I am trying to create an extension that will ring 2
ip phones if the receptionist doesnt answer. I have used the following
extension:
exten = 1401,1,Dial(H323/10.0.3.14,10,r)
exten = 1401,2,Dial(H323/10.0.3.13H323/10.0.3.14,10,r) exten
= 1401,3,Hangup
It appears to work well, i.e. it
Steve,
For analog, isn't it just a case of getting asterisk to listen out for
specific tones such as busy, or ringing. Isn't this what the
callprogress option is for in zapata.conf? I thought that it works for the
US at the moment but no where else.
Tan
- Original Message -
From:
hi,
r u saying that, if i use E100P or E400P interface, and if i make a call, can i differenciate between a answered and non answered call, and can i bill only to the answered call?
Surajee
On Sun, 2003-07-06 at 23:17, Kim C. Callis wrote: Steve, What exactly would be classified as a digital ZAP
I have a X100P and am calling out from a desktop
within the same network. I connect to * then dialout a local phone number
to my cell phone. It rings 2 times then hangs up.
I'mtesting Sipps as the
softphone.
* is saying "retries exceeded".
Has anyone had this problem? It's probably
with
Back to the discussion at early April 2003,
I have been interested with Rhino Channel Bank, because the price offering
at USD1,295 this time. I was wondering if anyone have any experience with
this.
Also do anyone find the cheaper price, as for your information because of
my region in Indonesia
I'll answer Kim's point more closely, since I know his configuration:
you would need to have accurate call supervision (connect
supervision/disconnect supervision) with messaging in order to have
completely accurate CDRs. If your T100P hooks into a channel bank
which then goes to a bunch of
Hylafax Dan. It isn't that elegant though as you'll need to wire an analogue
port to each fax/modem. AFAIK there isn't a virtual fax/modem provided by *
that another programme can use.
W
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 05, 2003
Hi,
What I need is a pure software solution, to avoid any other hardware to get
that functionality.
Thanks,
Dan
- Original Message -
From: Simon Woodhead [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 11:50 AM
Subject: Re: [Asterisk-Users] Virtual fax on the
Hi Paul,
thanks for your insights
On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote:
To dial a PSTN number through Nikotel used to work from Asterisk, but
they had a very serious security issue (you could make calls anytime
anywhere and their billing wouldn't charge it) and after I
I don't think such a thing exists Dan but please let me know if you find one
as we need the same.
The only additional hardware you'd need is a fax/modem or two for Hylafax
(assuming your * already has analogue extensions set up). One appealing
thing about this solution is that the fax can be on a
Hi,
Can someone explain how or if voicemail2 contexts relate to the dialplan??
If I had the following in voicemail.conf..
[default]
1234 = 4567,Company1..
[company2]
1234 = ,Company2..
[company3]
1234 = 9876,Company3..
How would I make sure that a user from Company2 did not get prompted
Hi,
I have seen all of this on a Cisco 17xx router, including IVR and sending
faxes through e-mail, but it is far too expensive for me...
Theoretically is possible to have let's say an IAX or SIP software phone,
even on a separate computer as Asterisk who can handle faxes.
The only additional
Your uplink is pretty limited at both ends. I'd be
using g.729 over IAX in your situation giving enough uplink for several calls,
or a call and normal use at least. GSM is a bit too hungry for that kind of
connection.
- Original Message -
From:
Jay
Tyndall
To: [EMAIL
Hi All,
This is the third time I'm sending this to the list, seems that all my
messages keep being rejected or something like that.
Is there anybody out there that has an idea how to make Asterisk
work with ZapRAS with an E1 interface? In addition, I would like to know
if anybody ever
I need to configure an * box to connect to a primary * box which is attached
to a PRI in order to make calls using both the same E1 connection.
I know the best solution is to use IAX (altough i could connect the IP
phones on the remote site directly to the main * box that is VPNed with the
On Sunday 06 July 2003 10:19 pm, Anthony Wood wrote:
So does that just leave regular single line phones ?
