Thanks Chris and Mike for the great ideas.Richard"Chris A. Icide" [EMAIL PROTECTED] wrote: Or you could skip the overhead associated with an AGI and use thedialplan command availabe after installing asterisk-addons MYSQL.exten = _X.,1,Read(PO-NUMBER,enter-yr-po-num)exten = _X.,2,MYS
,Richard
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thers especially those who have built conferencing servers recently.Thank you very much.richard ___
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Thank you very much. Will go ahead and build the system. Hope everything goes smoothly.richardBJ Weschke [EMAIL PROTECTED] wrote: On 2/22/06, Richard OSS <[EMAIL PROTECTED]>wrote: Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference
fine. --- Richard OSS <[EMAIL PROTECTED]>wrote: Hello, Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardware We are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a
I have never requested that. not sure what to make of it actually other than they do bundle it sometimes (most likely on purpose).
I do know that the part number for the injector itself is D0023-0031-00-00
Richard
On 2/20/06, mustardman29 [EMAIL PROTECTED] wrote:
Kinda like what Cisco does
Carleton University to use one of their facilities on weekends (free parking) for group meetings.richardVirTERM [EMAIL PROTECTED] wrote: You can use Sprint (Group Telecom) and/or Magma. Keep us posted about the group meetings.. Thanks,Wojtek- Original Message - From: Richard
One thing to keep in mind with PoE is that you can simply use an injector at the phone location. At least with the 480i you can easily order the phone with the power injector.
Richard
On 2/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Mon, 20 Feb 2006, Soner Tari wrote: I have Data Sheet
.Can anybody recommend an alternative server that works well with TE205P and RHEL ES 4?This is our fi
rst time
using Asterisk so we would like to have it pain free as much as possible.Thank you very much.richard___
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in the near future and likely using them as well.
Richard
On 2/18/06, Michael J. Liberatore [EMAIL PROTECTED] wrote:
Well the gxp-2000 has BLF, the polycom 501 does not correct?I had anastra 480i and it was prety bad, but I was going to test the 9133i for
an inexpensive phone to compete
Anybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server.One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer
Hello,I am new to the world of Asterisk but I am excited.Do you still meet first Saturday afternoon each month, usually at the Royal Oak on Wellington at Holland?richard___
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for us in the future.
--
Richard Perini Internet: [EMAIL
PROTECTED]
Corinthian Engineering Pty Ltd PHONE: +61 2 9552 5500
Sydney, AustraliaFAX: +61 2 9552 5549
Just a reminder, there is now a list dedicated to the 480i:
http://groups.google.com/group/Aastra-480i-Users
Come join in the fun ;-)
Also, remember that there is quite a bit of good info on the voip-info.org 480i page
Richard
On 2/12/06, Carlos Chavez [EMAIL PROTECTED] wrote:
Does anyone know
I want to say thanks to everyone for the help so far. I figured out a
way to modify some AAH code that worked for me (well sort of). The line
I modified is s,14 in macro-dialout-trunk. Then I just added a variable
and passed it from 9_outside.
I just have one last problem. This waits for an
AFIK, fax is supported and installed with with app_txfax app_rxfax
If this proves to be true why would you need the ATA?
RCS
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Parcina
Sent: Thursday, February 09, 2006 2:18 AM
To:
I am thinking of setting up a * system for a remote office in china. I
was going to use a tdm400p to setup a basic 3X8 system.
I will setup the system in the US and ship it over.
Does anyone know of any problems that I should watch out for.
Signaling, caller id, ..
Same thing here. I had this problem awhile ago and made this
workaround.
Going to another trunk does not work because they are answering and not
sending a error code. If you are using AAH code then this waits 10
seconds on your Voip then times out and goes to PSTN. You can modify
for your
, the picture is diferent.
The box I have been using is the Dell 1850 with SCSI RAID. May be overkill for most applications, but it was under $2,000 so it made sense to me.
Richard
On 2/9/06, Nora Lavelle [EMAIL PROTECTED] wrote:
Hi Dovid,Thank you for the book. I'm already reading it.I have a dell 650
- Original Message -
From:
Richard
Smith
To: Richard Smith
Sent: Wednesday, June 08, 2005 1:03
PM
Subject: Re: Sipura SPA 3000 logic
Thanks for the help guys!
