Yeah, I have CVS-HEAD as of today and it's still no good. Thoughts?
I have CVS-HEAD as of yesterday and it's still not working for me. Maybe
I'll try updating again and post the results.
Thanks,
Tim
[EMAIL PROTECTED] wrote:
Hello everybody. Recently I've been trying to limit the
Here is some info that may allow some locked PAP2 and
PAP2-NA units to be used with Asterisk:
I have a PAP2-NA (from a provider other than Vonage) for
which I did not know the admin password, though the user
pages were accessible to me. The provider had set it up to
fetch at startup, its
Hi all;
i want to make an asterisk server as a SIP provider to other asterisk servers
so other servers register a trunk to the asterisk SIP provider.
can this work? how can i implement this?
Powered by Hellacious Riders - http://www.hriders.com - Home of the Free 1TB
email accounts.
Want to
To start ztmonitor
in quantitative mode you do the following.
Assuming you are
running Asterisk V1.0.9 you need toedit one line to ztmonitor.c in
/usr/src/zaptel as per patch 2783
http://bugs.digium.com/bug_view_page.php?bug_id=0002783
Change line
261
fprintf(stderr, "Usage: ztmonitor
On Sun, 2005-08-21 at 17:06 -0700, jennyw wrote:
The Digium cards actually are sharing IRQs with other devices -- the
installer mentioned it could be an issue initially, but when he saw that
the devices that the cards were sharing with were the network card and
the video card, he said to
Hi:
BTW, note that unless you do some serious hacking, the fine print in their
ad will certainly hold. At the bottom of the page, it says:
* The VTech IP8100-2 requires Vonage service to operate and is not
compatible with any other service.
Scott
On Sun, 21 Aug 2005 13:12:49 -0600,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I have 2 outgoing calling companies. Say A and B.
With A company, I'm given 250 Minutes free per month.
I would like to start off by using A company until 250 minutes were
depleted for that month then use B and of course switch back to A once
At first, thanks for an answer.
Well the problem starts at the side of a cell phone operator. In the IAM
this operator sends ALERTING, eventhough the cell phone is switched
off. It means, that the gateway sends to the asterisk 180 with SDP and
also with the RTP stream, that contains the
On Monday 22 Aug 2005 04:30, root linux wrote:
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
You do not appear to be in the US but Malaysia. Not sure what these
should be.
My zapata.conf config: -
Hi Michael,
Can I ask roughly how much the Hitachi wifi phone is, and where can you get
them?
Thanks,
- Andre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Sunday, August 21, 2005 9:58 PM
To: Asterisk Users Mailing List -
Joshua Abbott wrote:
Hello,
I have 2 outgoing calling companies. Say A and B.
With A company, I'm given 250 Minutes free per month.
I would like to start off by using A company until 250 minutes were
depleted for that month then use B and of course switch back to A once
the next month turns
Hi,
anyone can write down a working example of a regex fuction ?
I'm using this syntax
Gotoif($[${REGEX(/B/ | A)}=1]?20)
But function always return 1, even if I write
Gotoif($[${REGEX()}=1]?20)
Tnx for any help !
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
There is a fee, but I believe you can call Vonage and get a box
unlocked after you are done with the service. If I'm remembering
right, the fee is about $10.
BTW, per http://forum.openwrt.org/viewtopic.php?id=1643
try
user: user
pw: tivonpw
For the web interface. You might be able to unlock it
This sounds suspiciously similar to a problem I've had at a customer
location.
Telco is running 5ESS switch, Asterisk is using TE410 board.
Asterisk CVS HEAD as of mid july.
Most incoming calls do not have the Caller Name appearing. Caller ID
number always comes in.
On almost every
canuck15 wrote:
To start ztmonitor in quantitative mode you do the following.
Change line 261
fprintf(stderr, Usage: ztmonitor channel num [-v] [-f FILE]\n);
to
fprintf(stderr, Usage: ztmonitor channel num [-v | -f FILE]\n);
Err.. changing that does absolutely nothing to the active part
Using a spa3000 with asterisk cvs head, and the spa3k is config'ed with
a dialplan that essentially routes any call starting with an 8 to
asterisk. All other US 7 and 10 digit calls, 911, etc, route via the
spa3k's fxo port.
