hi daniel,
I'd guess because there are several enqueued with the same
name and when they aren't send they'll be overwritten in queue?!
no they won't be overwritten, a queue can hold many sms to send.
How can I fix this?
no need to fix
regards,
stefan
In article [EMAIL PROTECTED],
John Morris [EMAIL PROTECTED] wrote:
Minor fix. I'm using this in my RPM specfile.
For this to get acted upon, you need to post it to bugs.digium.com
Cheers
Tony
John
--- ./zaptel-1.0-RC1/Makefile.bigu 2004-07-16 17:09:07.0 -0500
+++
This was recently added as README.mp3:
Yes, and there was recently added a
$ make mpg123
target to Asterisk's Makefile. It will use wget to download mpg123
0.59r, extract it and compile it. When you later
$ make install
then it this working mpg123 will be installed into /usr/local/bin
--- Senthil Murugan [EMAIL PROTECTED] wrote:
Hello All,
Currently my setup uses Xlite and Asterisk and i found that all the RTP
voice packets are transfered via the asterisk server from one xlite to
another. Is there any possibility that we can make all the RTP Packets to be
transfered
Back to the ACTUAL TOPIC of this thread... This phone looks kinda nice,
where can one get hold of it? How about it's * compatibility?
The phone works with Asterisk without problem, e.g. the standard (very
simple) sip.conf entry that DeStar creates worked out of the box with
this phone.
I
Hey All,
I now have Festival compiled, installed and running using the instructions on the Wiki
page.
When I try to change the voice that is being used however, I am running into a problem. I get
the following in the festival server log:
Cannot open file /tmp/est_10877_0/utt.wav as
On Wed, 2004-08-18 at 15:29 -0500, Christopher L. Wade wrote:
Hi all,
Anybody had problems with 'service zaptel start' on Fedora Core 2?
Works just fine on RedHat 9. On FC2, after executing 'service zaptel
start', the modprobe zaptel line from zaptel init script executes just
fine, but
You cannot compile zaphfc with latest CVS head. You have to donwload
specific date version using the download.sh included script.
BTW I have some problems with RC4. It works fine with my 2 isdn pci boards,
but it seems to be unable to drive my TDM400 ...
Try RC3, at the moment seems to be more
Would be strange if it supported SIPRTP but not UDP... I think that
most SIP support at least 10 protocols:
SIP
SDP
RTP
UDP
DNS
TFTP
DHCP
TCP
IP
ARP
802.3 (Ethernet)
;-)
Dominique
Mike Reed wrote:
1) Who bought Pingtel's phone line?
2) Anyone seen this chinese-made VoIP phone that supports 8
In data Wed, 18 Aug 2004 16:18:48 -0400, Steve Szmidt [EMAIL PROTECTED]
ha scritto:
Thanks. It works fine in Opera and IE but I guess I'll check it with some
Mozillas.
BTW the windows server is a high-reliability cluster of Linux boxes
running Apache Tomcat I have parked the page on. :-)
Hi,
I would like to have some more informations on asterisk,
I want to setup a linux based pbx and asterisk seems to be the best solution,
I have some questions for configuration:
1) I have a PRI, so I must buy a digium card to interface with PRI, right?
2) If I connect an ethernet card from
On Thu, 2004-08-19 at 17:42, Nicola Murino wrote:
I would like to have some more informations on asterisk,
Read more of the mailing list, more of the various websites like
www.digium.com and www.voip-info.org
I want to setup a linux based pbx and asterisk seems to be the best solution,
Of
On Wed, 18 Aug 2004, Maros RAJNOCH wrote:
can anyone show me a exemple config for call-back?
1) I call asterisk server from my cellular
2) asterisk hang up my call (on d-channel)
3) asterisk recall to my cellular and give me a PSTN tone, so
I can to pick up a call and to dial new phone
Hi,
thanks for your answer.