- Original Message -
From:
Richard
Smith
To: Asterisk Users
allows for 16, somewhat more flexible ones.
The potential here is for software that can effectively utilise the
extra ones, will gain some benefit.
See http://www.techreport.com/reviews/2005q1/64-bits/index.x?pg=2 for
more information.
Regards,
Richard
to the * server (depending on dialplan) to handle?
So basically, the caller does not get charged until the appropriate
extension hanging of the * server answers.
Cheers,
Richard
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Doug,
Can you provide any information on how you deployed that card into your setup? If it works for you we could put up a page on voip-info.org
Richard
On 2/6/06, Doug Lytle [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote: On Mon, 6 Feb 2006, trixter aka Bret McDanel wrote:
Again, I know
Does anyone have any experience (good or bad) with ArtDio gateways?
I am having two problems, the configuration does not seem to be sticking
(part does, part does not) and it ignores * commands from the phone. I
checked and the phone is definitely sending the *.
Thanks for you help
[EMAIL
I use Asterisk and I live in Winnipeg. I use it at home and I have a
client install coming up this year.
++
Best regards,
-Richard Houston
-R.L.H. Consulting
-E-Mail [EMAIL PROTECTED]
-WWW http://www.rlhc.net
-Bloghttp://www.rlhc.net/blog
All,
Thanks for the help. Checking on and changing the route based on
dialstatus is the way to go.
Thanks,
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Richard
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causes the phone to
do the same once pressed.
This however this does not occur when the phone
registers direct with a VOIP provider.
I was wondering whether anybody else has come
across this kind of problem before.
Thanks for your anticipated help.
Cheers,
Richard
Here is a snippet from
Jan 2006, richard Coco wrote:
Hi Armin,
thx for your feedback, but what do you mean with
Did
you load the card with config for DID on that
port?
I have loaded the modules with:
modprobe capi
modprobe kernelcapi
modprobe divacapi
modprobe divas
and then loaded divactrl
that this is ok (it works without did)? Or
have i forgotten something?
thx in advance..
--- Armin Schindler [EMAIL PROTECTED] wrote:
On Mon, 16 Jan 2006, richard Coco wrote:
Hi all,
i have asterisk 1.0.9 with an Eicon Diva 4bri and
chan_capi-cm-0.6. I have 2 NTBAs (one with did and
one
Hi all,
i have asterisk 1.0.9 with an Eicon Diva 4bri and
chan_capi-cm-0.6. I have 2 NTBAs (one with did and one
without).
When using the one without did, i am able to place
outgoing and incoming calls. When i use the NTBAs with
did i have a layer 2 error.
Anyone an idea?
-- Executing
Hi folks,
Just a quick question. Does the GrandStream
GXP-2000 phone support presence (hints)?
Cheers,
Richard.
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..Laa..Laa..Laa...,
when the AMD guys came round with benchmarks of their current hardware...
Supermicro do not do Opteron (or Athlon64) systems.
Regards,
Richard
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on their home page I regard as an indication
of their commitment. The page listing their AMD boards :
http://www.supermicro.com/Aplus/motherboard/
is headed For OEM Customers, so I take it from that I cannot order one
from my local supplier.
Regards,
Richard
of analog phones that work with plantronic headset and their amplifiers?ThanksRichard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Hi,
Happy New Year to all of you!
I was wondering what would be the best recommended
entry level IP phone that
works well with * if buying say around 10
handsets.
Linksys spa-941 and the grandstreamgxp-2000look like good
phones but
I'm open to recommendations
Cheers,
Reggie
Hi,
we have interconnected Asterisk with a HiPath4000 V1.0
using a H.323 Trunk. You have to install the oh323
channel from [1]. On your HiPath4000 V1.0 or V2.0 you
need a HG3550 board for IP-Trunking.
If you have the version 3.0 then the HiPath supports
SIP-Trunking but i have not tested it yet.
Steve Kann wrote:
Richard Scobie wrote:
My SVN asterisk systems use the following topologies:
1) PolycomSIP - *1 -IAX- *2 - H323 Gateway
2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog
3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog
There's a few points in here so far:
1) the new
Using a TDM400P with an FXO module and an FXS module, and a zapata.conf
with echocancel=yes above both channel definitions, is echo cancelling
applied individually to each module when a call is made out to the PSTN?