Besides IP phones.
What else can be used ?
You can plug your old PABX into asterisk and use its phones through that.
Softphones (software + computer + soundcard + microphone
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 10:13 AM
Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box
Hi,
What I need is a pure software solution, to avoid any other hardware to get
that functionality.
i recall t38modem
On Sunday 06 July 2003 11:07 pm, [EMAIL PROTECTED] wrote:
Phi everyone,/P
PI know this issue has been raised many times before, i think still the
problem remains. When a call is made through a Zap channel, whether it is
actually made or not (irrespective of whether, engaged, busy, or actually
I have 2 asterisk systems connected by a 56kbps internet dialup (so at best 33.6k in
both directions) using IAX and GSM...
The one * box is at my home and the other is in the office (before anyone freaks this
is a test environment)... Provided nothing is using the line at the same time I am
Hi everybody,
I have the same problem severals month ago, and the only solution that I
found was modify code on app_dial.c in order to return severals values:
-Return the time when the call have been established (time() unix time),
so you have to see this value and calculate time just before
Setup all your phones to connect to the closest * server..
Setup IAX between the two servers..
Use the switch command on each server and point it to the other server so that
extentions will be available to both..
You will only need to register one server with the other if it have a dynamic IP
On Mon, 7 Jul 2003, Andrea Venturi wrote:
i recall t38modem (a soft modem working as an h323 endpoint)
http://www.openh323.org/t38.html
anyone with some experience to share?
I tried it a few months ago with a maxtnt and a couple t1's. It would
usually receive one page but die on the
On Monday 07 July 2003 08:24 am, Dan wrote:
Hi Andrea,
This is very interesting starting point.
Thanks,
Dan
Great going Dan,
You posted 6 new lines, left an additional 134 lines of old junk, 4 original
headers and 5 old footers.
Can people trim their posts when they reply.
Hi.
First of all : please disable html. Not everyone is able
to read html email. Also mailing lists netiquette expects
that a members writes only in plain text.
Second one:
cdr -provides- accurate info on a call. every call, estabilished
or not, is logged, the is up to you to get what u need.
VoicemailMain2(@context)
Mark
On Mon, 7 Jul 2003, WipeOut . wrote:
Hi,
Can someone explain how or if voicemail2 contexts relate to the dialplan??
If I had the following in voicemail.conf..
[default]
1234 = 4567,Company1..
[company2]
1234 = ,Company2..
[company3]
1234 =
Without quite just saying me too, see below...
I'm experiencing some problems with a TDM40P and was wondering if
anyone else on this list has similar experiences, or maybe even
a possible solution.
Similar experiences...
My setup is a dual PIII-750 with 1 gig of RAM, with an X100P,
You plug a channel bank to a T1 in your PC connected either over T100P or
T400P.
regards
Martin
On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:
Hello everybody
My doubt is about configuration. Can I use a channel bank like zplex-10 or
adtran, plug on it an T1, 24 POTS, an ethernet
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk as pbx. I need feature called as 'three way calling' or
'transfer with consultation'. Registering,calling and 'blind transfer'
work fine.
Is this feature provided by sip clients or by asterisk itself ?
What I have to
Thank you for the answer.
Anyone working on that?
I am trying in the meantime to disable the RTCP reports on the gateways,
hoping that it will work like that.
hristo
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: Saturday, July 05,
Hi,
There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?
I want to be the same extension for all phones, not a specific one for each
of them.
It
Hey Adam,
On Tue, Jul 08, 2003 at 00:58:08 +1000, Adam Goryachev wrote:
Without quite just saying me too, see below...
[...]
My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected
to an analog line to my telco, and a TDM40P with analog phones
I have an AMD XP 1800 with
Hi everybody,
From now I have been working with IVR feature of *.
