First of all thanks for the patch it works great,
but i think it breaks the distinctive ringing,
I have 2 incoming numbers in one x100p in contexts home1 and home2 but
'default' is always chosen has anyone else seen this?
if you need any more info just ask
Robb
Tony Hoyle wrote:
David J Carter
Hi TH,
Asterisk works fine as a Voicemail only server. I have it setup like that in
a production setup.
Configuration is simple, I will try and post something here soon. What will
you integrate it with ? another asterisk system ?
Umar.
-Original Message-
From: [EMAIL PROTECTED]
Philipp von Klitzing wrote:
Hi!
my problem is to forwarding a call to a SIP phone and record the call at
the same time. How can I do?
This should help you to solve your problem:
http://www.voip-info.org/wiki-Monitor+setup+sample
Cheers, Philipp
Cheers Tony.
Your a star.
Works a treat.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle
Sent: 28 May 2004 00:48
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Caller ID with BT CD50
David J Carter wrote:
Where would I find
I'm going to have to go against this statement, there's one bug that I
need to fix so unfortunately it will have to be Monday now.
For those after the IAX/SIP firefly (albeit an old version) get
http://www.virbiage.com/firefly/download/firefly-dev.exe
apologies,
Adam
Adam Hart wrote:
They'll
Umar,
The plan is to integrate with a Cisco Callmanager.
We currently have a very old VM system, based on a Netscape product that was
installed before my time.
The current project is to upgrade CM and replace the voicemail. I think
Asterisk will do the job for us, now I just need to convince the
Robert Boardman [EMAIL PROTECTED] wrote:
First of all thanks for the patch it works great,
but i think it breaks the distinctive ringing,
I have 2 incoming numbers in one x100p in contexts home1 and home2 but
'default' is always chosen has anyone else seen this?
Yes - it does break the
Darren, yes, I'd be happy to help.
I'll contact you off list to sort out the
arrangements.
I should warn you that it may be a
wasted journey for you, as I really
dont know if it will exhibit the problem.
Tim.
Storer, Darren [EMAIL PROTECTED] wrote:
__
Hi Tim,
TP So it _may_ not be a
Hi Steve,
SU If you are using CTR4, then I guess they use CTR4. :-)
SU CTR4 == Net 5 == various other names == EuroISDN.
Reasonable logic but bad assumption in this case.
The Dialogic Q.931 stack (D/300, DM3 etc.) is solid and quite tolerant of
ISDN 85 as are most hardware PBXs. Other (PC
Hi, I successfully installed 2 avm card
in my asterisk box but I'm unable to make call. My capi.conf is:
msn=072,0725
incomingmsn=*
controller=1,2
softdtmf=1
context=default
echocancel=yes
callgroup=1
devices=2,2
my capi info :
Contr1: 2 B channels total, 2 B channels
free.
Contr2: 2
Kevin,
Could you add this to
http://bugs.digium.com/bug_view_page.php?bug_id=0001719
Chris
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 28, 2004 9:12 AM
Subject: RE: [Asterisk-Users] Caller ID with BT CD50
Robert Boardman [EMAIL
Hi to all!!
Is there another method to download asterisk
addons???
Thanks
F
I am unable to get a my Draytek working with our Asterisk server. I can
make/recieve calls but get no audio. I have tried the various codecs at the
Vigor end but still getting nothing. I looked at sip debug (below) but am
new to Asterisk and don't really know what I am looking for. Asterisk
was wondering if someone could give any indication of the messages that are
appearing on the console of an Asterisk PBX
WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 103 (non-critical request)
192.168.90.1 is a 7940 ip phone
[EMAIL PROTECTED] wrote:
msn=072,0725
[..]
== found capi with omsn =072
May 28 10:36:56 NOTICE[180241]: app_dial.c:655 dial_exec: Unable to
create channel of type 'CAPI'
== Everyone is busy at this time
Are you sure, that your format for the msn definition is correct for
Italy?
Just tried to apply the patch:
Just checked out asterisk stable and zaptel, patched using Tony's
patches (which worked, and compiled previously)
Then got this when applying your patch.
bash # cat ../chan_zap.c.diff | patch -p0
patching file channels/chan_zap.c
Hunk #1 succeeded at 4642 (offset
Hi Everybody
Any significant changes to CVS HEAD over the last couple of days. I've got
two asterisk boxes - both on public IP but one is dynamic. The one on
dynamic IP registers at the other one - that part is fine.
