No it points to Cell phone companies having better hardware echo
cancellation on their lines, also cell phones themselves have a hardware
echo can built in.
- Original Message -
From: Mike Benoit [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 12, 2004 1:52 AM
Subject: Re:
So isn't this the problem * has? The first client registers as the address
of record, then the second client comes in with the same registration and
becomes the address of record?
I think you are making this look more complicated than it actually is.
We do this with our SER Network all the
That doesn't explain why a incoming call from a land line has nearly no
echo, while an outgoing call to the same land line has echo.
Also it has always been near end echo I'm hearing, and prior to
upgrading the mainboard/CPU I heard echo when calling the same cell
phones.
On Mon, 2004-07-12
Well Andres is right but there are numerous problems with quite a few SIP
clients that do NOT follow the the SIP RFC correctly. There is a problem
with dialog creation in a number of SIP products out there. SIP dialog
creation is the critical part of the spec that supports parallel forking -
so be
Brian K. West wrote:
per peer
bkw
Brian,
What will happen to SIP UA call flow and notransfer is left at its
default value?(Presumming SIP UA has canreinvite=yes)
Would SIP UA stay with original server? Or?
Ta
SJ
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Brian K. West wrote:
per peer
bkw
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 9:25 PM
Subject: Re: [Asterisk-Users] Stopping reinvite with IAX2?
Is this set on a per peer basis, or in the general section?
Michael
Dr. Rich Murphey wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 15:39 -0500, Dr. Rich Murphey wrote:
You might check login class in login.conf for the user that invokes
asterisk. Setting cputime=unlimited may
IsChanAvail() application might help
Atif
Sent via the WebMail system at convergence.com.pk
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On Mon, 2004-07-12 at 02:38, Richard Scobie wrote:
A slightly similar observation, which I assume is normal as the boxes
work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load,
about every 30 seconds, with no calls being handled.
You don't mention it, but it sounds like you
Howdy,
I just did an apt-get dist-upgrade on my Debian unstable box,
and noticed that the Asterisk version appears to be 1.0-1 in the
unstable tree. I KNOW that 1.0 hasn't been released yet, so I am
wondering who is responsible for the Debian packages? This will be VERY
VERY confusing
is there any option of inviting some one to conference, I mean, I press * for menu,
then system asks me to invite some one dial 1, and then asks me to dial the extension
of that person, and then call is placed to invite that person to conference.
Thank you
Atif
Steven Critchfield wrote:
On Mon, 2004-07-12 at 02:38, Richard Scobie wrote:
A slightly similar observation, which I assume is normal as the boxes
work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load,
about every 30 seconds, with no calls being handled.
You don't mention it,
Paul Mahler wrote:
Well, this is certainly getting exciting.
Yes, it is. Sorry for coming in late to this debate...
Andy, I took your advice and re-read the RFP.
It's actually RFC, not RFP. (teasing :-)
So, gentlemen, help me out here. The spec says:
The Address of record is the SIP address
Hi
I recently noticed that asterisk passes Caller IDs and SendText messages
containing sepcial characters (such as the german umlaut characters äöü)
with ISO-8859-1 encoding to the SIP phone. Hence user names and text
strings like Müller are not correctly displayed on the receiving phone.
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
-Kannaiyan.
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Hello
I'm toying with adding a feature request to provide some sort of
gain setting for voicemail when accessed from certain interfaces.
Maybe something like voicemail=6.0 (db) within a specific channel
section of zapata.conf corresponding to a pstn line.
That gets my vote. We experience this
Excellent Post! Very Informative. Thanks a lot Sir!
Regards, Girish
From: Olle E. Johansson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Date: Mon, 12 Jul 2004 10:52:33 +0200
Paul Mahler wrote:
Well, this is certainly getting exciting.
Yes, it is. Sorry for
On 11 Jul 2004 at 19:16, Rich Adamson wrote:
QoS is most certainly an issue when making the decision to move off
the PSTN. Is the performance of your VoIP system going to be
comparable to the performance of your PSTN system? Sounds like a
reasonable question to me.
Not trying to get
Kannaiyan Natesan wrote:
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
I apologize for my Swenglish language...
I don't believe there's a quick fix at all.