That's strange because definitely SMS messages get lost when I send them
without waiting aboout 10 seconds...
regards,
Daniel
hi daniel,
I'd guess because there are several enqueued with the same
name and when they aren't send they'll be overwritten in queue?!
I updated the file at the same URL -
http://demo.xcept.it/xc-ast/XC-AST.htm - with a new version that I tested
with Firefox too.
There are new statistics available that show agent's averages and activity
and lets you analyze more than a queue at once.
Any feedback appreciated.
l.
In data
Hi
I've two IP's(one is public and other is private). I want to use SER(SIP Express
Route) to forward my incoming request from outside to private IP. The scenario is
below:
10.x.x.x(private) == 203.x.x.x(public) == outside world(ex:192.x.x.x)
|| || ||
http://demo.xcept.it/xc-ast/XC-AST.htm - with a new version that I
tested with Firefox too.
It's still bad on Konqueror from KDE 3.1
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On Wed, 18 Aug 2004, John Morris wrote:
It's nice to be able to define the list of asterisk modules we want to
load from the /etc/sysconfig/zaptel file rather than directly in
/etc/init.d/zaptel. I'm using nufone and don't require anything but the
ztdummy (is the rtc-based module better,
Holger Schurig a écrit :
http://demo.xcept.it/xc-ast/XC-AST.htm - with a new version that I
tested with Firefox too.
It's still bad on Konqueror from KDE 3.1
But ok with Konqueror from KDE 3.3
--
Daniel
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On Thu, 19 Aug 2004, Tobias Jönsson wrote:
exten = h,2,System(echo Channel: SIP/[EMAIL PROTECTED] /tmp/${UNIQUEID}.call)
exten = h,3,System(echo MaxRetries: 2 /tmp/${UNIQUEID}.call)
exten = h,4,System(echo RetryTime: 60 /tmp/${UNIQUEID}.call)
exten = h,5,System(echo WaitTime: 30
Title: Message
I have an Internet
PhoneCard for my IBM T30 running Linux RH9 2.4.20-31.9and Asterisk
0.9.0.
I have installed the
appropriate Openh323 drivers.
The card works and I
can call to an extension that plays a recorded message, hang-up, call again and
so forth.
Suddenly, the
I'm running a fresh install of * (CVS-HEAD-08/13/04 with bristuff from
bri-stuff.0.1.0-RC4) on a Debian Sarge box...
I've got a generic HFC-S BRI ISDN card using i4l for inbound outbound
calls (I gave up on getting zaphfc working).
Asterisk is crashing out with a floating point exception a
On Thu, 2004-08-19 at 18:26, lenz wrote:
I updated the file at the same URL -
http://demo.xcept.it/xc-ast/XC-AST.htm - with a new version that I tested
with Firefox too.
There are new statistics available that show agent's averages and activity
and lets you analyze more than a queue at
Can someone confirm what I should expect the correct behaviour to be on
incoming calls if I have multiple SIP phones configured for the same
username?
I'd expect all the phones registered under the username that that extension
is associated with to ring, and the first one that answers gets it.
On Thu, 19 Aug 2004, Gary Pigott wrote:
| I'm running a fresh install of * (CVS-HEAD-08/13/04 with bristuff from
| bri-stuff.0.1.0-RC4) on a Debian Sarge box...
| I've got a generic HFC-S BRI ISDN card using i4l for inbound outbound calls
| (I gave up on getting zaphfc working).
|
| Asterisk is
John Williams [EMAIL PROTECTED] wrote:
[...]
For example, walmart.com has microtel boxes with no OS. Will RH9 and
Asterisk run on these boxes?
I can't see why not. The only thing that fouls up Linux on generic
cheap PCs is some of the weird video chipsets, but who cares that X
won't work? You
On Thu, 19 Aug 2004, David Gurr wrote:
Can someone confirm what I should expect the correct behaviour to be on
incoming calls if I have multiple SIP phones configured for the same
username?
