Hello all - I'm just starting to play around w/ asterisk, and I've run
into a seemingly simple problem that has really manged to frustrate
me...
I'm running the latest cvs version of *, and am trying to dial in to
the default extention 1000 demo using x-lite. I can dial and hear the
greeting no
Hey guys,
Cal someone help me. I'm register voiceplus DID i try to config
fllow example but not work. When i test call to number and debug
iax2 in my asterisk not found packet.
My iax.conf
register = in-xxx:[EMAIL PROTECTED]
[voicepulse]
context = voicepulse-incoming
secret=yyy
Consider hot swappable SCSI RAID 1 instead of IDE. You'll appreciate it
every couple of years when you lose a disk but the PBX stays up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Saturday, August 21, 2004 2:23 PM
To: [EMAIL
Thanks for help - already found the right way in dev-Mailinglist.
BetaTeilchen schrieb:
Hi !
Just found the beta-firmware-release for iaxy. Which way to get it
into IAXy ?
Thx for help.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
I'm a student and my thesis work consist in testing
Asterisk with Swyx(SwyxWare).
My approach is to declare asterisk as h323 gateway for
the Swyxserver using oh323 Plugin.
Is there any possibility to connect Asterisk with
Swyx? how?
the outgoing call must pass from Swyxit-to
Swyxserver-
Lol all of these would look pretty funny plastered inside a wall... I think
you would be better using an ATA adapter and a regular analog DoorPhone or
Intercom. Then you get the best of both worlds... it's cheap... and it uses
SIP...
The GrandStream HT486 and also the 286 has a feature where it
Why didn't I think of this before! Better yet, use an analog doorphone or
intercom and an IAXy! I haven't had the pleasure of using one yet but I'll
bet they can do some pretty neat tricks, especially since they're speaking
IAX! :)
-Chris
___
Just a little pun there!
I've been mostly lurking for a couple of weeks and realize how little I
know and understand about this PBX and phone stuff. I did a little
looking about and came across a glossary but they terms are -- for me --
kind of out of context. I'm wondering if there is (much as
[EMAIL PROTECTED] wrote:
Geoff Nordli wrote:
I am trying to compile the meetme2 application with the latest CVS
head and it fails. Here is the error message that I get. Can
someone point me in the right direction?
[snipped lengthy error message]
Here's a patch. You can apply it by hand,
Hi guys,
I got 2 different linux boxex each one with a TDM22B (2 fxs and 2fxo). Bot
linuxes are connected to the same dsl company with a dynamic ip and both are
doing an ip_gre tunnel VPN. I could work with zapta.conf zapata.conf and
extensions.conf locally on each linux. Each linux is
Hello All,
Thanks for your suggestion.I had changed my sip.conf
as your advice and also check the link
www.voip-info.org. My asterisk server configuration
follows all the rules correctly.sometimes xlite send
their RTP packets to destination without servers
interaction but sometimes it sill goes
Hi-
I have an upcoming order for a bunch of asterisk boxes, and I'm considering
using an assembled package for the server, instead of building them from
components as I usually do.
Does anyone have experience with the Dell PowerEdge 750 server, or any other
1U rackmount server for use with
Hello!
I can connect to asterisk with kphone (sip) with no problem.
This is my extentions.conf
[kphone]
exten = 123,1,Answer()
exten = 123,2,Playback(vm-goodbye)
exten = 123,3,Hangup()
So when i call 123 i expect to hear the Playback sound.
However, when i dial 123 i get this at the astersisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
debian sid
kernel 2.6.7
cpu: AMD Duron(tm) Processor
kernel.log:
Aug 23 17:33:40 weblogin kernel: Zapata Telephony Interface Registered on major 196
Aug 23 17:33:40 weblogin kernel: zaphfc: no version for zt_receive found: kernel
tainted.
Aug 23
Hello all,
Anyone have any experience using the Motorola SBV4200 cable modem with *.
At my first try the CM was complaining about Incorrect Version.
The problem was that the chan_mgcp in * is version MGCP 1.0 and the
endpoint expects MGCP 1.0 NCS 1.0.
Now from what I understand this NCS 1.0 is a
[EMAIL PROTECTED] wrote:
Geoff Nordli wrote:
I am trying to compile the meetme2 application with the latest CVS
head and it fails. Here is the error message that I get. Can
someone point me in the right direction?