If you want a quote for a fix, contact me off-list. But remember, that I believe
that fixing this is
That gets my vote. We experience this low-volume voicemail
problem. (and I spent a long time looking for the proposed
setting to tweak!)
Think about a dynamic sound compressor that would possibly auto-adjust.
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[EMAIL
So the two questions remain.
1. Why do incoming calls have nearly no echo (sound great), and outgoing
calls are bad during the first 30 seconds, and okay (but not good) after
that.
2. Why do outgoing calls to cell phone numbers sound great?
Seeing as an outgoing call to a land line
Hello
That gets my vote. We experience this low-volume voicemail
problem. (and I spent a long time looking for the proposed
setting to tweak!)
Think about a dynamic sound compressor that would possibly auto-adjust.
Are you suggesting such a thing exists, or that that would be a
proposed
On Sun, 11 Jul 2004 23:02:56 +0100, Richard Airlie [EMAIL PROTECTED] wrote:
On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote:
Richard Airlie [EMAIL PROTECTED] wrote:
First things first. Scrap the ports and build from the latest
CVS source. 0.9 is far to old and buggy, and
Hi everybody,
Is the only way to use asterisk _not_ as root to change the permission of all
the directories where asterisk need to create a file? (/var/run/,
/var/log/asterisk/messages)
any help will be appreciated,
Cyprien
___
Asterisk-Users
Hi,
I'm testind Diax. I have flashing note about 1 new voice message. Can I hear
it somehow from Diax gui, or must I call pbx to get message ?
Thanks,
Robert.
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On 11/07/2004 at 18:11 Paul Mahler wrote:
Well, this is certainly getting exciting.
Andy, I took your advice and re-read the RFP. Andy--I don't think you are a
Sorry, I was sleeping when these new emails came in
I've read the other responses which seem to make it pretty clear.. and
Hi Robert,
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
I'm testind Diax. I have flashing note about 1 new voice message. Can I
hear
it somehow from Diax gui, or must I call pbx to get message ?
You need to call Asterisk to get the message.
Diax just gives you the
I modified the permissions of /var/spool/asterisk and /var/log/asterisk
and it seems that asterisk is launching now. But I still have messages at the
beginning telling me that:
Unable to open pid file '/var/run/asterisk.pid': Permission denied
Unable to bind socket to /var/run/asterisk.ctl:
Are you suggesting such a thing exists, or that that would be a
proposed future application?
I propose to think if an AGC / dynamic compressor could be used instead of
a config variable.
Most sound editors have modules for this.
___
Asterisk-Users
Could you kind Asterians (should we pick Asteroids then?) confirm if I
can use an E100P card with a T1 channel bank via * please? I live in the
UK hence the question.
Luan
One UK Asteroid (...this sounds better I think)
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Asterisk-Users mailing list
I don't think we should let these misunderstandings judge the quality of
Paul's Asterisk book. Even authors need to learn now and then :-)
Can I just point out that the reason I said what I said (see, I can't write)
was because Paul steadfastly refused to believe what we were saying, rather
Are you suggesting such a thing exists, or that that would be a
proposed future application?
I propose to think if an AGC / dynamic compressor could be used instead of
a config variable.
Most sound editors have modules for this.
So how would you detect the remote caller is 14.7 db
how do you ping a TDM connection ?
On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote:
On 11 Jul 2004 at 19:16, Rich Adamson wrote:
QoS is most certainly an issue when making the decision to move off
the PSTN. Is the performance of your VoIP system going to be
comparable to the
On Monday 12 July 2004 07:36, luan au wrote:
Could you kind Asterians (should we pick Asteroids then?) confirm if I
can use an E100P card with a T1 channel bank via * please? I live in the
UK hence the question.
Yes. You''l only get 24 channels but it shoudl work fine.
And I prefer the term
On Monday 12 July 2004 05:43, [EMAIL PROTECTED] wrote:
Not trying to get in the middle of whatever argument you're trying to
make, the poster's original question (although probably not worded all
that clear) can be answered by... no, asterisk cannot make a decision
to route calls via a
Dear All,
I just do cvsup for asterisk (7/12/2004),and yesterday cvs with the same result.