I'd expect all the phones registered under the username that that extension
is associated with to
Hi,
I have a latest asterisk cvs sources on my box, the rest of enviroment
is:
gcc version 3.3.3 20040412 (Red Hat Linux 3.3.3-7)
linux 2.6.7
and after checkout I get:
In file included from chan_phone.c:36:
/usr/include/linux/ixjuser.h:353: error: syntax error before '*' token
make[1]: ***
On Thu, 19 Aug 2004, Peter Svensson wrote:
On Thu, 19 Aug 2004, Tobias Jönsson wrote:
exten = h,2,System(echo Channel: SIP/[EMAIL PROTECTED] /tmp/${UNIQUEID}.call)
exten = h,3,System(echo MaxRetries: 2 /tmp/${UNIQUEID}.call)
exten = h,4,System(echo RetryTime: 60 /tmp/${UNIQUEID}.call)
On Thu, 2004-08-19 at 13:40 +0200, Marcin Mazurek wrote:
Hi,
I have a latest asterisk cvs sources on my box, the rest of enviroment
is:
gcc version 3.3.3 20040412 (Red Hat Linux 3.3.3-7)
linux 2.6.7
and after checkout I get:
In file included from chan_phone.c:36:
On Thu, 19 Aug 2004, Tobias Jönsson wrote:
There is a race in this solution of /tmp and
/var/spool/asterisk/outgoing are on different file systems. Then the
rename operation (mv) is not atomic but rather a copy.
Thank you Peter for pointing out that! My mistake. Unfortunately Asterisk
Hi Manfred,
I applied the patch and recompiled and reinstalled and I got the folowing
warning during my first test call:
Aug 19 12:26:51 WARNING[294927]: dsp.c:1234 __ast_dsp_silence: zero length
packet
It looks like that could be the problem... and the fix! I'll let you know if
the problem
Hi,
We're routing SIP calls through Asterisk and we want to
be able to reinvite calls without Asterisk performing
codec conversion.
We've performed the following test:
Asterisk has license for G.729 installed
sip.conf
[general]
context=default
autocreatepeer=yes
disallow=all
allow=alaw
We have a Definity G3Si (small user base). Is anyone using Asterisk as
voicemail only with a legacy switch? If so, are there any best
practices or documents on this type of integration?
-Brian
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Hello guys,
I'm new to asterisk and I have some problems -
would you help me please.
I installed following configuration: Linux server
with Asterisk + Digium X100P FXO card
I dial the line - and I hear the voice :
Congratiolations, you succesfully installed... and so on
I want to use this
Good day ALL.
Could anyone tell me is there a way to get fax debug output into the
file when running safe_asterisk ?
--
V.8 capable
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter, V.29 and V.17
R8x7.7lines/mm
Hi,
David Gurr wrote:
Can someone confirm what I should expect the correct behaviour to be on
incoming calls if I have multiple SIP phones configured for the same
username?
I'd expect all the phones registered under the username that that extension
is associated with to ring, and the first one
Dear Pavel,
Go to http://www.voip-info.org/wiki-Asterisk
or search in google
Best Regards,
Miroslav Nachev
Hello guys,
I'm new to asterisk and I have some problems -
would you help me please.
I installed following configuration: Linux server
with Asterisk + Digium X100P
I'm experience echo on outgoing calls:
Snom 200 Asterisk T100P PRI called party
I am getting echo on the Snom 200 phone. The called party does not
hear the echo.
Since the PRI is digital, I don't really understand where the
echo is coming from.
I turned on echotraining,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 17 August 2004 10:19 am, Steve Szmidt wrote:
Hmm,
My music on hold has always worked fine. But I discovered that under
incoming IAX2 calls they don't get any MOH! All I could find was a comment
saying let me know if you find a
On Thu, 19 Aug 2004, Mike Schwartz wrote:
I'm experience echo on outgoing calls:
Snom 200 Asterisk T100P PRI called party
I am getting echo on the Snom 200 phone. The called party does not
hear the echo.