But now We want to began to use VoIP, SIP, IAX,
etc. Now we have severals initials scenarios where we need to know which
tecnology must apply in order to have a good application. So If somebody can
give us the tecnology
Hi,
There is any experience using Asterisk with VMWare?
I think about installing a virtual linux box over VMWare and then Asterisk
over it.
Thanks,
Dan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi everybody,
I have two little question about modules.conf.
1)As I have seen, to make Asterisk load chan_capi.so
and chan_modem.so you must have: load=chan_capi.so
and load = chan_modem.so in your modules.conf. But I
had understood some time ago that setting autoload =
yes made Asterisk load
e.g.
exten = 8501,1,VoiceMailMain2(${CALLERIDNUM})
Tan
telappliant.com
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 4:47 PM
Subject: [Asterisk-Users] Direct entry to your own voice mailbox
Hi,
There is any possibility to dial a
Hi,
I asked a time ago what were callgroup and pickup
group used for. I have done some proofs and all, and
I'm not sure if I have pick the idea up well!!
That's what I understand:
For example: group=1 callgroup =2 and pickupgroup=2
and my phone is a membership of the group 1.
that's mean that
Thanks for that. It seems one now needs something like this in sip.conf:
[fwd.pulver.com]
type=peer
host=fwd.pulver.com
username=12345
secret=mysecret
fromdomain=fwd.pulver.com
callerid=Free World Dialup
All is well again ...
Iain
--On Saturday, July 5, 2003 9:31 pm -0400 James H. Cloos Jr.
Hey Dan,
On Mon, Jul 07, 2003 at 18:47:07 +0300, Dan wrote:
Hi,
There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?
Sure. Pass the
Hi,
Only one question: Is this the generic format for a
register line on sip.conf?
[EMAIL PROTECTED]:[EMAIL PROTECTED]:port/exten
Or better this one?
user:[EMAIL PROTECTED]
Or if there is another one, please tell me!
Thans a lot in advance!
cmayor
Can´t I connect to the channel bank using ethernet? Why?
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 11:52 AM
Subject: Re: [Asterisk-Users] Newbie Doubts
You plug a channel bank to a T1 in your PC connected either over
Hello!
My business is a wireless ISP. I would like to offer voice to several
business customers. I have a * server, but still need some hardware cards
for it.
I would like to provide individualized billing to these customers for their
usage. Most of these customers have Nortel PBX's.
Hi,
Thank you very much for your help.
It is possible to have a time stamp in the recorded message? I want to
know
when the message has been recorded.
I think someone here was working on a patch for that, which was waiting
for prompts to be recorded. Not sure of the current status.
Can the system command be used to retrieve a variable from a mysql database using the
mysql command line client??
or would it be simpler to write some sort of AGI type application??
--
__
http://www.linuxmail.org/
Now with e-mail forwarding for only
Thanks,
It seems to work with VoiceMailMain application too.
BR,
Dan
- Original Message -
From: Tan Aks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 7:25 PM
Subject: Re: [Asterisk-Users] Direct entry to your own voice mailbox
e.g.
exten =
Conceptually, one could purchase a Cisco(or Cisco like device) that has a
DVM module(digital voice module) and route the TDM traffic from the DVM out
the Ethernet port(VoIP). In a previous life some could and did call this
device a channel bank. Although I disagree. Wade is right on the money.
On Mon, 2003-07-07 at 11:48, WipeOut . wrote:
If you want to use it for IP only it __MAY__ work but probably not all
that well unless you have a really strong processor..
VMWare abstracts the hardware layer you will not be able to get it to
communicate fully with any cards in order to
Couple of things that come to mind.
What's the latency or your point to point wireless connections.
What type of voice interface do the PBX's have(analog, digital)?
If you plan on using a ATA like device and the Nortel phonesets are digital
you probably will run into echo problems.
What do you
On Mon, 2003-07-07 at 12:33, Asterisk wrote:
Hello!
My business is a wireless ISP. I would like to offer voice to several
business customers. I have a * server, but still need some hardware cards
for it.