Is there a way in extensions.conf to:
- inspect the dialed exten
Hi Michael,
What phones are you using as this will affect your implementation. For
example do you want to dial zero, then hear a dialtone and dial the full
number or do you wish to dial the whole number with a preceeding zero in one
hit?
Craig
- Original Message -
From: Michael
Hi Tommy,
have you seen the Asterisk @ Home distribution? IMHO the easiest way to
install AMP.
Craig
- Original Message -
From: Tommy Denton [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 22, 2005 10:07 AM
Subject: [Asterisk-Users] perl-cpan
Dear
Hi rootlinux,
I'm in Australia where we also uses crc4 on the span line, could you also
show the relevant section of your zapata.conf? Looking at your
extensions.conf excerpt, it is customary to group the b channels eg
Dial(Zap/g1/12345678) and there should be an entry in zapata.conf under
Hello -
I have just purchased an Aastra 9133i SIP phone for
testing with Asterisk. Its a little flakey but overall is a far superior
phone to the others in the $179 range.
I have an issue regarding the message waiting
indicator. The phone does not seem to respond to the "NOTIFY" command
Hello -
I have just purchased an Aastra 9133i SIP phone for testing with Asterisk.
Its a little flakey but overall is a far superior phone to the others in the
$179 range.
I have an issue regarding the message waiting indicator. The phone does not
seem to respond to the NOTIFY command from
Hello,
I have several *
serversbehind a SER server (in a local ip range).The
SERserveris also publicy reachable. On the other site, I have SIP
clients that are behind another NAT or in the same NAT range as the * server.
Can someone give me some directions/hints etc. on how to make this
Polycom most definately should add a provision for off hook voice
announcing. It would be nice to have a receptionist announce a call for you
over the speakerphone while you are using the handset. I don't see why it
would be anything more than a programming limitation, but then it becomes an
I could only get *ANI*DNIS* working one way and that was setting my
signalling type on the Asterisk side to 'featd' The Definity won't send
*ANI*DNIS* information back to the asterisk as far as I can tell.
Other than that, I've been running it with wink/wink EM for a while
now. TN464 circuit
Hi,
all your phones need to be registerwith SER. The
asterisk will be just PSTN gateway, voicemail server or something else (I prefer
to forward all the calls from ser to asterisk because it's easy to manage the
dialplan). I have the same configuration,I balance the traffic with SER
and I
Same problem as before. The features.conf has the default settings of *1
the this and *2 for that, and the # for transfers. They are uncommented.
They do not work for me.
And if this is something I'm doing wrong (I hope), I would really prefer a
double-# strike in quick action, since so
In the US they are distributed through ABP Tech near Dallas TX, but I
bought mine from www.voipsupply.com for $319.00
Michael
On Mon, 22 Aug 2005 06:45:12 -0400, Andre Normandin wrote:
Hi Michael,
Can I ask roughly how much the Hitachi wifi phone is, and where can you get
them?
Thanks,
Im using a TE110P as a
trunk to a Panasonic KD-500 everything works well.but Im having this
problem where one of the channels becomes blocked with a partial phone number
after about two days.So if the channel that becomes blocked is
channels 23 no calls can get in. If the channel that
Lars Dybdahl wrote:
I would like to know how to install asterisk 1.0.9 with zaphfc working
on a SuSE 9.2.
Any ideas?
Forget RPM.
First of all read:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+SuSE
then download
http://www.junghanns.net/downloads/bristuff-0.2.0-RC8n.tar.gz
try something like this.. the EXTEN:1 strips the leading 8 off
exten = _8.,s,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},60,Ttr)
On Mon, 2005-08-22 at 05:43, Rich Adamson wrote:
Using a spa3000 with asterisk cvs head, and the spa3k is config'ed with
a dialplan that essentially routes any call
Although I'm convinced that Broadvoice doesn't have the most stable of
ping times, it seems like I get ping results that are approximately the
ping time +2000ms at times. Has anyone experienced this problem with
qualify on a SIP connection before?