Calls going from the one with dynamic to the static one goes fine.
Call the
usedcanon [EMAIL PROTECTED] wrote:
Thanks, suddenly makes sense now. I guessed that is the case however
was not sure. Any opinion on what is more/most efficient, using a
scripting language like perl or a compile app in C/pascal.
Define efficient.
A C program would normally be expected to be
hi Peter,
Your feedback is greatly appreciated. Having not done
any AGI before I was not sure what to expect. My
requirements are very basic at the moment, and time as
you say is money. my best option is to find something
simmillar and customise it to my needs.
Umar.
--- Peter Corlett [EMAIL
On May 27, 2004, at 11:01 PM, Aaron J. Angel wrote:
Michael George wrote:
But, this isn't a big deal, we can live without it. I just
thought there might be a way. If I could do a
Backtround(Playtone()), that would do what I want...
There's no need for that. The playtone application continues to
Karl Dyson [EMAIL PROTECTED] wrote:
Just checked out asterisk stable and zaptel, patched using Tony's
patches (which worked, and compiled previously)
Then got this when applying your patch.
bash # cat ../chan_zap.c.diff | patch -p0
patching file channels/chan_zap.c
Hunk #1 succeeded at
Karl Dyson wrote:
Just tried to apply the patch:
Just checked out asterisk stable and zaptel, patched using Tony's
patches (which worked, and compiled previously)
Then got this when applying your patch.
bash # cat ../chan_zap.c.diff | patch -p0
patching file channels/chan_zap.c
Hunk #1 succeeded
Michael George [EMAIL PROTECTED] wrote:
I get the 9 and start PlayTones().
I go to the next context (with the tones playing).
In the next context (tones still playing) my matches are all several
digits long, so the tone is playing as the digits are pressed. That is
disorienting because
I there a problem with CVS ? My card TDM04B does not want to answer calls
on 2 ports. Strange.
Yes there is a problem. Pull an older copy of wcfxs.c in zaptel (from about
5/24) and it will work again. Mark is aware of the problem.
___
Oddly, it looks like the changes were made(!?)
It might be, having read Tony's reply, that it's because I applied the
uk cli patches from Tony and yourself to the stable rather than head
branches?
I'll try compiling and let you know.
Cheers for now,
Karl
-Original Message-
From:
I am using CISCO 30 VIP and CP 12+ IP phones. I am
using 2 analog phones connected to a SIPURA. I am
using chan_skinny for the CISCO phones. On the CISCO
phones, only the basic phone functionality works. I
can not transfer calls or anything using the
chan_skinny. The analog phones also work as
Hi Lars,
I met the same problems yesterday and even posted it to the list.
Unfortunately nobody answered yet.
Is it so clear to solve that no one is willing to help us? :-/
Regards,
Julian Pawlowski
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Well compiles and runs OK, but it doesn't identify the dring. I only
started playing with it this morning (only realised it *did* dring when
I saw your it's broken dring post)
This is what I have in zapata.conf
dring1=95,0,0
dring1context=inbound-pstn-1
dring2=325,95,0
I am using CISCO 30 VIP and CP 12+ IP phones. I am
using 2 analog phones connected to a SIPURA. I am
using chan_skinny for the CISCO phones. On the CISCO
phones, only the basic phone functionality works. I
can not transfer calls or anything using the
chan_skinny. The analog phones also work as
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our current PBX which will just connect us to an outgoing
line (with a dialtone) when we hit 9. (Though I don't want asterisk to
mimic that
Title: Message
Is there an
interface (direct or indirect)in Asterisk that can be used by JTAPI to do
third party call control and the other functionality supported by
JTAPI?
Does anyone have an
example of such a thing?
Jim
Karl Dyson [EMAIL PROTECTED] wrote:
Well compiles and runs OK, but it doesn't identify the dring. I only
started playing with it this morning (only realised it *did* dring when
I saw your it's broken dring post)
This is what I have in zapata.conf
dring1=95,0,0
Hi all,
I just upgrade my ix66 ...
the new firmware 2.07 have this:
(SIP) Tolerance against Asterisk PBX registration
deviation.
regards
Miklos
Kevin Walsh wrote:
Yes - it does break the distinctive ring detection, but that's easily
sorted out.