I would like to provide individualized billing to these customers for their
On Mon, 2003-07-07 at 12:43, WipeOut . wrote:
Can the system command be used to retrieve a variable from a mysql database using
the mysql command line client??
or would it be simpler to write some sort of AGI type application??
-= Info about application 'System' =-
[Synopsis]:
Execute a
Citeren Dan [EMAIL PROTECTED]:
There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?
I want to be the same extension for all phones, not a
The system at the moment can run some program/script but there is no way
to retrieve the results. Although you could have tried with
${ENV(VARIABLE)}
Martin
On Mon, 7 Jul 2003, WipeOut . wrote:
Can the system command be used to retrieve a variable from a mysql database using
the mysql
Hello!
My business is a wireless ISP. I would like to offer voice to
several business customers. I have a * server, but still need some
hardware cards for it.
I would like to provide individualized billing to these customers
for their usage. Most of these customers have Nortel PBX's.
I want to be able to check my Bell voicemail
(*98) using a phone attached to my asterisk box.
In extensions.conf I have defined
exten=*98,1,Dial,Zap/g1/BYEXTENSION
However, when I dial *98, I just get a fast busy
signal.
Is the * digit reserved for internal purposes?
Any help is
I need a recommendation on a good motherboard/processor
combination. I would like a
motherboard that has lots of PCI slots and works well with Asterisk without
problems getting drivers working, etc.
Onboard LAN would be nice to keep from using a slot. Plan to use RedHat 8 for the OS.
Just purchased a couple of T100 and E100 cards in order to interface from
our company's proprietary system through a linux gateway. I am new to
Asterisk and Digium.
After installing the T100 card, I went looking for drivers for this card.
Are the drivers built into the Asterisk application? If
On Monday 07 July 2003 12:07, Dan wrote:
Hi,
There is any experience using Asterisk with VMWare?
I think about installing a virtual linux box over VMWare and then
Asterisk over it.
I use it with VMware, but Asterisk runs on the host OS, not in the VM.
You should have no problem running
On Mon, 2003-07-07 at 13:23, Martin Pycko wrote:
The system at the moment can run some program/script but there is no way
to retrieve the results. Although you could have tried with
${ENV(VARIABLE)}
This shouldn't work as the system command will run in it's own
environment and should not
Hi,
I am lost trying to figure out how to enable overlap dialing for calls
coming in and coing out on a pri span. DISA looked promising at first,
but does not seem to support overlap dialing. Just picking up a call by
and trying to dial out does not seem the way to do it either. I tried:
On Mon, 2003-07-07 at 13:31, Derek Beaumont wrote:
I want to be able to check my Bell voicemail
(*98) using a phone attached to my asterisk box.
In extensions.conf I have defined
exten=*98,1,Dial,Zap/g1/BYEXTENSION
However, when I dial *98, I just get a fast busy
signal.
Is the *
On Mon, 2003-07-07 at 13:29, Clay Graner wrote:
I need a recommendation on a good motherboard/processor combination.
I would like a motherboard that has lots of PCI slots and works well
with Asterisk without problems getting drivers working, etc. Onboard
LAN would be nice to keep from using a
Hi,
The reason I ask this is because I have a Win2K PC running 24/7 which has
enough power left, but if I cannot use any of the Digium hardware from
inside VMWare then is useless.
Thank you,
Dan
- Original Message -
From: Gary Gapinski [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
overlapdial=yes in zapata.conf
for those channels that you want the overlapdialing be activated.
By default only incoming overlap dialing is enabled.
regards
Martin
On 7 Jul 2003, Thilo Salmon wrote:
Hi,
I am lost trying to figure out how to enable overlap dialing for calls
coming in and
Come on,
exten = *98,1,Dial,Zap/g1/BYEXTENSION
should work since it's old sytax.