Regards,
Richard
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Zap EC works?
Regards,
Richard
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correctly.
This presumably means that if I want to use IAX trunking effectively, I
have to enable the IAX JB on all Asterisks.
Thanks,
Richard
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I'd tell.
- Original Message -
From:
Richard
Smith
To: asterisk-users@lists.digium.com
Sent: Monday, December 05, 2005
01:44
Subject: [Asterisk-Users] DISA
function
Hi all,
I was wondering whether the DISA function
Hi all,
i run into problems using park calling with chan_capi.
My setup looks like this
[200X]--[Asterisk]--[PSTN]
For internal calls [1] and for incoming call from
PSTN[2] every thing works fine. Unfortunately when a
sip extension (say 2007) makes an outgoing call to
PSTN and 2007 tranfers to
Several times a day I get this happening when I try to dial out. Is
there something on your side limiting concurrent calls or is it in my
config
Thanks for any help,
*
Called voipjet_out/XX
-- Call accepted by 64.34.45.100
Hello Michael,
It was a big pain for my provider as well. I ended up having to burn a
local DID for each toll-free DID to differentiate.
You can ask them to display the toll free number in RDNIS. My provider
wanted stupid dollars to do it.
--
Richard Cook
[EMAIL PROTECTED]
T: 705-223-2000
, it disconnects.
Cheers,
Richard.
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Hi all,
i'm wondering if anyone has ever managed to get moh
working on Siemens OptiPoint400?
if yes, can you please explain how you did it...
thx.
__
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Larry Alkoff wrote:
I've just heard about DECT which is used for about 50 million phones in
Europe and is just starting to appear in the US.
DECT stands for Digitally Enhanced Cordless Telephone
and supposedly has much greater range than other cordless telephony.
Additionally, you can purchase
being looked at by the ooh323 developers and a
couple of patches were submitted to asterisk-addons cvs today.
You could try checking this out from CVS, but as I believe they are
using the cvs head version of Asterisk, you may need to using this also
in order to help them debug.
Regards,
Richard
the vigor,
but have the same problem.
Has anybody managed to get a similar setup running?
any ideas, suggestions are wellcome... thx in
advance...
richard.
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Hi Alessio
[SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone]
I tried a similar setup some times ago and it was
working, have you
put the private ip address of the asterisk box in
the vigor setup ?
Can you ping the private address of the vigor from
the asterisk box
Alessio, Sergio
So an upgrade is of course necessary.
i have upgraded the vigor. Bad news... i am not able
to register the draytek anymore. But using a XLite on
my pc behind the Vigor works now fine (no one way
audio).
however i have an other question. I saw you put for
the bindaddr same
On Tuesday 22 November 2005 18:33, trixter aka Bret McDanel wrote:
I saw your second post and agree with it :) However had you not posted
this I wouldnt have noticed the dup between 'Antartica' and 'Antartic
Terrortories' which is now corrected, as is an Austria entry, some
Brazil fixes, UK
will be added somewhere, and then the fraud can start...
Richard.
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On Tuesday 22 November 2005 22:39, trixter aka Bret McDanel wrote:
On Tue, 2005-11-22 at 21:03 +0100, Richard Bennett wrote:
What would be really usefull would be a collaborative block-list
containing all the premium destinations and and special tarrif lines
worldwide. I already have
Thank you very much. Your response was very helpful.
--- Ken Godee [EMAIL PROTECTED] wrote:
Does anyone know where I can buy a 50ft crossover
cable to connect my
digium card -- I believe it's a T100P -- to my
Adit 600. The one I have
now works fine but I need a longer one.
Why not search the wiki first?
Actually, I did search the wiki first and was not able
to get the wires to line up inside the RJ-45
connector. So I am looking to buy. Also, if you read
at the bottom one person said that the pinout there
did not work for him.