So here, was the ping 20ms or 2020ms as
I have some netweb 302 phones which we used when we first started
evaluating Asterisk. These phones would do this all the time. It seemed
to have nothing to do with the network ping time. I never really did
check in to what was going on because it only happened with those phones
and we don't use
Sherwood McGowan wrote:
since delete is a reserved word, what do you name a column in your
voicemail options table to allow setting of the delete option for
realtime voicemail? Anyone?
[delete] should work, or on some databases 'delete'.
Tony
___
Isnt Firefly and for that matter any other IAX2 Softphone an IAX2
Endpoint in real sense?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Walsh
Sent: Saturday, August 20, 2005 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial
This newest version Includes several bugfixes, and optimizations The
last release (0.3) turned out to be broken, so everybody using it or
tried using it, should upgrade.
Many thanks to Cliff and Julian for their feedback.
Screenshots and download link at:
Just if anyone is interested, the asterisk-uk community website is now
up. If you're a voip company or service provider operating or based here
in the UK, please add your details to the site. If you're an asterisk
user or admin, please signup and help us build content for the uk
community.
The
I have (sort of) made this work in our environment.
There was an AGI script (Google for polycom allcall.agi) written in Perl
that would implement a hack to do this. Basically, you can set the
Alert-Info SIP header and cause the Polycom phones to auto-answer. The
general idea behind
I did something like this I had to work on the perl to get it working
myself.
I found if I put a wait and a beep in the dial plain for the calling user
then
they would get beep 3 or 4 seconds later this give the calls time to set up.
Worked great for me.
-Original Message-
From: [EMAIL
Is the php script available somewhere?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Gault
Sent: Monday, August 22, 2005 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] All Page ??
I have (sort
Unfortunately, documentation from polycom is kind of obscure, all 301 and
501 plus 600 models say they support XML, but only 600 has the microbrowser
so only 600 can support pushing XML.
So I guess after asking a lot of times.. Only polycom 600 and Cisco phones
using skinny support XML push
Hello,
I am having an issue with hangups being handled within Asterisk. Right now,
when an inbound call hits
the Asterisk box, Asterisk picks up the call just fine.
When the caller enters an extension to call, the Asterisk dials out on Zap/3
and rings the extension with
no problem. If the
Thanks, Lance.
The two softphones and the Asterisk server are on the same subnet.
There is no NATing here. I'll turn on the show peers option in sip.conf
and iax.conf to see if that help.
I also have a hardphone connected to FXS on the asterisk server, it has
no problem talking to either the
Specify a different context for each Zap channel (context=homephone)
(context=workline) in zapata.conf instead of just inbound-analog. Then
in your extensions define a context for each that includes a different
dialplan.
On the second problem, you could remove the forward from the Verizon
Dave Cotton wrote:
This was your first experience with *, was it also the installers?
Only sharing with the next busiest card in the machine the one feeding
the IP phones.
Yeah, I know, in retrospect it sounds really odd that we did that, but
at the time he thought there was a chance it
On Mon, 22 Aug 2005, Guy C. Guckenberger wrote:
Im using a TE110P as a trunk to a Panasonic KD-500 everything works
well.but Im having this problem where one of the channels becomes
blocked with a partial phone number after about two days. So if the
channel that becomes blocked is
I'm running SIP 1.5.2 firmware with a 2.6.2 bootrom on a mix of Polycom
500s and 501s. I'm having a problem with NTP and I'm not sure if it's a
configuration issue or a bug in the firmware.
I've got the NTP server and GMT offset set in sip.cfg. However, not all
phones are in the same timezone. I
Hi all,
I have a SPA 1001 with DTMF set to auto. A sip.conf peer with codec
alaw, ulaw, g729. I have dtmf=inband as this peer was only supporting
alaw/ulaw. They just add g729, it's a GW to landline phones.
A call passing through this peer give me WARNING Inband not supported
with g729, use
jennyw wrote:
AMD Sempron 2400+
3Ware Escalade 8006-2LP (2 channel SATA RAID)
Asus A7N8X-E Deluxe
1 GB RAM
Jetway Radeon 9000 64MB 128Bit 4X AGP DVI/TV Out Dual Head
This is a bad sign... are you running a graphical environment on this
machine?