I'm trying to make gnophone work with asterisk.
Following the wiki pages, here's my iax.conf
[general]
port=5036
;bindaddr=192.168.1.145
iaxcompat=yes
delayreject=yes
bandwidth=low
;
;allow=all
On 12 Jul 2004 at 14:06, Michael Bielicki wrote:
how do you ping a TDM connection ?
Sorry, where does it say this is regarding a TDM connection?
I use IAX trunking and a ping script to check times and fluctuations
to my remote offices.
Matt Riddell
On Mon, 2004-07-12 at 11:43, [EMAIL
Hello!
I have an E100P connected to our partner's PBX. They want the
following:
Called number must have numbering plan/type set as: unknown/unknown
and calling number in: ISDN/national.
I searched for the config file, but I found only pridialplan option on
zaptel.conf. When I set it to unknown,
On 12 Jul 2004 at 8:22, Andrew Kohlsmith wrote:
On Monday 12 July 2004 05:43, [EMAIL PROTECTED] wrote:
Not trying to get in the middle of whatever argument you're trying
to make, the poster's original question (although probably not
worded all that clear) can be answered by... no,
Doesn't make any difference 'how' one might ping a remote site,
ping will never qualify the Quality of the channel between two points.
It will only suggest its up/down and possibly the delay at that
specific point in time. Has nothing to do with whether packets were
dropped or delayed some
Andrew Kohlsmith wrote:
On Monday 12 July 2004 07:36, luan au wrote:
Could you kind Asterians (should we pick Asteroids then?) confirm if I
can use an E100P card with a T1 channel bank via * please? I live in the
UK hence the question.
Yes. You''l only get 24 channels but it shoudl work fine.
Hi folks!
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
german numbering, is not. So if I get a call from a mobile phone
0177-1234567
pridialplan=unknown
prilocaldialplan=national
Thomas wrote:
Hello!
I have an E100P connected to our partner's PBX. They want the
following:
Called number must have numbering plan/type set as: unknown/unknown
and calling number in: ISDN/national.
I searched for the config file, but I found only
Would you consider posting this this to the wiki? :)
I think that would be great.
On Mon, 2004-07-12 at 08:35, [EMAIL PROTECTED] wrote:
On 12 Jul 2004 at 14:06, Michael Bielicki wrote:
how do you ping a TDM connection ?
Sorry, where does it say this is regarding a TDM connection?
I
On Mon, Jul 12, 2004 at 03:30:24PM +0700, Isianto Istiadi wrote:
and then I do nmap -sU ip (I don't see port 4569 or 5036 available).
I can't register gnophone with *, when I do ethereal, I can see that
gnophone tried to connect to port 5036, but the * replied destination unreachable.
Is there
On Mon, Jul 12, 2004 at 01:32:39PM +0200, Cyprien Simons wrote:
I modified the permissions of /var/spool/asterisk and /var/log/asterisk
and it seems that asterisk is launching now. But I still have messages at the
beginning telling me that:
Unable to open pid file
Differences in how poll() works is probably responsible.
Try this and see if it helps.
Cheers,
Rich
-Original Message-
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 16:03 -0500, Dr. Rich Murphey wrote:
That sounds like a bug. One should be able to
Is it possible to tell asterisk not to strip the leading 0
of *incoming* MSNs? I use asterisk with i4l and whenever
I get a call from an long-distance party, the leading 0, which
should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the
Kai Militzer schrieb:
...
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
german numbering, is not. So if I get a call from a mobile
Interestingly you do not get the same problem of FreeBSD 5.2.1.
Chris
On Sun, 2004-07-11 at 23:55, Jean-Yves Avenard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
On 12/07/2004, at 4:24 AM, Arjan wrote:
43676 root63 0 10244K 7628K RUN 2:44 99.05% 99.02%
Kai Militzer wrote:
Hi folks!
Is it possible to tell asterisk not to strip the leading 0 of
*incoming* MSNs? I use asterisk with i4l and whenever I get a call
from an long-distance party, the leading 0, which should be there
according the german numbering, is not. So if I get a call from a
Darren,
Many thanks for your help - I've got further, but am still stumped. Have a
look at the following table:
LED | ISDN| Asterisk
--+---+-
OOS | Out | Red
ACT | Green | Green
RED | Out | Red
YEL | Out | Out
LBK | Out | Out
CC
Hi, which IP Centrex setup are you using?