Does anyone know where I can buy a 50ft crossover cable to connect my digium card -- I believe it's a T100P -- to my Adit 600. The one I have now works fine but I need a longer one.Thanks,Richard
Yahoo! FareChase - Search multiple travel sites in one click
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Richard Watson wrote:
[888120]
type=friend
username=888120
mailbox=888120
canreinvite=no
nat=yes
secret=secret
host=dynamic
qualify=yes
context=sipdemo
subscribecontext=sipdemo
Just for fun I had a play yesterday using SER as a stateless
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Richard Watson wrote:
Richard Watson wrote:
[888120]
type=friend
username=888120
mailbox=888120
canreinvite=no
nat=yes
secret=secret
host=dynamic
qualify=yes
context=sipdemo
subscribecontext=sipdemo
Just for fun I had a play yesterday using
Do I need licenses to use the codec in New Zealand?
smime.p7s
Description: S/MIME Cryptographic Signature
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. This is
however very basic stuff.
Jan
Richard Malcolm-Smith wrote:
Do I need licenses to use the codec in New Zealand?
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trixter aka Bret McDanel wrote:
If you know where to look there is another option out there that doesnt
use either method, but I have doubts about how legal that one is, so I
will not comment on that.
Can someone give me some pointers for this?
smime.p7s
Description: S/MIME Cryptographic
Michael Crown wrote:
Does the phone ocasionally prompt the user for a password? -Mike
Yes it does
How did you know?
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Does anyone know how to put an Aastra PT 390 in
headset mode, so it will only give a dial tone when
you are ready ? Right now I can't figure how to keep
it hung up? If I hit googbye it merely flashes (give
me a dial tone again).
Any help would be greatly appreciated?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The VoIP Connection wrote:
There is a setting on the Advanced page called Challenge Response on
Phone. Turn this setting to Off and your problem will be solved. Also, we
usually set the Proposed Expiry to 1 minute On the SIP page when phones
are
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael Crown wrote:
Did you change the proposed expiry? -Mike
Yes, now set to 1 minute.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
the frequency with which asterisk sends keepalives?
3) Does SER handle this better - would placing this outside the NAT help
handle connections from inside?
4) Do newer versions of asterisk handle this better?
5) Any other suggestions?
TIA.
- --
Richard Watson
-BEGIN PGP SIGNATURE-
Version: GnuPG
--
Thank-you,
Richard James Blundell II
Internet Engineer
Telepacket, Inc.
27455 Tierra Alta Way, Suite A. Temecula, CA 92590
Direct:(714)263-9090 Mobile:(951)757-5899 Fax:(714)263-9001
E-Mail: [EMAIL PROTECTED]
Please note that:
This email message may contain confidential and privileged information
to the calls.
In a previous thread on this same error, Steve Kaan asked if there were
any DNS lookups involved. I can't see it unless there are reverse
lookups being done.
Both boxes are 600km apart on a good 10Mb connection.
Regards,
Richard
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.
Is this correct? I do not believe having these echo parameters in
sip.conf will achieve anything.
They should be at the top of zapata.conf.
Regards,
Richard
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Hi all,
i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1
2.6.9-22.0.1.EL) using latest package from
sourceforge (chan_capi-cm-0.6.tar.gz).
I have installed divactrl_2.1.tar.gz and untared
protocols_all.tar.bz2 in /usr/share/eicon.
---
Sorry guys, I forgot to add;
It works fine if set-up in the features.conf file,
but not when the dedicated transfer button on the phone is used.
I know it uses the refer method for this, but this
just creates zombies according to the *console
output
Sorry guys, I forgot to add; It works fine if
set-up in the features.conf file, but not when the dedicated transfer button on
the phone is used.I know it uses the refer method for this, but this
just creates zombies according to the * console
output
Hi Everyone,
I have a problem withasterisk-at-home beta 4.
Whenever I do an attended transfer (softphone or IP phone),once the 2
parties have finished talking the asterisk switch reboots with the following
error;
usr/sbin/safe_asterisk:line 42:14265 aborted
${astsbindir}/asterisk
Thank you Hadley, that was the problem.
Cheers,
Richard
- Original Message -
From: Hadley Rich [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, October 28, 2005 12:20 AM
Subject: Re: [Asterisk-Users] Not saving
[EMAIL PROTECTED]
1.2.0 beta4 writes to the respective voicemail directory and when the call is
hung-up the .wav file disappears.
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I am looking for a little help. Last night I brought my first * box
into production use. The configuration is as follows:
PRI/VoIPJet --- * --- Nortel
Everything is working except I want to make all local numbers and 911 go
out over the PRI line.
I thought adding these lines to fromnortel
I am looking for a little help. The configuration is as follows:
PRI/VoIPJet --- * --- Nortel
Everything is working except I want to make all 911 and local numbers go
out over the PRI line.