It's more propable that you have that *98 in a diffrent context
than assigned for those channels or you don't have the group 1 defined
properly
Martin
On 7 Jul 2003, Steven Critchfield wrote:
On Mon,
To enable any of TxxxP or ExxxP card you must compile the zaptel package
from cvs.
Then to enable the T100P or E100P you should load the wct1xxp module.
Then you can use the zttool (/sbin/zttool) to check the status of the cards
and of the spans you've configured in /etc/zaptel.conf
--
Stefano
Thanks Florian,
Dan
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 9:15 PM
Subject: Re: [Asterisk-Users] Direct entry to your own voice mailbox
Citeren Dan [EMAIL PROTECTED]:
There is any possibility to dial a
On Monday 07 July 2003 15:26, Dan wrote:
The reason I ask this is because I have a Win2K PC running 24/7 which
has enough power left, but if I cannot use any of the Digium hardware
from inside VMWare then is useless.
If you have not yet purchased the VMware license, run Linux, Asterisk,
I use Windows Messenger ( I duck as to let the hurled
penguins barely miss my head J ) and I am able to
place and receive calls. So what is the problem you ask??? If I specify a
password in the password field of WM I get a Proxy Authentication Error during
SIP debug and I am not able to
Hey Dan,
On Mon, Jul 07, 2003 at 20:42:16 +0300, Dan wrote:
It is possible to have a time stamp in the recorded message? I want to
know
when the message has been recorded.
I think someone here was working on a patch for that, which was waiting
for prompts to be recorded. Not sure
Hi
thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service.
I have drafted a mini-how-to which is available at
http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf
This is a
I am always getting multiple PCI Master Abort messages when I try to
plug in a second TDM400P.
I have asked this question before, but I nothing really solved my
problem and I just put it on the back burner for a while.
I am at a point where this is a crucial issue.
I have read that the Zaptel
Hi
I have just been struggling for four days to get SIP working and now as
I created a voicemail box, Asterisk has become very unstable and it
can't bridge SIP phone to SIP provider calls anymore.
Calling internally from one SIP phone to another works fine.
Calling internally from a SIP phone
Zaptel hardware, rule #1
don't share irq, each card on it's own irq
(disable usb, sound, rtc and other fancy hw you don't need)
Matteo.
Il lun, 2003-07-07 alle 22:15, Derek Beaumont ha scritto:
I am always getting multiple PCI Master Abort messages when I try to
plug in a second TDM400P.
I
You can force IRQs in your BIOS config, I would set each card to its own IRQ
that doesn't get shared with anything else. Disable your serial and
parallel ports if you aren't using them and use 3,4,5,7
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Not 100% sure here but its probably somthing to do with the fact that MS doesn't
support MD5 and I think * makes use of md5 password hashing during authentication..
Maybe you can try adding auth=plaintext to that account in the sip.conf I know this
option works in the iax.conf..
Later..
I
Hi Steven,
The PC is behind a NAT firewall/router, so this is not a real issue for the
moment.
It is online 24/7 for allmost 2 years now without any major problem.
It hosts my Home Automation engine and it must remain powered on full time.
I do not intend to install X just for VMWare.
Thanks,
Hi Dan,
For a totally unrelated reason I did this today. * runs fine here
under VMware, athough I haven't stressed it at all.
Andy
*** REPLY SEPARATOR ***
On 07/07/2003 at 19:07 Dan wrote:
Hi,
There is any experience using Asterisk with VMWare?
I think about installing a
Hi All
I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S
based ISDN card using ISDN4Linux.
I have setup the BudgeTone-100's to use Early Dial which for calling between
the three phones works well, but for the external calls using the following
extension
exten =
Subject: Re: [Asterisk-Users] IAX Bandwidth Question
Hi,
I have changed the codec to lpc10 (I see what they mean by Mr. Roboto!!)
and
the ping times generally dont go over 300ms.
It seems very odd that GSM saturates the link.