Gary
I am using asterisk as a voicemail server for our IP Centrex SoftPBX.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten
Sent: 09 July 2004 22:46
To: [EMAIL PROTECTED]
Subject: Re:
TrTryingo get * to register to a service that uses account and pin but
the PIN must be encrypted using MD5. The service does not require the
phone number to register to the SIP Proxy.
I can get the REGISTER message to send the account by using the below
register line in the [general] section of
Cyprien Simons wrote:
Is the only way to use asterisk _not_ as root to change the permission of all
the directories where asterisk need to create a file? (/var/run/,
/var/log/asterisk/messages)
http://voip-info.org/wiki-Asterisk+non-root
F
___
[EMAIL PROTECTED] wrote:
I use IAX trunking and a ping script to check times and fluctuations
to my remote offices.
Could you share this AGI?
- seems like a useful example :)
Thanks a lot,
F
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Roger Schreiter schrieb:
I have currently the same problem with my E1 card and I wonder,
...
SetCallerID(0${CALLERIDNUM})
O.k. this works fine for me too.
I hope, I won't have to take special care, when
calls came from local or from international.
Roger.
Roger Schreiter [EMAIL PROTECTED] wrote:
[...]
I have currently the same problem with my E1 card and I wonder, how
I can get asterisk to append a leading 0 before forwarding the call,
for my IP phones show the correct callee number with leading 0.
I ended up just writing a Perl AGI script to
On Tue, 13 Jul 2004 [EMAIL PROTECTED] wrote:
Qualify will only stop the call going through if for example the ping
is above 200ms. I find most of my problems come from fluctuating
ping times (~100ms) than from a stable high ping.
I agree that the overall delay isn't really the problem
Hello everybody,
Is there any alternative to Asterisk ZapBarge
command for SIP and IAX channels?
Thanks
Lamine
Thanks for this. I think I have it working as desired.
What are the implications of allowing the transfer to occur? I'm not
confidetn about allowing my server to lose control of the call. I would
be in effect allowing my cell phone to communicate directly with VPC.
Can I be certain about call
What's your relevant dial peer sip.conf config?
-g
On Fri, 2004-07-09 at 03:49, Mikael Andersson wrote:
Glen Hinkle wrote:
I assume the pstn is your * system.
Can you get audio both ways if you send the traffic back to *?
pstn - as5350 - pstn ?
-g
Iuse the as5350
On Mon, 12 Jul 2004 14:57:42 +0200, Kai Militzer [EMAIL PROTECTED] wrote:
Hi folks!
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
Hello Guys,
after an update to cvs head (thanks oej!) my CiscoGW can now flag unkown
caller's
to Number AND Name Unkown.
Before i again open a new bug (which isn't a bug :-)), can someone confirm
this:
- PrivacyManager does not recognize this as an unknown number
- it's not possible to set ANY
On 12/07/04 11:11, Michael Sandee wrote:
pridialplan=unknown
prilocaldialplan=national
Not only is this that undocumented, but the string prilocaldialplan
doesn't even show up in the latest CVS HEAD source code, so that's not
going to work...
On 12/07/04 13:36, Thomas wrote:
I have an E100P
What about a post processor that performs Compression/Normalization on
the recorded voice mail file?
On the down side I can see this being a big CPU hog if you are handling
a huge amount of calls and trying to normalize a 5 minute long voicemail
at the same time.
On the upside you don't have to
I am using the Pocket PC 2003 version of SJPhone and it seems to be
working OK.
I however do notice hudreds of the following warning message in my
asterisk log whenever I use the sjphone:
Jul 12 10:37:11 WARNING[-1426744400]: dsp.c:1467 ast_dsp_process: Unable
to process inband DTMF on 2 frames
Using an example provided by The Hitchhiker's Guide to Asterisk, I
made the following addition to my extensions.conf file:
[inbound-analog]
exten = s,1,Wait(1)
exten = s,2,SetVar(counter=0)
exten = s,3,Answer()
exten = s,4,Wait(1)
exten =
The 0 never is there.