[snipped lengthy error message]
Here's a patch. You can apply it by hand,
Hi,
maybe I oversee somth. very obvious, but I'm a little puzzled about the
following 'error':
When I make a call from the PBX to * I get number not available, but on
debug I see, that asterisk is searching just for the first digit in the
extension, which of course doesn't exist, eg:
I dial
Hello,
since yesterday 16:55 (MET) I only got some mails from this list but
not very much, only about 10 mails since then ... is there something
wrong with the mailing list or do I just have a problem, please respond
off-list, as I guess I might not see it else. ;)
kind regards
bo
Thanks for the explanation, I'm checking with my boss now to see if we
do have hunt groups set up. For now we have 2 companies to service
with our PBX and 8 lines, so we could probably split it 4/4 or 5/3.
On Mon, 23 Aug 2004 09:21:11 -0700, Chris Shaw [EMAIL PROTECTED] wrote:
- Original
Hi Guys,
Im having some problems with a Wildcard TE410P card.. During a call I get
some strange messages and the voice drops out:
Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Write
returned -1 (Resource temporarily unavailable) on channel 1
Aug 24 16:40:17 DEBUG[1101416512]:
Thank you -- funny thing is, I had the same bookmarked, but it just
seemed too expensive for the application -- for $300, I can stick a
cheap IP phone in a hole in the wall :) I think it's time to get a
Budgetone.
-Original Message-
From: Chris Shaw [mailto:[EMAIL PROTECTED]
Sent:
spectro wrote:
- You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581
(rport), and will not reply to requests that contain it. Using
'nat=never' in sip.conf disables *'s support for this rfc. Uniden has
acknowledged the issue (DR#60).
Are you running RC1 or RC2?. We are running a
Hey All,
Just wondering if there is a version of the G729 Codec available for Mac OSX? I can see almost
all the x86 infrastructures ...
Regards
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to
yes :)
Shawn Parker wrote:
i know asterisk itself will install on a linux kernel 2.6.x, but i've
seen places say that the zaptel drivers wont? is this still true? is
it possible to build asterisk/zaptel on a linux 2.6.x kernel?
--
-
Appradius Project: RADIUS authentication and
This one's classic:
http://www.grandstream.com/user_manuals/budgetone100.pdf
Bottom of page 30:
Auto Answer: Default is NO. When set to Yes, any incoming call will be
automatically answered via speakerphone after a Beep. This is somewhat
similar to Intercom, but still different.
Thanks for
test message. No list messages received today.
Gary
- Original Message -
From: Soren Rathje [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 5:08 PM
Subject: Re: [Asterisk-Users] Asterisk MIBS
Alagalah wrote:
Hi,
I was wondering if there are any Asterisk
is it possible to build asterisk/zaptel on a linux 2.6.x kernel?
Yes.
___
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To UNSUBSCRIBE or update options visit:
Sorry not to directly answer your questions, but I would recommend
http://sourceforge.net/projects/chan-sccp/
I do not know the number of people using skinny vs sccp2, but I believe for
are using sccp2. Maybe others can comment here.
-Original Message-
From: [EMAIL PROTECTED]
Hello
All,
We are looking for aSIP
providerteminating calls inIndia, Pakistan and
Bengladesh.
Any one knows a good
one?
Regards,
Cesar
___
Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
To
I have no experience with Quintum but since it is SIP registration, you
might want to look at this:
http://voip-info.org/tiki-index.php?page=Asterisk%20sip%20md5secret
I was unable to get X-Lite and SJPhone (both softphones) to register with *
until 'after' I switched to using md5secret.
Hello,
I have noticed a little problem in chan_mgcp.so.
After a few unsuccessful attempts to call an endpoint using
MGCP/aaln/[EMAIL PROTECTED] I have noticed the following on the system running
asterisk using netstat.
udp50524 0 XX.XX.85.XX:2427 0.0.0.0:* 2666/asterisk
Hi all,
I am hoping that someone has experienced a similar problem
to mine. I have a X100P connected to a Nortel Meridan-1 PBX system via an
analog extension. When the extension is called and then hung up the X100P
refuses to disconnect the call. I have to reload Asterisk to free up
Hi Folks,
I found some old postings about Sangoma card support in *
but nothing indicative if this is supported or not for dialin/dialout.
I found only support indication for VOFR using Sangoma...
Anybody other driver available for Sangoma even not free like
chan_dialogic ?