I'm not sure if I can change the IRQ setting
Geia Sou my friend. I am getting consused here.
When I am using pridialplan = unknown I am getting the following messages
Display (len= 2) [ IZ ]
[6c 03 21 81 33]
Calling Number (len= 5) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Paul Crick wrote:
I've got the NTP server and GMT offset set in sip.cfg. However, not all
phones are in the same timezone. I was under the impression that I could
override the offset on a per-phone basis (and would expect it to update the
-phone.cfg file). This doesn't seem to work though.
I just upgraded to Fedora Core 4 and Asterisk won't run any more. When
launching asterisk, I get asterisk: error while loading shared
libraries: libssl.so.4: cannot open shared object file: No such file or
directory.
A quick search (find / -name libssl.so.4) for the file shows the file
nowhere
jennyw wrote:
Dave Cotton wrote:
This was your first experience with *, was it also the installers?
Only sharing with the next busiest card in the machine the one
feeding the IP phones.
Yeah, I know, in retrospect it sounds really odd that we did that, but
at the time he thought there was
In the 1.4.x firmware release, there was a config file setting that would
cause the screen focus to change to a new call that came in whilst a call
was in progress. When set, presentation of a new call would cause that call
to be selected, and the softkeys would display Answer, Reject, Forward.
Or IRQs can be set via ACPI
On 8/22/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
jennyw wrote:
AMD Sempron 2400+
3Ware Escalade 8006-2LP (2 channel SATA RAID)
Asus A7N8X-E Deluxe
1 GB RAM
Jetway Radeon 9000 64MB 128Bit 4X AGP DVI/TV Out Dual Head
This is a bad sign... are you
Did you recompile everything * after your upgrade?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Stahl
Sent: Monday, August 22, 2005 10:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk 1.0.7 won't run after
On Sun, Aug 21, 2005 at 09:07:37PM -0500, Tommy Denton wrote:
Dear list,
I was installing Asterisk via the AMP method off the AMP website.
There is a portion in there where they want you to use perl-cpan to
install telnet.
What distro do you use? Doesn't it provide its own pre-packaged
I need to execute account number, device number after dialing main
number; what is the best solution?
Is it possible to pause during dialing.
Dial 1-800-number
press 1 for English
wait 5sec enter device number
wait 5sec enter device ID
What are my best options?
--
#Joseph
I'm trying to use the fxotune tool and I always have this error message:
Tuning module 1Failure!
Tuning module 2Failure!
The first thing I would need to know is what is the meaning of the -i
option. The tiki said that it is the number to break the dial tone, but
Paul Crick wrote:
Has anyone else noticed this or had experience with it? We like the
features of the new firmware release (like being able to split a conference
call then drop one party) but the change in behaviour of call waiting is
giving a few users grief.
Look at the 'callwaitingprompt'
After more
investigation, I decided to just recompile asterisk (on my newly upgraded Fedora
core 4 system). Make dies with this error:
"No rule to make
target
'usr/lib/gcc/i386-redhat-linux/3.4.3/include/stddef.h"
It seems this
directory is gone under FC4, and replaced by
No rule to
I say start small and then go big... Oh I don't know a Proliant 1500 or
3000 should work nicely -- if you can handle the noise =)~
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Latham
Sent: Monday, August 22, 2005 1:40 PM
To: Asterisk
On Monday 22 August 2005 13:29, Kevin P. Fleming wrote:
No PCI card can 'change IRQs'. IRQ assignment is under the control of
the motherboard wiring and the BIOS settings.
Actually that is untrue. There are four INT# lines on a PCI bus.
*everything* uses INTA# and relies on the chipset and
Greetings,
I am trying to get either of the above features to work with *, but
can't seem to get it quite right. If anyone has them working, I'd
sure appreciate an extract from the relevant config files.