I would love to give g.723.1 a try but have no idea where to
Dan == Dan [EMAIL PROTECTED] writes:
Dan I have seen all of this on a Cisco 17xx router, including IVR
Dan and sending faxes through e-mail, but it is far too expensive for
Dan me... Theoretically is possible to have let's say an IAX or SIP
Dan software phone, even on a separate computer as
Paul,
First, make sure that you use inband DTMF. As far as I know, out of band
still does not work. Second, make sure that the firmware is up to date.
The silent DTMF problem was fixed a few releases ago (at rev
xx.xx.xx.60 I believe).
--
Stephen R. Besch, Ph.D.
SachsLab
320 Cary Hall
SUNY
Martin,
I probably should have mentioned that: overlapdial=yes was set in
zapata.conf (I take it this option is inherited through all the
channels I configure in zapata.conf). I also did a fresh checkout today.
My guess is that the main problem for now lies in the fact that asterisk
won't
# make clean ; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
Hi All,
I'm looking at setting up a Asterisk system, and hope to use ATA 186's
with it.
Im in Australia, and am getting mixed answers to if its the I1 or I2 i
need, does anyone have any experience with using ATA 186's in Australia
Also, can anyone recommend a good place to obtain these locally?
When I first to the add extension the show dialplan has the lines that
say SIP/ but after I do a save dialplan and a stop gracfully and
restart the lines with SIP/ are gone.
Show dialplan before:
asterisk01*CLI
[ Context 'default' created by
A general tip:
- use APIC even on a single processor system
APIC (Advanced Programmable Interrupt Controller?) or more specifically
IO-APIC, is a feature on newer motherboards (=2000) that allows the OS
to allocate more than 15 IRQs. Windows XP depends on this feature to
solve the IRQ sharing
Read the original config file, especially near the top. It isn't
implemented yet.
On Mon, 2003-07-07 at 17:23, [EMAIL PROTECTED] wrote:
When I first to the add extension the show dialplan has the lines that
say SIP/ but after I do a save dialplan and a stop gracfully and
restart the lines
Do you have the source that your kernel was compiled from? Do you at
least have the appropriate headers for you kernel and the config file
that was used?
On Mon, 2003-07-07 at 18:28, marrandy wrote:
# make clean ; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg
Hello,every one!
I would like to know if asterisk could run under
uclinux.
Regards.
Hi.
Has anyone experienced hangup detection problems with the VoiceMail2 app?
I have a console phone on the FXS port. When I call a SIP phone, and get
its voicemail greeting, I can enter the VoiceMail2 app, leave a message,
and then hit # to stop message recording.
Recording does stop, but the
I updated to the latest CVS about 4 hours ago. If I let
asterisk run, connect a few times via r after about one hour the
system does not let me in and it starts sucking up resources. One time it
spawned about 100 processes. Had to do a reboot.
This time it just did not let me in and
So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I
cannot get the phones to talk/RTP to each other. jtodd has had this problem in the
past with the 186's. Just wondering if anyone has a reason why Cisco sometimes poop
on reinvite is the Cisco code broke? if so
Hi Steven
Thanx for your reply . U said i can store data in Postgres would u plz me
tell the steps to Configure it ,as i am newbie i didnt understand properly
ur reply if u want i can send u the whole scenario.
Thanx in Advance
Syed Obaid Amin
From: Steven Critchfield [EMAIL PROTECTED]
If you type ls /proc/first pid of asterisk/fds do you see an unusually
high ( 200 or so) file descriptors? if so, what do they say?
Mark
On Mon, 7 Jul 2003, Alex Lopez wrote:
I updated to the latest CVS about 4 hours ago. If I let asterisk run,
connect a few times via -r after about one
I'm experiencing one-way voice paths, followed by a hangup on one
softphoine and not the other. Also, caller ID has odd outputs -- and I
wonder if the problems are related.
My configuration has Asterisk and a Linphone softphone running on the
same box. I have a PC, and on that PC I use X-Lite
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