Check for my post here:
http://lists.digium.com/pipermail/asterisk-users/2004-July/053985.html
And the solution here:
http://lists.digium.com/pipermail/asterisk-users/2004-July/053989.html
Kind regards,
Martin List-Petersen
On Mon, 2004-07-12 at 14:28, Roger Schreiter
Thanks for your post, that solved it.
It was just not documented anywhere.
/Martin
On Fri, 2004-07-09 at 15:41, Michael Sandee wrote:
Hi MLP
nationalprefix=0
internationalprefix=00
Regards,
Martin List-Petersen wrote:
I've got a bit trouble with callerid and zaphfc cards.
Hi guys, I create a topology like fellow:
/** / /***
* GK *---* Asterisk *-- Sip Prov *
**/ /
***/
||
|
Hi List!
Thanks for the numerous replys. The SetCallerID workaround did it so far
for me. Thank you very much!
Regards
Kai
Am Mo, den 12.07.2004 schrieb Manuel Wenger um 15:24:
Is it possible to tell asterisk not to strip the leading 0
of *incoming* MSNs? I use asterisk with i4l and
Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via *
(either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58)
I didn't hear any ringing sound get the following on the console:
-- Called 5503
-- SIP/5503-f6b5 is ringing
WARNING[-1323201616]: channel.c:1375 ast_indicate:
On Mon, 2004-07-12 at 15:11, Peter Corlett wrote:
Roger Schreiter [EMAIL PROTECTED] wrote:
[...]
I have currently the same problem with my E1 card and I wonder, how
I can get asterisk to append a leading 0 before forwarding the call,
for my IP phones show the correct callee number with
iH
went to the link to take a look but admin/admin doesn't work
- hcir
On Jul 9, 2004, at 10:56 AM, San Singhania wrote:
Hello everyone,
I am developing an online SMDR / call log system for asterisk. This is
going to take the form of an executable with embedded sql and
webserver,
pdf
Hi folks,
I found that I can config my IAXy to connect to a * server that is has a
fixed IP.
I'm using dynamic dns solusion, and I want the IAXy to be able to connect to
domain.name.server instead of IP.
Do you know how to do that?
if it is not possible, do you know when will it be?
thanks
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions the same caller id?
I do something with my zaptel and x lite phones... I assign them both the
same number
Hi all,
I have installed a test machine with asterisk in order to try it. I have a
problem with capi channel (chan_capi 0.3.4a). When an external call directed
to an internal Ip phone is not answered I obtain this warning repeated many
times:
Jul 12 16:13:43 WARNING[1209214400]:
hi,
we use ser for signalling and asterisk as gateway.
is there a possibility to configure the pri-causes
for SIP Responses.
SER = 404 NOT FOUND = PSTN ..
At this moment the Caller gets
no connection under this number
It would be nice to signalling something like:
participant not available
See inline comments...
asterisk wrote:
Darren,
Many thanks for your help - I've got further, but am still stumped. Have a
look at the following table:
LED | ISDN| Asterisk
--+---+-
OOS | Out | Red
ACT | Green | Green
RED | Out | Red
YEL | Out
I built a system and then changed the IP and
subnet. Now the phones will not register, getting a 403.
Any ideas?
Hmmm... I don't know if playing with the * code would really be the best
here... Although if it was a plug-in app like app_volume or something I
guess it couldn't hurt... It really sounds like you have a line issue here.
You said that adjusting the gain on your card introduced echo issues. It
Hi can anyone help me on this error msg??
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
thnx
St
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At 5:00 PM -0600 on 7/11/04, Rich Adamson wrote:
I'm toying with adding a feature request to provide some sort of
gain setting for voicemail when accessed from certain interfaces.
Maybe something like voicemail=6.0 (db) within a specific channel
section of zapata.conf corresponding to a pstn
On Mon, 2004-07-12 at 09:31, Seth Remington wrote:
What about a post processor that performs Compression/Normalization on
the recorded voice mail file?
On the down side I can see this being a big CPU hog if you are handling
a huge amount of calls and trying to normalize a 5 minute long
Are there any debugging tools for the digium zaptel cards that would
report the activity on the line, such as DTMF and/or connection
protocol?