Or, maybe I'm tilting at windmills, and * doesn't support them - in
which case, the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here's my public PGP Key
- --
=
Joshua Abbott, Support Technician
http://www.successfulhosting.com/
Direct Line: PENDING
Phone: (866) 494-5096 x1207
E-Fax: (419) 858-3241
Alt E-Fax: (801) 217-1123
[EMAIL
Hey, all... If this is too off-topic, I'd be grateful for directions to a
more appropriate mailing list.
I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000
which is registering twice with Asterisk - once for its FXS/Line1/VoIP1
and once for its FXO/PSTN/VoIP2.
My eventual
I haven't been able
to find an answerand got no response whatsoever to my previous questions
concerning it.
Has anyone found a
fix for the remote connections to the CLI causing crashes? Also, is there a
known limit?
I have a huge need
for using asterisk -rx in scripts, which seems is
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes a pause. I haven't tested it just referenced
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and
Anybody here using iax2 for one call leg and other call leg for oh323? I am
getting broken sounds from Iax2 call get.
Can somebody here help?
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Anthony:
I am trying to get either of the above features to work with *, but
can't seem to get it quite right. If anyone has them working, I'd
sure appreciate an extract from the relevant config files.
Or, maybe I'm tilting at windmills, and * doesn't support them - in
which case, the
Paul Crick wrote:
Has anyone else noticed this or had experience with it? We like the
features of the new firmware release (like being able to split a
conference
call then drop one party) but the change in behaviour of call waiting is
giving a few users grief.
Kevin P. Fleming wrote:
Look at
In article [EMAIL PROTECTED],
Michael Stahl [EMAIL PROTECTED] wrote:
After more investigation, I decided to just recompile asterisk (on my
newly upgraded Fedora core 4 system). Make dies with this error:
No rule to make target
'usr/lib/gcc/i386-redhat-linux/3.4.3/include/stddef.h
It
Hi Anthony
I'm pretty sure Asterisk doesn't support the shared line appearances the
way Polycom does, but you CAN get calls to simultaneously ring multiple
destinations. The wiki has documentation for the Dial command - you're
going to specify multiple targets.
Example: 123 is the boss's phone
Ooh. Jenny, your problem might be the motherboard. (I should know, I run
that same motherboard at home as my gaming machine.) It's a piece of
crud. I actually blew through four of them before I found one that
worked properly. I wouldn't worry about the AGP running on there, but do
make sure
You will need asterik head or to patch ztmonitor.
I use putty(a windows ssh client) to connect to my asterisk server and
to get a larger sreen definition (putty in full screen)
Cd /usr/src/zaptel
./ztmonitor 1 -vv
Where 1 is the channel
Hope that can help
Ken
-Original Message-
Sherwood McGowan wrote:
I haven't been able to find an answerand got no response whatsoever to
my previous questions concerning it.
Has anyone found a fix for the remote connections to the CLI causing
crashes? Also, is there a known limit?
I have a huge need for using asterisk -rx in
Hello,
Did Doug ever buy your card from you? If it works and it's known to
work with asterisk, I'd like it. PayPal?
-Trent Tuggle
On Aug 9, 2005, at 2:24 PM, Dan Littlejohn wrote:
Doug:
If you find that you like intel chipsets, I bought a compatable card
and did not need it.
Andrew Kohlsmith wrote:
Actually that is untrue. There are four INT# lines on a PCI bus.
*everything* uses INTA# and relies on the chipset and motherboard to
correctly wire it up so they don't share... I was going to ask Digium if
they could bring all four INT# lines into the FPGA and
Paul Crick wrote:
That's the setting I've been changing - it seems that whether set to 0 or
set to 1 the behaviour on the phone is unchanged. A new call being
presented shows up on the screen, I get a beep in my ear, but the focus on
the phone stays on the original call. I have to use the arrow
Try setting your logger.conf to allow full output (uncomment the full
section) and see if there is something specific to the CLI crash.
Be careful though and do not let the logging get out of control, especially
on a big system. The file can get huge.
-Original Message-
From: [EMAIL
William Suffill wrote:
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes a
pause. I haven't tested it just referenced
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and
AAH 1.3 with Cisco 7960 phones/ SIP 7.5 software mostly works great,
but there is a problem with one of the phones I use most: It disconnects
calls if I dial on speakerphone and then pick up the handset after the
other side answers.
