology.co.uk/open/ipp-codecs
The memory leak in codec_g729 is now fixed. This was due to a
problem in a section of code copied from the Intel example. Thanks
to those who assisted in
I have bought and collected equipment since being in
Telecommunications, VoIP and Internet Technologies for 15 years that
are irreplaceable but I will re-build my VoIP laboratory bigger and
better than ever. If anyone has any trade secrets on successfully
recovering waterlogged electronic
If anyone has any trade secrets on successfully recovering waterlogged
electronic equipment, please let me know.
Dear JR,
I am realy sorry about all this desaster, but happy to see you alive.
For waterlogged equipments, no problem until they are under the water level
(out of oxygen contact).
HI All,
Have searched a bit for this one, but cant seem to find any indication
of how you might use a dial prefix with the hash / pound sign as part
fo the prefix.
I have a gsm pod set up on asterisk which all works fine, but i want
to disable cli on the outbound calls over this particular
Okay, here is the background. I have a PRI with 15 active channels on
it. I originally setup all of them in group=1 and all outgoing and
incoming calls used this group. The phone number that I have associated
with these channels ends with 750 and that is how I direct the calls.
i.e. In my
Hi!
Is there any SIP hardware phone that provides an API that can be used to
control and monitor the phone by external applications?
Thanks in advance!
___
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Asterisk-Users mailing list
AbdelRahman Tarzi wrote:
I know almost nothing linux, and don't really know that much about
Asterisk (proper).. but I was 'pulled' by this subject and grabbed an
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] installation CD (version 1.3) and
just went with it. Newbie doesn't quite describe it,
In v1.2beta1, README.sms mentions that SMS() allows sending and
receiving of text messages over the PSTN (and that it's not likely to
work over a compressed link.) I think this likely answers my
following question, but I wanted to ask because maybe someone has
accomplished this:
Is it
Hey folks,
The reason I ask, is I'm on my way down to north western LA to
donate/help setup gear from a WISP used to own/operate.
VOIP is being mentioned quite a bit. So if you have any spare gear,
please donate it.
We are using this site to coordinate I'm sure there are others:
This is something that you wouldn't have any control over in most
cases. Your provider is the one that is sending the lines over the
whole PRI, they don't and shouldn't care about which channel it comes
over, they would use some system like gGrR that you can use in
Asterisk to send it to the next
For waterlogged equipments, no problem until they are under the water level
(out of oxygen contact).
The best way after that will be to clean all of this waste and poluted
equipments with clear water during a long time (no more chimical products in
the clean water), and to quickly dry them.
Depending on the mother board. The following might or might not help:
1. Move the card to a different PCI slot, some motherboards will
assign you a different IRQ based on the PCI slot, while other
motherboards might just have one PCI slot that gets its dedicated IRQ.
Changing it around will tell
Sorry to write so many consecutive messages in such a short period of
time, but this problem is really bugging me as it has been going on for
days.
When I look in Ethereal, there are actually two calls going on -- in
this particular call, Source call #4 and Source call #10318, #4 coming
[EMAIL PROTECTED] wrote:
If anyone has any trade secrets on successfully recovering waterlogged
electronic equipment, please let me know.
If it's not electrically broken, there is a big chance to save it ...
open it, and wash it using hand shower and some brushes until it
is clean (the best is
On Sun, 2005-09-04 at 07:39 -0500, Derrick Stensrud wrote:
Re-sending your message every 12 hours isn't nice wait at least a
couple of days, and while you wait, try to read/test more things, so
that the second time around, you can actually demonstrate that you have
progressed somewhat
Op zondag 4 september 2005 11:10, schreef Stijn Jonker:
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Hi Guilhermo.
Could you share with us your experience?
What is the hardware(CPU, RAM, etc) that are you using for this server?
What is your Linux distribution?
How many concurrent calls do you have in the high traffic moment?
Which is the unicall version that are you using?
Thanks a lot!