Since the PRI is digital, I don't really understand where
On Wed, 2004-08-18 at 17:01, Kris Boutilier wrote:
For the inbound digit problem try adding :
debounce=50 ; Needed to reduce the initial off hook
debounce
in the relevant context for those trunks in /etc/asterisk/zapata.conf
Also, are you using 'immediate=yes'?
On Wed, 2004-08-18 at 23:20, Adam Goryachev wrote:
On Thu, 2004-08-19 at 07:01, Seth Remington wrote:
On Wed, 2004-08-18 at 14:45, David Filion wrote:
Does anyone know of an alternate source for spandsp? opencall.org is
down and all the links returned by Google just point to the dead
We're almost there the next problem is with the inbound calls over i4l.
When the call is received, the thank you for calling. press 1 for
announcement is supposed to start... but something strange happens. The last
fraction of a second of audio from the previous call plays first (weird
I managed to record it by dialing out to the ISDN phone number. The
recording (113KB) is at
http://www.garypigott.net/files/asterisk/recording-in.wav
Regards,
Gary
- Original Message -
From: Gary Pigott [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 19, 2004 3:12 PM
Hello,
the analyzer is written as a Java model-2 webapp, that runs on any Java
application server, like Tomcat or WebSphere, using an in-house
developement architecture and MySQL for user authentication. I usually
develop it on Windows and deploy on Linux, but you can choose freely. What
I
FWI, I'm using it very successfully. Where's your problem?
My system is running Fedora Core 2, libtiff and libtiff-devel as
provided by distro (installed with apt-get, version 3.5.7). I've got to
add some headers in /usr/include to get it work: tiffiop.h, tif_dir.h
and ports.h
Hope this helps.
I'm experience echo on outgoing calls:
Snom 200 Asterisk T100P PRI called party
I am getting echo on the Snom 200 phone. The called party does not
hear the echo.
Rule of thumb (i.e. a good starting point): if you hear the echo it is
coming from somewhere else,
You cannot compile zaphfc with latest CVS head. You have to donwload
specific date version using the download.sh included script.
BTW I have some problems with RC4. It works fine with my 2 isdn pci boards,
but it seems to be unable to drive my TDM400 ...
Try RC3, at the moment seems to be more
Original Message
Message: 11
Subject: Re: [Asterisk-Users] spandsp
From: Seth Remington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Wed, 18 Aug 2004 17:01:42 -0400
Reply-To: [EMAIL PROTECTED]
On Wed, 2004-08-18 at 14:45, David Filion wrote:
Does anyone know of an alternate
Just a little update. I installed Asterisk stable of 08/19/04 and tried
to compile bri-stuff RC3 and RC4 with it. Same problem as described below.
I cant't believe that my whole asterisk setup is riuned by that §$%%$
ISDN drivers. ;-)
Maybe the cause for the problem is a missing library or
FWI, I'm using it very successfully. Where's your problem?
My system is running Fedora Core 2, libtiff and libtiff-devel as
provided by distro (installed with apt-get, version 3.5.7).
I've got to add some headers in /usr/include to get it work:
tiffiop.h, tif_dir.h and ports.h
Hope this
Mike Schwartz wrote:
I'm experience echo on outgoing calls:
Snom 200 Asterisk T100P PRI called party
I am getting echo on the Snom 200 phone. The called party does not
hear the echo.
Rule of thumb (i.e. a good starting point): if you hear the echo it is
coming from
Hey all,
I have looked around the Digium site to no avail. Can someone tell me of the
T100P is pci or pcix? Do I need to look for anything specific when selecting
a motherboard?
Also, the spec sheet says it will fit in a 2U case, but in the picture it
looks like it would be possible to squeeze
Simone Ricci a écrit :
FWI, I'm using it very successfully. Where's your problem?
My system is running Fedora Core 2, libtiff and libtiff-devel as
provided by distro (installed with apt-get, version 3.5.7). I've got
to add some headers in /usr/include to get it work: tiffiop.h,
tif_dir.h and
Hello Simone,
Here's the text from a my post earlier this week (below). Seems that (at
least) William G and I are having the same problem sending. I started out
with Fedora core 2 some months ago but had problems with Asterisk back then.