Thanks in advance for any clues on this. And apologies if
Hello
My single line extension users (connected via channel banks)
need to be able to hang up faster. If they just flash the hook it doesnt
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
___
Asterisk-Users
John Novack wrote:
William Suffill wrote:
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes a
pause. I haven't tested it just referenced
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and
Eric Wieling aka ManxPower wrote:
John Novack wrote:
William Suffill wrote:
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes
a pause. I haven't tested it just referenced
w and W allow call recording when passes as options to DIAL, in this
case they are being passed as options to D().
-jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Monday, August 22, 2005 3:51 PM
To: Asterisk Users Mailing List
John Novack wrote:
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be dialed
If you are using analog ports, yes. Dial(Zap/g1/ww15551212). If using
digital ports then no, you can't have a delay before dialing the number,
but
Now, there actually is actually documented problems with too many remote
connections to the manager (CLI). . . I'm asking if someone's figured out
how to fix that.
http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+experience
Since my scripts apparently get some funky data from the
Karl S. Katzke wrote:
Ooh. Jenny, your problem might be the motherboard. (I should know, I
run that same motherboard at home as my gaming machine.) It's a piece
of crud.
Oh, great. ;-) For next time, does anyone have recommendations for a
particular motherboard or a particular type of
Sherwood McGowan wrote:
Now, there actually is actually documented problems with too many remote
connections to the manager (CLI). . . I'm asking if someone's figured out
how to fix that.
http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+experience
Since my scripts apparently get
Hi Guys,
Sorry about writing to that list but could not find
better place.
I have Grandstream BT-100 phone, btw, was working
great with Asterisk.
I have upgraded the phone, and during upgrade
something went wrong.
Right now when I power the phone I can only see
some garbage on the LCD
I recorded a greeting from one of my extensions by creating an extension
(6001) and then using record(filename:gsm) to record the greeting.
Callers have reported that when they call in, the greeting is VERY murky
and quiet and fades in and out. Is there a better way to create record
Maybe I am, I don't doubt it.
But why does asterisk deadlock then when about 5 or 6 scripts hang while
getting output from *?
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Eric Wieling aka ManxPower
-Sent: Monday, August 22, 2005 5:08 PM
-To:
Ive had similar problems, once I
was able to fix, once I had to return to factory.
Are you able to log in via web server at
all? Can you reset the phone to factory settings from there?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
On Mon, August 22, 2005 20:21, CM Rahman Jr. said:
Anybody here using iax2 for one call leg and other call leg for oh323? I
am
getting broken sounds from Iax2 call get.
Can somebody here help?
Thanks
OT: What does this have to do with the small office / analog thread?
Anyway:
H.323 isn't
Oh, great. ;-) For next time, does anyone have recommendations for a
particular motherboard or a particular type of motherboard?
Next time, I'd actually buy a Dell SC420 or something similar. :-P
X isn't installed, so you can spend more time w/ your sister. ;-)
Aww, drat ... I mean,
Do you have 5 or 6 scripts running against the interface for one instance of
an outside script? Or, do you have multiple connections (outside users)
attempting to run multiple instances of a script that are pulling 5-6 CLI
scripts?
This would exponentially increase the real number of scripts
I cannot even ping the phone, web-config is not working.
The phone also does not react to any buttons.
I've had similar problems, once I was able to fix, once I had to return to
factory.
Are you able to log in via web server at all? Can you reset the phone to
factory settings from there?
Sherwood McGowan wrote:
Maybe I am, I don't doubt it.
But why does asterisk deadlock then when about 5 or 6 scripts hang while
getting output from *?
I don't know. The AMI (Asterisk Manager Interface), accessed via a TCP
connection to port 5038, has some known problems with many
John Novack wrote:
William Suffill wrote:
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes a
pause. I haven't tested it just referenced
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and
Derek Whitten wrote:
try something like this.. the EXTEN:1 strips the leading 8 off
exten = _8.,s,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},60,Ttr)
Not only is the above not a valid extensions.conf line, the poster even
put Ttr in the sample. Horrible.
Assuming NANPA dialing, try
exten =
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