Actually it's the first time I've ever heard of distinctive ring being
available in the UK... :)
The correct way would be to move the if (p-use_callerid == 2)
code within the existing if
Karl Dyson wrote:
Well compiles and runs OK, but it doesn't identify the dring. I only
started playing with it this morning (only realised it *did* dring when
I saw your it's broken dring post)
This is what I have in zapata.conf
dring1=95,0,0
dring1context=inbound-pstn-1
dring2=325,95,0
In article [EMAIL PROTECTED],
Fabio Donaggio [EMAIL PROTECTED] wrote:
Hi to all!!
Is there another method to download asterisk addons???
Another method in addition to what?
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] -
Michael George [EMAIL PROTECTED] wrote:
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our current PBX which will just connect us to an outgoing
line (with a dialtone) when we hit 9. (Though I
Julian Pawlowski [EMAIL PROTECTED] wrote:
I met the same problems yesterday and even posted it to the list.
Unfortunately nobody answered yet.
Is it so clear to solve that no one is willing to help us? :-/
It sometimes helps if you quote some context above your text.
--
_/ _/
On Fri, 28 May 2004, Michael George wrote:
Yes, I see what you are saying. And I tried this. Here's what happens:
I get the 9 and start PlayTones().
I go to the next context (with the tones playing).
In the next context (tones still playing) my matches are all several
digits long, so
The code changes that fixed the cisco choppy sound for Stable went in last
Friday. That change corrected iax2 issues that had been known for well over
a month but never got applied to Stable. That same code is in Head, however
many other changes have happened to Head, and some of those apparently
Hi!
Newbie question; my server has no sound card, in effect I have commented out
the loading of alsa and oss modules.
When I make a call I do not here any sound however I do notice the activity
from the tethereal trace and the debug.
Is there a relation? I would think so but I am just shocked
Tony Hoyle [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
Yes - it does break the distinctive ring detection, but that's easily
sorted out.
Actually it's the first time I've ever heard of distinctive ring being
available in the UK... :)
It costs the same as Caller*ID, so I just got it to
- Original Message -
From: Fabio Donaggio
To: [EMAIL PROTECTED]
Sent: Friday, May 28, 2004 6:16 AM
Subject: [Asterisk-Users] Asterisk addons
Hi to all!!
Is there another method to download asterisk addons???
Thanks
F
Man! Try to investigate for yourself! Use google!
Hi to all!!
I'm successful to connect Asterisk to MySQL database...
Can anyone learn me how to store sip user in
MySQL database and how to configure voicemail??
Thanks for all!!!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Fri, 2004-05-28 at 07:59, Rich Adamson wrote:
Although many of us that have worked in a production I/T arena assume
something called Stable would truly have known bugs fixed, that's hardly the
case for *. That branch really should be renamed to something like v1.0 and
remove any reference
Hermann Wecke wrote:
On Thu, 27 May 2004, Harry Flink wrote:
www.cvshome.org is home for CVS but the site is currently down.
Is down due to security issues:
I'm surprised that was exploitable... it's much more likely to crash the
server than do anything nasty.
That's the patch that sourceforge
It's true, if you're not careful, you could give incoming callers access
to your outside lines. But it is possible, with careful use of contexts,
to ensure that callers coming in on the context you specify for incoming
calls does not have access to the context that contains the dialplan for
On Fri, 2004-05-28 at 08:13, Fabio Donaggio wrote:
Hi to all!!
I'm successful to connect Asterisk to MySQL database...
Can anyone learn me how to store sip user in
MySQL database and how to configure voicemail??
Can I learn ya with a 2x4?
BTW, what happened to your postgres connection
Graham Turner wrote:
was wondering if someone could give any indication of the messages that are
appearing on the console of an Asterisk PBX
WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 103 (non-critical request)
192.168.90.1 is a
Kevin Walsh wrote:
I applied your new patch but it resulted in the caller hearing a ring
tone but no phones actually ringing. I don't have time to look into
it right now, but I'll take a look later and see what's going on.