I'm looking to debug the connection with a T100P, I don't have $2000
for a T1 test set.
Thanks,
Glen
[EMAIL PROTECTED] (Cyprien Simons) writes:
Is the only way to use asterisk _not_ as root to change the
permission of all the directories where asterisk need to create a
file? (/var/run/, /var/log/asterisk/messages)
any help will be appreciated,
Grab my patches below. It does both chroot
in response to Olle's excellent post, ...
(B
(Bin particular ...
(B
(BAsterisk is *not* a SIP proxy. It's a SIP registrar and
(Blocation server.
(BIt's a very clever SIP UA. It wants to be in the middle
(Bof the call
(Band wants to be in control of each device. This
(Bdevice-slave view
snip
-- Executing SetVar(Zap/99-1, counter=[0+1])
in new stack
-- Executing GotoIf(Zap/99-1,
[[0+1]3]?s|7:h|1) in new stack
-- Goto (inbound-analog,h,1)
snip
It looks to me as if the Gotoif thinks that [0+1] is
greater than or
equal to 3 and therefore jumps to hangup.
On Mon, 2004-07-12 at 15:30, Alastair Maw wrote:
On 12/07/04 11:11, Michael Sandee wrote:
pridialplan=unknown
prilocaldialplan=national
Not only is this that undocumented, but the string prilocaldialplan
doesn't even show up in the latest CVS HEAD source code, so that's not
going to
Richard,
1. don't run 0.5 zaptel driver with asterisk-head it will panic the kernel.
2. I am pretty sure that the current BSD zaptel driver only supports the fxs
modules and the x100p card.
Chris
- Original Message -
From: Richard Airlie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
One of my coworkers needs to get a softphone set up to my Asterisk system;
he's a Linux user, so it looks like about the only IAX2 option is iaxComm.
For ease of use (he'll be using this a fair bit), I'm recommending that he
get a USB handset; I'm just having trouble finding any US retailers
Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via *
(either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58)
I didn't hear any ringing sound get the following on the console:
-- Called 5503
-- SIP/5503-f6b5 is ringing
WARNING[-1323201616]: channel.c:1375
Oh I'm sorry... this setting was probably bri-stuff specific. I didn't
know... I've been using it for a while now and got used to it.
On most ISDN2/BRI lines you need the setting below to actually have a
correctly functioning line (with proper outgoing callerid). It is
probably why it was
On Mon, 2004-07-12 at 16:09, Martin List-Petersen wrote:
On Mon, 2004-07-12 at 15:11, Peter Corlett wrote:
Roger Schreiter [EMAIL PROTECTED] wrote:
[...]
I have currently the same problem with my E1 card and I wonder, how
I can get asterisk to append a leading 0 before forwarding the
On Mon, Jul 12, 2004 at 10:51:24AM -0400, Steve Woolley wrote:
exten = t,1,SetVar(counter=[${counter}+1])
exten = t,2,Gotoif([${counter}3]?s,7:h,1)
You need $2
Example:
SetVar(lala=$[1 + 2]);
GotoIf($[${CALLERIDNUM} = 303]?3:2)
http://www.voip-info.org/wiki-Asterisk+Expressions
Hi All
I've been away from Asterisk for some time. I was wdonering what the
development status is on this?
We've already got a couple of Siemens ISDN phones on an ISDN line, and
I was wondering what the development status was for using them with
Asterisk?
My hope is that it is possible to attach
On Mon, Jul 12, 2004 at 08:47:12AM +0700, Isianto Istiadi wrote:
On Fri, 09 Jul 2004 13:58:30 +1000
Dear Gonzalo Servat,
I'm successfully using your wake-up script, but found 1 problem. Other than that it
works perfectly good. Thanks man. ^_^
anyway, my problem seems to be the timezone or
I currently have a Fractional T1 coming into my site which runs into an adtran
device which splits out 10 channels for data in the form of an ethernet
interface and 14 analog lines for voice.
The ethernet goes directly to a pix firewall.
How would I split out the T1 so that it sends a T1 to
Are you using the lastest cvs? If not you have a broken gotoif...
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Woolley
Sent: Monday, July 12, 2004 9:51 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Gogoif with
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