Hello,
I am getting lots of messages like that:
Sep 4 15:31:55 WARNING[10271]: chan_sip.c:946 __sip_xmit: sip_xmit of
0x835bad8 (len 758) to xxx.xxx.xxx.xxx returned -1: Invalid argument
Could somebody tell me more about that warning? I would appreciate it.
I could not find anything
Hi,
Does anyone have some experience with Nokia 32 Terminal (it is an analog
GSM Gateway)? After a configuration I can make only incoming calls, I'm not
able to do any outgoing. Nokia signalize an error (4 short tones), when I try
to phone someone. I tried postpaid simcards as well as
Interestingly this phone works fine if I configure it for broadvoice
directly (bypassing my local AAH).
All the software sip phones seem to work fine with aah. The phone
does not seem to know what to do if I leave the outbound sip proxy
fields blank. And itt probably does the wrong thing if I
Jeroen,
On 04-Sep-2005 19:34, Jeroen Baten wrote:
Op zondag 4 september 2005 11:10, schreef Stijn Jonker:
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what
On Sun, 4 Sep 2005, Stijn Jonker wrote:
On 04-Sep-2005 19:34, Jeroen Baten wrote:
Op zondag 4 september 2005 11:10, schreef Stijn Jonker:
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21,
Sorry to bug all of you with this, but I have to assume there are a
number of Sipura experts here...
I have a Sipura SPA 2000 that I've been using for nearly 2 years now.
It's flashed up to firmware 3.1.5.
On line 1, I no longer get Caller ID (it used to work, and I can't
remember when it
Hi all,
I am new to asterisk and I can not find any detailed info on using SIP
MySQL support (sipfriends) with clients behind NAT. I've heard that I
have to patch chan_sip.c and Makefile to get it working.
I tried on voip-info.org but found no answer for my questions.
I found some answer on
Can someone send me a copy of the sipura spc.exe used to compile
centralized configs via private email?
I've applied for their access but haven't been approved just yet.
Need to do some testing with it over the next few days.
Rich
___
--Bandwidth and
Hi all,
I'm trying to setup a simple IVR menu in a context in extensions.conf. So far, I have:
extension s for playing back the menu
# to repeat it
* for directory
0 for operator
1 which goes to another context: exten = 1,1,GoTo(option_1,s,1)
Here is what I have in extensions.conf:
[incoming]
;
Hi to all,
I need help to setting up messagenet.it account in Asterisk.
My * is connected with static IP to the net.
No other cards at the moment, just the network-card.
I'm able to receive call on the geographical number, but I'm not able to setup the outgoing calls.
All that it does is:
I dial
On Sun, 2005-09-04 at 14:09 -0700, Adrian A wrote:
Hi all,
I'm trying to setup a simple IVR menu in a context in extensions.conf.
So far, I have:
extension s for playing back the menu
# to repeat it
* for directory
0 for operator
1 which goes to another context: exten =
Thank you (for spamming) - it was the clue I needed to push this through.
Sorry it took me a while (and a google :-) ) to realize you'd addressed my
initial query - basically, my loss.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent:
Hello,
What's the status on using eyebeam with Asterisk, does it still
require a patch to Asterisk to support the video component? I'm
intererested in starting to use Video and audio telephony but wary of
anything that requires patches.
cvs head works out of the box, just enable the h.323+
I use a Nokia 32 as an extension and as a trunk (two sites).. Which are you
doing ?
If you wish to connect it to an FXS you will need a special cable which
Nokia sells..
Connecting to an FXO (which expects a line) is the default.
Check the normal stuff (like dialstring) before you suspect the
Thanks again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Sunday, September 04, 2005 18:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: [EMAIL PROTECTED] - requesting help on
Hi,
Does anybody have an updated Chan Unistim that compiles on Asterisk
1.2beta?