I put my chan_zap.c back in and the re-tests were ok.
I've changed it
Lars Boegild Thomsen wrote:
Hi Everybody
Any significant changes to CVS HEAD over the last couple of days. I've got
two asterisk boxes - both on public IP but one is dynamic. The one on
dynamic IP registers at the other one - that part is fine.
Calls going from the one with dynamic to the static
If you have used * to support a pri as pri_net (as opposed to pri_cpe),
either to talk to another * system or a PBX of some sort, I would be very
interested in hearing about your experiences. Imparticular, I would like
to know that it works before I invest in the extra hardware.
TIA
Bruce
I've made a couple of small contributions to the wiki but recently I
read the Terms of service, they are pretty draconian:
Download (other than page caching), or modify this site.
Reproduce, duplicate, copy, sell, resell, visit or use for other
commercial purposes this site or any
Ignace CARIA wrote:
I know I know this subject have been The most written subject about VoIP
:-)
If Asterisk is on a Public IP Address and a softphone behind the nat,
sip.conf must contains for this phone: nat=yes
And in most cases qualify=yes
The nat=yes makes asterisk don't trust the
Isn't it about time to lock down added functionality to v1.1 and fix
the remaining bugs?
There has been a significant amount of traffic on the cvs list, the irc
and other channels with folks spending time adding new functionality to
Head. Think its time to lock it down, fix the bugs that have
Michael George [EMAIL PROTECTED] wrote:
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our current PBX which will just connect us to an outgoing
line (with a dialtone) when we hit 9. (Though I
Although many of us that have worked in a production I/T arena assume
something called Stable would truly have known bugs fixed, that's hardly the
case for *. That branch really should be renamed to something like v1.0 and
remove any reference to Stable and bug fixes as its treated as a
Hi Rich,
Sounds like a good idea.
Umar
--- Rich Adamson [EMAIL PROTECTED] wrote:
Isn't it about time to lock down added functionality
to v1.1 and fix
the remaining bugs?
There has been a significant amount of traffic on
the cvs list, the irc
and other channels with folks spending
Hello Olle!
Please add a SIP debug of the call so we can see what happens, who
refuses what call.
Situation: I'm behind an NAT firewall and get an incoming call from my
SIP provider. I have the following entries in sip.conf:
register = 1838933:[EMAIL PROTECTED]/1838933
[sipgate.de]
type=user
Rich Adamson wrote:
It's a known fact that bugs are not being fixed in Stable, and even Mark
has suggested no one should be running Stable in a production environment.
On the other hand, there's not many bugs open in the bug tracker. Feature
requests and patches, but not bugs.
If you are aware of
Rich Adamson [EMAIL PROTECTED] wrote:
Isn't it about time to lock down added functionality to v1.1 and fix
the remaining bugs?
There has been a significant amount of traffic on the cvs list, the irc
and other channels with folks spending time adding new functionality to
Head. Think its time
Hi all.
I have and E1 channel bank from Loop Telecom.
there's a little issue with it, I cannot ring
the phones on fxs interface, but can connect
without issue them.
What happens:
I dial the phone on port 1, asterisk says
Zap/1 is ringing, but the phone on the
analog port doesn't ring. but if I
On Fri, 2004-05-28 at 08:37, Andrew Kohlsmith wrote:
Please do not trim out attribution tags.
The double quoted is from Julien Levi [EMAIL PROTECTED]
What worries me most is that the current terms seem crafted so as to
ensure that should the people who run voip-info ever decide to remove
I'd like to be able to add additional fields to the the Asterisk
database. I'm using Mysql for most of my data lookup and manipulation,
and it seems to work pretty well. In keeping with what I know how to do,
it would be very handy to be able to insert say a call forward number
into a customer
Please do not trim out attribution tags.
The double quoted is from Julien Levi [EMAIL PROTECTED]
Why not? I replied to Julien Levi's post, so the attribution should be
implied, just as I am replying to your post, and I don't have a Steven
Critchfield sez line... I've been doing this for
We are using Asterisk as pri_net connected to Merlin Legeng over DS100
card. It works quite stable and did not see any problem for past months.