As it seems you're not now, and if RxFax seems to work
William Glynn ha scritto:
I'm seeing training errors that often stop fax machines from even
starting delivery, lots of errors from libtiff regarding unexpected
line lengths, and garbled data even if RxFax thinks it's okay.
Strange. Which version of libtiff are you using?
However, I've gotten TxFax
administrator tootai ha scritto:
From where you got port.h? I install tiff-3.5.7 (I'm running an RH73
with those rpms installed) and didn't manage to create this file.
tiffiop.h and tif_dir.h are parts of tar.gz package. I had to modify
tiffiop.h and replace port.h with tiffcomp.h but I'm
Title: Can PSTN CallerID be fowarded to a SIP phone extension?
Hi All,
I have a server setup with an incomming PSTN line and a bunch of
Grandstream BT100 phones. Is there a way for asterisk to foward an
incomming callerID from the PSTN to the SIP phone that is setup as an extension? We
T100P is regular PCI and will fit in a 5V slot like over 90% of the
motherboards out there have. It will fit vertically in a 2U case(we have one
in a 2U case), but for a 1U case you need a riser to mount it
horizontally(because the card is over an inch and a half high).
MATT---
-Original
Jon Bebeau ha scritto:
1) Where did you get the C headers from (what dir).
tiffiop.h and tif_dir.h somewhere in the libtiff's 3.5.7 source .tar.gz
port.h with google's help. I can send my version to you if you wish. It
worked for me.
2) What HW are you using (processor, motherboard)
Intel(R)
I've seen people mention that they have fax reception working with
Asterisk, spandsp, and app_rxfax.
I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest
app_rxfax.c (as mirrored by friendly list members recently), and libtiff
3.5.7. Asterisk is detecting the fax signal
I'm seeing training errors that often stop fax machines from even
starting delivery, lots of errors from libtiff regarding unexpected
line lengths, and garbled data even if RxFax thinks it's okay.
Strange. Which version of libtiff are you using?
I'm using Gentoo's tiff-3.5.7-r1 package,
On Thu, Aug 19, 2004 at 12:07:09PM -0400, James Freire said:
I have a server setup with an incomming PSTN line and a bunch of
Grandstream BT100 phones. Is there a way for asterisk to foward an
incomming callerID from the PSTN to the SIP phone that is setup as an
extension? We have a Voice
Title: Message
That
should happen automatically -- it does on my system.
-Original Message-From: James Freire
[mailto:[EMAIL PROTECTED] Sent: Thursday, August 19, 2004
11:07 AMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone
James Freire wrote:
Hi All,
I have a server setup with an incomming PSTN line and a bunch of
Grandstream BT100 phones. Is there a way for asterisk to foward an
incomming callerID from the PSTN to the SIP phone that is setup as an
extension? We have a Voice menu setup for incomming calls and I
Sorry about that. I am in the US and using the Digium FXO TDM400 and I have enabled
all the callerID options in my zapata.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walt Reed
Sent: Thursday, August 19, 2004 12:33 PM
To: [EMAIL PROTECTED]
I'm not sure that the problem lies in the NAT because the phone is talking
to Asterisk. I'm hoping this is a simple config thing I've overlooked but
I've tried all kinds of combos inside the [] in my mgcp.cfg file.
The phone's IP is 192.168.1.116 (my comp is .110). The router to which the
phone
Has anyone tried using a debit/credit card terminal as such:
Terminal - SPA-2000 - Public Internet - * - PRI
I'm hoping someone will tell me they have done this successfully and
rarely experience dropped calls. Though I'd like to hear from anyone
who has tried and failed as well.
Thanks,
forget the asterisk source you have downloaded.
Zaphfc is not only a driver, it's a patch that have to be applied to
specific source version too,
You have to run the install.sh script that is included in the tarball.