Below is the output when compiling on Red Hat 9.0
Thanks,
[EMAIL PROTECTED] chan_unistim-0.9.2]# make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
-I/usr/include -D_REENTRANT -D_GNU_SOURCE
Hi,
When using asterisk real-time with mysql voicemail integration...
where exactly do I put the options like the [PBX] tag, and how long
silence can be, etc?
___
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Asterisk-Users mailing list
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
Anybody having issues with ztdummy under the current 2.6 RC7? I get the
following errors when trying to modprobe ztdummy:
Unable to register zaptel rtc driver
Doing a Google on the error shows
I should add to this... I understand to make the table.. but when I
make it.. asterisk selects it but seems to ignore things. No where
have I found documented what the var_category and such are... what
numbers do I put in there?!?!
On 9/4/05, Matt [EMAIL PROTECTED] wrote:
Hi,
When using
Rich Adamson wrote:
Sorry to write so many consecutive messages in such a short period of
time, but this problem is really bugging me as it has been going on for
days.
When I look in Ethereal, there are actually two calls going on -- in
this particular call, Source call #4 and Source call
You need to care about the _actual_ error, not the report there is an
error. The error is (usually) reported to the console. Reboot the
computer and type this:
dmesg -c /dev/null
modprobe ztdummy
dmesg
The output of the second dmesg will show you exactly what the error
message is.
Being that
I updated our system here to the latest CVS (we were previously running
25/08/05) and the lights work perfectly now.
PaulH
CAUTION: This email message and accompanying data may contain information that
is confidential. If you are not the intended recipient, you are notified that
any use,
I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates).
The system has a TE110P card, and zaptel.conf is configured for an E1.
When I do a 'zaptel stop' I get a kernel panic.
Has anyone else seen this?
Thanks,
___
--Bandwidth and
Perhaps I read that wrong, as the different filters don't seem to show
those packets as lost or jittered. Plus, this crackle happens fairly
often, so I don't know if it's any indication of dropped packets (or the
root of the problem, at least), but then again, I'm not sure. I will try
Rob Thomas wrote:
So. Turn that off, and recompile the rtc module and it'll start working
Thank you very much!
Doug
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi all,
Has anyone had problems with not being able to hear callers
and them not being able to hear you? And had any success on how to fix it? Our
call centre staff are complaining that this is a continual problem.
Appreciate any thoughts on this.
Regards
Jennifer Hales
You must store voicemail.conf using RealTime Static in order to use the
options you have mentioned from database.
-Matthew
From: Matt [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sun, 4 Sep 2005 19:51:36 -0400
To:
How did you convert your voicemail.conf file into RT Static? Did you use the
perl script?
-Matthew
From: Matt [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sun, 4 Sep 2005 20:37:34 -0400
To: Asterisk Users Mailing
Guillermo,
Switchtype depends on to which kind of PSTN are you connected to.
Are you connected to Telecom or Telefonica?, using PRI or FXO/FXS lines?
Normally both follows European Standards for Telephony (CCITT), not Bell
standars.
And in the case of Telecom they have a lot of Telettra
Guys.
How are filenames determined for automon and queue recordings enabled on
queues.conf?
I see the names have some tomestamps or something but is there a way to
predefine the filenames to use?
Thx!
___
--Bandwidth and Colocation sponsored by
I am attempting to assemble a proposal for a client of mine that is
looking to replace their phone system. I think it's a good first
installation with 4 POTS incoming and 15 extensions, with an overhead
paging system. I also think that it would make a good case for OSS
applications in general.
I've read alot on the wiki about sending and receiving faxes thru asterisk.
I've gotten the receive to work great.My question is how does one send a
fax?
I see lots of instructions about how to send the image to asterisk by email,
etc. The problem is how does one make the image of the
On Mon, 2005-09-05 at 01:31 -0400, Kurth Bemis wrote:
I am attempting to assemble a proposal for a client of mine that is
looking to replace their phone system. I think it's a good first
installation with 4 POTS incoming and 15 extensions, with an overhead
paging system. I also think that
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