Thanks,
Isamar
On Mon, 23 Aug 2004 18:20:58 -0500, Erik Anderson [EMAIL PROTECTED] wrote:
Hello all - I'm just starting to play around w/ asterisk, and I've run
into a seemingly simple problem that has really manged to frustrate
me...
I'm running the latest cvs version of *, and am trying to dial in to
I implemented successfully with guidance from this document
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
However, I had to make a small change to the sip.conf sample files:
From:
exten = _1XXX,1,Dial(IAX/asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED])
and
exten =
Does asterisk support using an H.323 provider for outgoing calls? From
everything I have found, it looks like it does. However, I have had no
success in getting it to work. I would really appreciate if somebody
could give me a hand. I am using the channel that comes with asterisk.
I have
Shawn Parker wrote:
i know asterisk itself will install on a linux kernel 2.6.x, but i've
seen places say that the zaptel drivers wont? is this still true? is
it possible to build asterisk/zaptel on a linux 2.6.x kernel?
# uname -a
Linux asterisk2 2.6.7 #1 Tue Aug 3 10:52:26 CET 2004 i686
Will you post to the list?
-Matt
On Fri, 20 Aug 2004 15:32:35 -0500, John Baker [EMAIL PROTECTED] wrote:
Still waiting on Polycom for something. Will make it available as soon
as I get it.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Quoting [EMAIL PROTECTED]:
From: Kanuri, Seshu [EMAIL PROTECTED]
Dave,
I am implementing this solution and would appreciate if you can send
me the doc at this email address - [EMAIL PROTECTED]
Thanks
Seshu Kanuri
Enough people have asked me for this that I will try and condense it for
I'm trying to connect my Asterisk server via sip using my vonage soft
phone account. Has any anyone successfully got to work? I get error from
asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing
'/etc/asterisk/sip.conf': Found
Aug 24 11:01:11 WARNING[1125329600]: acl.c:146
David Cook wrote:
snip
3. Asterisk as a SIP server behind nat, clients on the outside
connecting to Asterisk
snip
Then it goes on to say:
* #3 Works with port forwarding and some header mangling magic
Can somebody explain a little more about the header mangling magic as
it is not discussed
Has anyone gotten CID from Bell Canada to work properly with *?
We have our * box down at our datacentre in St Louis, and whenever we
call it from a Bell Canada Telephone line, all we see is '' for the CID.
I did some digging on google and the mailing lists and couldn't find
anything pertaining
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 23 August 2004 03:14 pm, Shawn Parker wrote:
i know asterisk itself will install on a linux kernel 2.6.x, but i've
seen places say that the zaptel drivers wont? is this still true? is
it possible to build asterisk/zaptel on a linux
On Mon, 2004-08-23 at 19:25, Scott Stingel wrote:
Hi-
I have an upcoming order for a bunch of asterisk boxes, and I'm considering
using an assembled package for the server, instead of building them from
components as I usually do.
Nothing against your ability, but buying a 1u server is much
On Tue, 2004-08-24 at 05:17, [EMAIL PROTECTED] wrote:
Hello!
I can connect to asterisk with kphone (sip) with no problem.
This is my extentions.conf
Inband DTMF is not supported on codec G.711 u-law. Use RFC2833
Inband DTMF is not supported on codec G.711 u-law. Use RFC2833
Inband DTMF
All,
Has anyone ever seen a problem where the autoattend detects the first digit twice?
What I am seeing is this:
My extensions are 421-468.
When a caller calls in and dials exten 433 from the autoattendant, they get
exten 443. This is happen for any extension that is valid in the 44x range
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
Extension logic looks good:
exten = 6153248305/_931NXXX,1,Queue(queue1);
exten = 6153248305/_615NXXX,1,Queue(queue2);
;exten = 6153248305,1,Queue(queue3);
show dialplan looks good:
Hello Rod,
Have you checked out the Wiki. There's lots of information in there:
http://www.voip-info.org/wiki-Asterisk
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roderick A.
Anderson
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 24 August 2004 01:34 am, david kwok wrote:
I have been bugging by a telemarketer who does not take any cue at all.
So I look up the Asterisk Handbook and send his call with the respect
caller id to my voicemail.