This script before downloads the right asterisk version (download.sh) and
then
Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 19:31, Ryan Courtnage wrote:
Theoretically, I know it's possible, but is any using multiple tdm400ps
(fxo) in single * box? In a production environment? Any gotchas aside
form irq sharing?
Buy a T100P and a channel bank with two FXO modules.
On Thursday 19 August 2004 13:02, Trevor Peirce wrote:
Has anyone tried using a debit/credit card terminal as such:
Terminal - SPA-2000 - Public Internet - * - PRI
Pretty much.
Terminal - Adit600 FXS - * - 1 hop - * - PRI
Must use ULAW and must have a decent connection (low low jitter and
On Thu, 2004-08-19 at 10:47, Christopher Jacob wrote:
Hey all,
I have looked around the Digium site to no avail. Can someone tell me of the
T100P is pci or pcix? Do I need to look for anything specific when selecting
a motherboard?
Also, the spec sheet says it will fit in a 2U case, but
Hi Matthew,
-Original Message-
I'm not sure that the problem lies in the NAT because the
phone is talking to Asterisk. I'm hoping this is a simple
config thing I've overlooked but I've tried all kinds of
combos inside the [] in my mgcp.cfg file.
from 64.72.107.1:52539MGCP read:
Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
From an AGI script so people can dial #* to hang up (and other things) but
when using AGI using 'EXEC DIAL IAX2/bla/1234567|50|tT' it dials ok but
nothing happens when they dial #, is there something special I need to do to
escape
James Freire wrote:
Sorry about that. I am in the US and using the Digium FXO TDM400 and I have enabled all the callerID options in my zapata.conf file.
Of course, the BT100 has only a numerical display, and will not display
CIDName, only number.
___
On Thursday 19 August 2004 13:00, Ryan Courtnage wrote:
A Rhino channel bank with 12 fxo will retail to about $1850, plus $500
for the t100p. It's a tough sale for a growing small company that has
already invested in (and outgrown) 2 tdm400ps.
So give them partial (say 50%) credit to take
I have the programable button led's working properly on my snom 200
except they don't flash during a ring event. I found a post by Andre
Bierwirth saying he had a patch that he submitted but didn't make it
into CVS. I would like to get a copy of that as a starting point to
implement button
Hi,
I finally switched (again) to chan_oh323, which
compiles without problems on opteron 64bit.
Roger.
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Wow.
[00059002042b]
context=main
host=dynamic
callerid = John Doe 123
nat=yes
Line = svip10
That did it. The phone registered with * and a debug msg flys up when I
pickup/put down the reciever.
When I pick up the handset, I can hear a dialtone. But pressing numbers on
the keypad doesn't
On Aug 19, 2004, at 10:00 AM, Ryan Courtnage wrote:
Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 19:31, Ryan Courtnage wrote:
Theoretically, I know it's possible, but is any using multiple
tdm400ps
(fxo) in single * box? In a production environment? Any gotchas
aside
form irq sharing?
On Thu, 2004-08-19 at 12:26, Morgan Gilroy wrote:
Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
From an AGI script so people can dial #* to hang up (and other things) but
when using AGI using 'EXEC DIAL IAX2/bla/1234567|50|tT' it dials ok but
nothing happens when they
Massimo De Nadal schrieb:
forget the asterisk source you have downloaded.
Zaphfc is not only a driver, it's a patch that have to be applied to
specific source version too,
You have to run the install.sh script that is included in the tarball.
This script before downloads the right asterisk version
Does anybody know what happened to the opencall.org website? I can't get
into the home page or the ftp site.
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Does anybody have any solution to get MGCP UserAgent support in *.
With the explosion of VOIP provider these days, this would allow tapping
into these providers that chosen MGCP as their protocol.
I have the CallAgent side working fine.
Thanks
MarcG.
On Aug 19, 2004, at 11:30 AM, Muiz Motani wrote:
Does anybody know what happened to the opencall.org website? I can't
get
into the home page or the ftp site.