Here is my configs:
=== /etc/zaptel.conf
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
Title: Re: [Asterisk-Users] Wiki TOS - worrying for an open sourceproject?
I am
using snom200 phones registering with Asterisk via SIP. I can see where the
phone registers without a problem, and then when you try and make a call I get a
proxy authentication required message on the phone and
I've done this too. Four E1's on one box, talking to four E1's on another
asterisk box. I just use it for load testing new Zap versions.
Note that you need a crossover E1 cable for this.
Cheers
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London
Hello all,
We have a TE405P set up with span 1 running to a channel bank, a PRI running
into span 2, and a PPP internet T1 running into span 3. We have the first 2
spans up and running without a problem. We have hdlc compiled into the
kernel and after making the appropriate changes to
On Thu, 2004-05-27 at 22:07, Aaron J. Angel wrote:
Did you know that by clicking reply, one is following proper netiquette? It
is especially helpful for those using threaded mail readers. On top of
that, if people delete messages simply because they don't like the subject,
who's problem is
The failure has just been fixed as I saw in mantis:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738
Thanks a lot! ;D
Regards
Julian Pawlowski
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi!
I've made a couple of small contributions to the wiki but recently I
read the Terms of service, they are pretty draconian:
[...]
What worries me most is that the current terms seem crafted so as to
ensure that should the people who run voip-info ever decide to remove
content, or stop
On Fri, 28 May 2004, CW_ASN wrote:
- Original Message -
From: Fabio Donaggio
To: [EMAIL PROTECTED]
Sent: Friday, May 28, 2004 6:16 AM
Subject: [Asterisk-Users] Asterisk addons
Hi to all!!
Is there another method to download asterisk addons???
Thanks
F
On Fri, 2004-05-28 at 08:34, Bruce Komito wrote:
If you have used * to support a pri as pri_net (as opposed to pri_cpe),
either to talk to another * system or a PBX of some sort, I would be very
interested in hearing about your experiences. Imparticular, I would like
to know that it works
Hi!
It's all ok with CVS login...I download
asterisk-addons.
I would try to store sip friends in MySQL database
and also the voicemailcan you help me???
Thanks
HI there,
Thanks everybody for all the answers. I took a look at the
asterisk timer ztdummy page
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy)
Unfortunaly, my PC has the USB OHCI module. So, I downloaded the zaprtc module
from
On Fri, 28 May 2004, Olle E. Johansson wrote:
Rich Adamson wrote:
It's a known fact that bugs are not being fixed in Stable, and even Mark
has suggested no one should be running Stable in a production environment.
On the other hand, there's not many bugs open in the bug tracker. Feature
Andrew Kohlsmith wrote:
Please do not trim out attribution tags.
The double quoted is from Julien Levi [EMAIL PROTECTED]
Why not? I replied to Julien Levi's post, so the attribution should be
implied, just as I am replying to your post, and I don't have a Steven
Critchfield sez line... I've
What is your kernel version?
Patrick J. Conroy wrote:
Hello all,
We have a TE405P set up with span 1 running to a channel bank, a PRI running
into span 2, and a PPP internet T1 running into span 3. We have the first 2
spans up and running without a problem. We have hdlc compiled into the
kernel
I'm willing to open my system up for those developers that cannot duplicate
the problem on their own systems. I have a nice flat network, good
hardware, no off-the-wall configurations, an up-to-date kernel and server
hardware, etc. Contact me on or off list and I'll arrange for SSH access
for
From: Steven Critchfield [EMAIL PROTECTED]
The other part is that a wiki is really unmirrorable using normal
methods of mirroring a site. You need to just run the same software and
have the database behind it mirrored. I'm sure if the wiki is running a
new enough version of mysql, and the
The disconnect between HEAD and stable is what concerns me. The fact that a
fix was put into Stable for the choppy audio on Cisco -*-IAX that I
couldn't find in HEAD, and that didn't work when fetching and rebuilding
HEAD is what concerns me. If it exists in stable (and works in stable), but
We are writing a program using the manager for * for our receptionist
to use once the system go live. If anyone is interested in helping us
with testing please let me know.
We are designing it for a touch screen monitor for her to do transfers,
see whose on the phone and a few other features.