Has any one
Check the wiki for dtmfmode. It is explained here:
http://voip-info.org/tiki-index.php?page=Asterisk%20sip%20dtmfmode
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson
Sent: Monday, August 23, 2004 7:21 PM
To: [EMAIL PROTECTED]
Subject:
Hi Darren,
Ok, asterisk's H.323 channels works fine. Do you know why you get
disconnect from (or can't connect to) your provider? There are some
debug available (h.323 debug command).
Probably, if you must use G.729 or G.723, you should know you need to
buy licenses for this codecs. If you
Hello I get this warning all the time when I am
using iax2 for inbound calls or outbound.
Aug 24 13:48:41 WARNING[-1105474640]:
chan_iax2.c:4873 socket_read: Error: Resource temporarily
unavailable
I get the calls and the sound is fine. But the
screen on the cli is full of these warnings
On Mon, 23 Aug 2004, Erik Anderson wrote:
Hello all - I'm just starting to play around w/ asterisk, and I've run
into a seemingly simple problem that has really manged to frustrate
me...
I'm running the latest cvs version of *, and am trying to dial in to
the default extention 1000 demo
I have a test box setup and I can make outbound
calls on the PSTN thru the diguim card, however I can not make a sip user to sip
user call by dialing the extensions. I am getting the following
error.
-- Called cisco7960 -- Got
SIP response 482 "Loop Detected" back from 208.218.14.123 == No
- Original Message -
From: Erik Anderson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 24, 2004 9:03 AM
Subject: [Asterisk-Users] Re: newb question regarding DTMF
On Mon, 23 Aug 2004 18:20:58 -0500, Erik Anderson [EMAIL PROTECTED]
wrote:
Hello all - I'm just starting
On Mon, 2004-08-23 at 18:12, Zineddin Karzazi wrote:
Hi,
I'm a student and my thesis work consist in testing
Asterisk with Swyx(SwyxWare).
My approach is to declare asterisk as h323 gateway for
the Swyxserver using oh323 Plugin.
Is there any possibility to connect Asterisk with
Swyx?
Try using kewlstart. signaling=fxs_ks in zapata.conf and
signaling=fxsks in zaptel.conf.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
MarkSent: Tuesday, August 24, 2004 3:02 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] R X100P
connected to Meridan-1
On Tuesday 24 August 2004 10:07, Matt G wrote:
Has anyone gotten CID from Bell Canada to work properly with *?
Yup, works perfectly fine with X100P/X101P as well as CAC1 and Adit600 FXO.
And naturally for PRI, too. :-)
We have our * box down at our datacentre in St Louis, and whenever we
Don't you need to Answer before Playback ?
[voicepulse-incoming]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,2,Playback(demo-congrats)
exten = h,1,Hangup
exten = i,1,Hangup
exten = t,1,Hangup
Marcelo
Em Seg 23 Ago 2004 20:20, [EMAIL PROTECTED] escreveu:
Hey guys,
Cal someone help me.
On Tue, 2004-08-24 at 08:44, Paul Concepcion wrote:
Thanks for the explanation, I'm checking with my boss now to see if we
do have hunt groups set up. For now we have 2 companies to service
with our PBX and 8 lines, so we could probably split it 4/4 or 5/3.
If you split it, you will want to
Guys I am having some serious issues with my call queue and Management
is breathing down my neck pretty bad, and I am running out of ideas.
I have a single queue for my tech support department. I originally was
using the AgentCallbackLogin for them and it tested out great on our
testing
Hi all,
I am hoping that someone has experienced a similar problem
to mine. I have a X100P connected to a Nortel Meridan-1 PBX system via an
analog extension. When the extension is called and then hung up the X100P
refuses to disconnect the call. I have to reload Asterisk to free up the
Scott Stingel wrote:
Hi-
I have an upcoming order for a bunch of asterisk boxes, and I'm considering
using an assembled package for the server, instead of building them from
components as I usually do.
Does anyone have experience with the Dell PowerEdge 750 server, or any other
1U rackmount server
Hi Geoff,
Geoff Nordli wrote:
I was able to compile the module and it loads correctly, but I am still
having problems with the app.
I see all the users in the conference, but I can't kick them out, or change
their mode from talk to listen-and-talk. No errors are showing up anywhere.
I am not
Scott Stingel wrote:
Hi-
I have an upcoming order for a bunch of asterisk boxes, and I'm considering
using an assembled package for the server, instead of building them from
components as I usually do.