His DNS servers seem to be down. opencall.org is served by
name[12].coppice.org, which are 202.76.92.17[23]. Neither one responds
to
Hello all,
I currently have an Eicon Diva Client isdn card using i4l. Outbound
dtmf doesn't work (and never has), but there has been an annoying
problem with false dtmf detection in calls (that could be triggered
easily by blowing into the receiver on the remote end).
I looked through the list
Andrew Kohlsmith wrote:
Terminal - Adit600 FXS - * - 1 hop - * - PRI
Must use ULAW and must have a decent connection (low low jitter and latency).
We have incoming and outgoing faxes doing this without issue. I can't get
faxes to go through Nufone with any kind of steady success but it's
I have loaded the latest cvs (19/08/04). When I try to compile I receive the
following error:-
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
ast_expr.y:110: unrecognized: %locations
ast_expr.y:110:Skipping to next %
ast_expr.y:141: invalid @-construct
ast_expr.y:141: $. is invalid
On Thu, 2004-08-19 at 13:56, [EMAIL PROTECTED] wrote:
I have loaded the latest cvs (19/08/04). When I try to compile I receive the
following error:-
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
ast_expr.y:110: unrecognized: %locations
ast_expr.y:110:Skipping to next %
On 19 Aug 2004 at 11:44, you wrote:
On Aug 19, 2004, at 11:30 AM, Muiz Motani wrote:
Does anybody know what happened to the opencall.org website? I can't
get
into the home page or the ftp site.
His DNS servers seem to be down. opencall.org is served by
name[12].coppice.org, which
Hello,
I have a Pingtel Xpressa and trying to get it working with *. When the
phone tries to register, it sends out a REGISTER request and * replies
with PROXY AUTHENTICATION but phone never replies back with the right
info and just sends REGISTER again and again. This is what Pingtel
support
Guten Tag,
ich bin im Moment auf einer ausreichend einsamen Inse
und werde Ihre Mail lesen, wenn ich wiederkomme.
In Notfällen bin ich unter +49-160-4439821 zu erreichen.
=
Hello,
I am currently on a reasonably remote island,
and I'll read your email when I get back.
In case of an
This is repetetive.
A message from yesterday from seth:
On Wed, 2004-08-18 at 14:45, David Filion wrote:
Does anyone know of an alternate source for spandsp? opencall.org is
down and all the links returned by Google just point to the dead site.
Thanks
David Filion
I threw a copy up here for
Title: Does Granstream BT100 Conference Button Work?
Hi All,
I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as
This brings up a good point that has had me scratching my head for a long
time. Is there a good searchable archive of the asterisk mailing lists? I don't
particularly want to download and keep updated the full 206 MB of the
asterisk-users .mbx file on my laptop. The current format is just not
On Thu, 19 Aug 2004, Anton Yurchenko wrote:
The Asterisk sends the replies to port 1031, the outbound port that
Pingtel used to send the message. This is wrong. In the REGISTER,
Pingtel specified a contact header field with no port, which means use a
default port of 5060. Asterisk is
Im using 'notransfer=yes' in the iax.conf so it shouldn't happen,
What I'm doing is bridging a 2 legged call over iax using .call files.
The .call file initiates the first leg and drops the user into a context
that calles an agi script that checks against a db (like max call length)
then
Steven,
Spot on. I haven't checked out since June 26th. However, my bison was the
errr lagging or shall we say pre-historic (v1.28) :)
Thanks for the clue.
Best regards
Steve Beaumont
On Thu, 2004-08-19 at 13:56, [EMAIL PROTECTED] wrote:
I have loaded the latest cvs (19/08/04). When I try
Title: Does Granstream BT100 Conference Button Work?
Nope, it does nothing... It's not an * problem
either, the button just does nothing... I think they're planning on making it
work in a future release, don't quote me on that... for now it just occupies
space..
-Chris
- Original
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