Bruce Komito wrote:
If you have used * to support a pri as pri_net (as opposed to pri_cpe),
either to talk to another * system or a PBX of some sort, I would be very
interested in hearing about your experiences. Imparticular, I would like
to know that it works before I invest in the extra
As a sidenote, your site doesn't work in Mozilla Firefox.
--
Vice President of N2Net, a New Age Consulting Service,
Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E. Johansson
Sent: Friday, May 28, 2004 9:38 AM
To: [EMAIL PROTECTED]
Cc: Asterisk-a-users-list
Subject: [Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
On the other hand,
On Fri, 2004-05-28 at 10:53, pesb wrote:
HI there,
Thanks everybody for all the answers. I took a look at the
asterisk timer ztdummy page
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy)
Unfortunaly, my PC has the USB OHCI module. So, I downloaded the
does anyone have an example they would please share for turning on stutter
dialtone for a zaptel channel when there is a message waiting?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
___
Asterisk-Users mailing list
I have digium E1s as pri_net connected to nms based softswitch - no problems
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito
Sent: Friday, May 28, 2004 3:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] * as pri_net?
If you have
The reason why I would like to use Freenet iPhone is their cheap rate
for calls to Germany (1 cent/min). It is correct that you have to sign
up for one of their DSL plans. But the pay as you go plan has neither
monthly fee and nor a minimum usage requirement.
The lack of incoming phone number /
On Fri, 2004-05-28 at 10:23, Andrew Kohlsmith wrote:
Please do not trim out attribution tags.
The double quoted is from Julien Levi [EMAIL PROTECTED]
Why not? I replied to Julien Levi's post, so the attribution should be
implied, just as I am replying to your post, and I don't have a
We are using redhat 8 with kernel 2.4.18-14. We recompiled the kernel with
the hdlc-2.4.20-1.14a.patch from http://hq.pm.waw.pl/hdlc/. That site
stated that this was the patch to use for 2.4.20 and earlier kernels. The
kernel seemed to compile and sethdlc seemed to compile fine and the hdlc
I'll RPM up whatever you guys decided to drop, and continue to run
1.0_stable on my production boxes and provide feedback to the Bug
Marshalls.
I'll do slackware 9.1 packages for anyone interested if there aren't any other
maintainers...
-A.
___
Hi!
The failure has just been fixed as I saw in mantis:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738
Unfortunately that didn't solve my problem - however I am not sure
anymore that this is related, and maybe I just have a basic
misunderstanding concerning type=peer and
Hello louis,
Friday, May 28, 2004, 6:32:33 PM, you wrote:
lg Hi Alessio
lg Thank you for the reply. Our configuration is as follows
lg Asterisk Server 192.168.0.250 is on our LAN
lg Vigor 192.168.1.1 connects to the LAN VPN (vigor to vigor)
lg Laptop 192.168.1.10 with XLite
I can suggest this:
On Fri, 28 May 2004, Tony Hoyle wrote:
Actually it's the first time I've ever heard of distinctive ring being
available in the UK... :)
BT launched Call Sign sometime in 1996.
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I can't get spandsp to compile. when I go to the */apps directory i
continually fails.
Makefile:80: warning: overriding commands for target `app_rxfax.so'
Makefile:77: warning: ignoring old commands for target `app_rxfax.so'
cc -fPIC -c -o app_rxfax.o app_rxfax.c
app_rxfax.c:45: error:
1.1 (today's head) is more of a let's try if this works' release.
Please spend time testing it. Remember, CVS HEAD, is not meant to be
stable. Now and then, it might not even compile cleanly. It's
a developer's release, at some point in future aimed to be stable.
Surely this is the reason of
Hey Vasyl,
this doesn't bode well for me I am going to hate having to recompile a new kernel, and zaptel, asterisk, etc, and restart everything This sucks
M
On Sunday, May 23, 2004, at 12:40 PM, Vasyl Rublyov wrote:
Thank you Michael,
I used that sethdlc which is in latest
For those still impacted by the iax2/gsm/cisco choppy sound, please add your
comments to bug #1742. The source of the problem tends to be the asterisk
box originating the iax2/gsm data flows (eg, if you hear choppy audio, the
* box at the distant end is the one originating inconsistent timestamps)
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