Does anyone have experience with the Dell PowerEdge 750 server, or any other
1U rackmount server
Hi all,
I'm trying to make a paper/presentation for astricon with a lot of graphs
and performance statistics for asterisk.
I'll try to handle:
- differences between the 2.6 kernel and the 2.4 series
- differences in hardware, ranging from slow embedded pc's to SMP setups.
- difference between
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Stingel
Sent: Monday, August 23, 2004 8:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dell PowerEdge 750 rackmount
Hi-
I have an upcoming order for a bunch of asterisk boxes,
Roderick A. Anderson wrote:
Just a little pun there!
I've been mostly lurking for a couple of weeks and realize how little
I know and understand about this PBX and phone stuff. I did a
little looking about and came across a glossary but they terms are --
for me -- kind of out of
On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
[EMAIL PROTECTED] wrote:
x-lite uses the RFC2833 style for DTMF out of the box (it can be set to
transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
for your x-lite user.
That's what I've read, and I have added
Check out my ATA idea though, with a regular cheap analog doorphone and a
HTX86 or even Sipura, you can program the ATA to dial an extension as soon
as the button on the intercom is pressed and then with some extension logic
you can do neat things... You can get a doorphone anywhere even radio
I had the telnet window to the phone open by chance and noticed this line
twice when I tried to call the IP10:
WARNING: may need to undo rtp port operation here
The warning line appeared immediately when I picked up the handset.
I have no idea what this means. I also tried calling the phone
Some asterisk drivers compile but don't work with 2.6.8, like wcusb, and
ztd-eth. I have TDM400P and X100P working fine.
But, there might be somebody else with other kernel versions where it might be
working fine.
Marcelo
Em Seg 23 Ago 2004 23:31, Lubomir Christov escreveu:
yes :)
Shawn
Is anyone interested in the RPMs I have built for Mandrake 10 (kernel 2.6) of a recent
CVS dump?
I have source RPMs for zaptel and asterisk which should work with any 2.6 kernel
version of Mandrake (including Cooker). I haven't done anything with PRI so that isn't
built. I also have binaries
Hi,
I have Asterisk running with the VoiceMail. Using the latest CVS. I have
my extensions.conf setup so that if a SIP caller dials *99 the
VoicemailMain() as follows:
exten = *99,1,Wait(1)
exten = *99,2,VoicemailMain()
A couple days ago I installed the MySQL/Voicemail
Richard Cook wrote:
Hello Rod,
Have you checked out the Wiki. There's lots of information in there:
http://www.voip-info.org/wiki-Asterisk
Obviously not enough. I found it and looked around a bit a few days ago
but never really got into it. With a slap of a clue-stick and a
I'm in Canada and experiencing the same thing - X101P.
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt G
Sent: Tuesday, August 24, 2004 10:07 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
I'm running Asterisk on Slackware 10.0 with linux kernel 2.6.7 and it runs
just fine. Had a few problems with install and several /dev paths being put
in the wrong place, but it runs fine. No performance boost though. Not sure
if there's really any reason to run Asterisk on linux 2.6 over 2.4
Just to share my solution with the rest of the list... You can't
upgrade from a really old version straight to 7.x. You have to upgrade
to a newer (but not 7.x) version, then upgrade to 7.x.
All the details on this exact problem happen to be in the VoIP
Telephony with Asterisk book by Paul
Hi Kurt!
maybe I oversee somth. very obvious, but I'm a little puzzled about the
following 'error':
When I make a call from the PBX to * I get number not available, but on
debug I see, that asterisk is searching just for the first digit in the
extension, which of course doesn't exist, eg:
I
spectro wrote:
- You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581
(rport), and will not reply to requests that contain it. Using
'nat=never' in sip.conf disables *'s support for this rfc. Uniden has
acknowledged the issue (DR#60).
Are you running RC1 or RC2?. We are running a
On Tue, 24 Aug 2004 [EMAIL PROTECTED] wrote:
Has anyone ever seen a problem where the autoattend detects the first
digit twice?
Yes, I have seen the same behaviour. In my case it seems to be less than
perfect dtmf tones. They are detected, but sometimes a very short break in
the middle of a
I got this problem fixed by putting the following in the sip.conf file.
dtmfmode=inband
Bill Dunn
- Original Message -
From: Bill
To: [EMAIL PROTECTED]
Sent: Tuesday, August 24, 2004 12:49 PM
Subject: Voicemail Couldn't read username error
Hi,
I have
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 24 August 2004 01:02 pm, Steven Critchfield wrote:
The big thing to look into is what PCI busses the machine supports. We
where very surprised with our Dell when it came with a PCIX slot and a
66mhz 64bit slot. The included ethernet card
Steven Critchfield [EMAIL PROTECTED] wrote:
buying a 1u server is much better than building it as there are
a lot of cooling problems to overcome
You're right about that - I learned a lot about 1U cooling and low-profile
fans in the last one I built. It was fun, but now they want 10 boxes,
On Tue, 24 Aug 2004, James Sizemore wrote:
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
Extension logic looks good:
exten = 6153248305/_931NXXX,1,Queue(queue1);
exten = 6153248305/_615NXXX,1,Queue(queue2);
;exten =
On Tue, 2004-08-24 at 12:03, Steven Critchfield wrote:
Inband DTMF is not supported, Use RFC2833
Go search your kphone configs and fic it to use some out of band
signalling of DTMF.
kphone does not support RFC2833 DTMF, only inband DTMF.
--
Eric Wieling * BTEL Consulting *
Ok what the flaming hell is up with the MailMan? It's taking over a day to
send posts now, before it was at most a couple hours...
Is digium doing maintenance on that server or something??
-Chris
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On Tue, 24 Aug 2004, Michael Welter wrote:
Scott Stingel wrote:
I have an upcoming order for a bunch of asterisk boxes, and I'm considering
using an assembled package for the server, instead of building them from
components as I usually do.
Does anyone have experience with the Dell
Maybe it's just me, but it looks as if you have one too many X's in
your pattern matching..
615NXX is all you need, i see 615NXXX. Same for 931.
-twisted
James Sizemore wrote:
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
When I woke up this morning * was giving me the message broken pipe
when I try to log into the CLI... Thats all I know.
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I do too, TDM400P and X100P work ok.
But TDMoE with 2.6 freezes one of my computers, causes non-stop stack faults
on another, working only on a third.
And wcusb doesn't work on any at all.
Marcelo Pacheco
Em Ter 24 Ago 2004 13:36, Steve Szmidt escreveu:
-BEGIN PGP SIGNED MESSAGE-
Hi, I've been messing with getting SIP working for days now, with
limited success. I've got Asterisk set up on a remote server with the
echo test. Please try it out to verify I've got the server working
right:
sip:[EMAIL PROTECTED]
Running FC1, ThinkPad T22, headset thru the soundcard.
I've used DL320's in the past with great success - and greatly
sympathise with the Dell availibility issues! ;-)
The one thing I found with the DL320's is that they didn't support
booting without a keyboard and our only solution was to KVM them all
(which we didn't want to do). This was back
On Tue, 2004-08-24 at 10:07, Matt G wrote:
Has anyone gotten CID from Bell Canada to work properly with *?
We have our * box down at our datacentre in St Louis, and whenever we
call it from a Bell Canada Telephone line, all we see is '' for the CID.
I did some digging on google and the
Hi everyone,
I have been trying to get call pickup to work and am having success
with group pickup by setting the callgroup and pickupgroup in the
zapata.conf and sip.conf files. However, I cannot get directed call
pickup to work.
According to the little documentation
We use the Monitor() command to record all incoming calls to our call center. After
about 100 incoming calls, the Monitor() command starts to hang, as follows:
- a call comes in
- Asterisk starts recording, the -in.wav and -out.wav files are created
- the partys talk
- after a while (between 1
On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said:
I have been searching thru all docs that I can find on wiki and such but
can
not get an answer. I am trying to collect a date from user input in the
form of digits dialed from the phone to use in an agi script to do a
database
Hi,
I'm a student and my thesis work consist in
testing
Asterisk with Swyx(SwyxWare).
My approach is to declare asterisk as h323
gateway
for
the Swyxserver using oh323 Plugin.
Is there any possibility to connect Asterisk
with
Swyx? how?
the outgoing call
On Tue, 2004-08-24 at 14:08, Bill wrote:
I got this problem fixed by putting the following in the sip.conf file.
dtmfmode=inband
That only works if your call is using the ulaw or alaw codecs.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the
Were looking at implementing Asterisk in our
department in the near future, were looking at anywhere from 15-25
extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. Ive heard bad
things about running Asterisk on SMP machines? Would
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