I thought adding these lines to fromnortel would work but it does not.
The call still tries to go out
Does anyone have a sample on how to do a
supervised transfer via the Manager API.
Incoming Zap - SIP - xfer -
Zap
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000
x2010
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A great stance. Another contributor most likely lost. Nice job.
--
R
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, October 20, 2005 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Thursday, October 20, 2005 1:05 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Re: Why Asterisk documentation is so poor...
Richard,
I'm sorry you and others feel the way you do
also offer filtering the other way - attenuating incoming
spikes and RF noise.
Regards,
Richard
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Is there no way to have asterisk determine its IP either via upnp or else
resolve a dyndns hostname rather then having an entry in the config file?
smime.p7s
Description: S/MIME Cryptographic Signature
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[EMAIL PROTECTED] wrote:
Put in your zapata.conf for the channel:
busydetect=yes
busypattern=1500,500
busycount=4
callprogress=no
Steve, is this a better solution than the COMPARE_TONE_AND_SILENCE
busydetect option that can be enabled in the Makefile?
Regards,
Richard
Hello,
Is anyone using FreeTDS version 0.63 with *?
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000
x2010
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http
I thought maybe someone was using 0.63 with code they
developed themselves. Where do you find 0.62.x?
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000
x2010
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy
KuoSent: Monday, October 03, 2005 7:16 PMTo:
[EMAIL PROTECTED
Hi Jacky,
thx for the feedback
rich.
--- Jacky [EMAIL PROTECTED] wrote:
Hi, Richard,
I still try, but fail with rtp transfer.
2005/9/27, richard Coco [EMAIL PROTECTED]:
I still find out how to let LCS 2005 accept SIP
invite from Asterisk,
Need more help.
Hi jacky
I still find out how to let LCS 2005 accept SIP
invite from Asterisk,
Need more help.
Hi jacky,
can you please share your experience and explain how
to let LCS accept SIP invite from Asterisk.
I deseperate trying to place a call from asterisk to
LCS. (calling from Asterisk to LCS using
to place a call from lcs to *.
thx in advance...
--- richard Coco [EMAIL PROTECTED] wrote:
Hi,
i have the same setup too.
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where
Of possible interest to people having various issues with TDM400 cards,
is that a fix has just been submitted to CVS for the issue where CPU
usage would regularly spike up to 100% with the wctdm driver loaded.
Regards,
Richard
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Hi,
i have the same setup too.
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where to configure this.
thx.
--- Jacky [EMAIL PROTECTED] wrote:
LCS 2005 just support SIP TCP or
On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote:
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead
releases fail with a combination of checksum and ss_thread errors?
I'm concerned when beta2 or the 1.2 release comes out it will not
work.
I have been through the configs I
are the
tricks?
Can macros have goto labels?
Thanks for any help,
--Richard Cook[EMAIL PROTECTED]T: 705-497-9320
x2010
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Hi,
The only thing i know is that you need a netbootserver
using five special files. So, if possible, ask Siemens
for the optipoint 600 netboot upgrade procedure. AFAIK
it is a known problem...
hope it helps...
--- Anthony Cox [EMAIL PROTECTED] wrote:
Not strictly a problem with Asterisk but
Are there any USB type devices(ex: analog modem) that can be used to
connect a local telephone company line to the Asterick PC? I've seen
some cheap cards on Ebay but they use a PCI connection. A USB
connection would help with packaging if there was one.
Thanks in advance.
Richard Davis
[EMAIL
Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, but attached to issue 3854 you will find patches you may be able
to apply to the current CVS-Head to acheive this.
Regards,
Richard
185
ACPI: PCI Interrupt Link [APCL] enabled at IRQ 21
ACPI: PCI interrupt :00:02.1[B] - GSI 21 (level, low) - IRQ 193
ACPI: PCI Interrupt Link [APCJ] enabled at IRQ 20
ACPI: PCI interrupt :00:04.0[A] - GSI 20 (level, low) - IRQ 201
Regards,
Richard
Michiel van Baak wrote:
Is there any other solution like this out there that works
with asterisk ?
If you find something, I would be interested in the outcome.
I want something for the house here, at the moment I just have 2 analog dect
bases plugged into the same line, but you cant roam
David,
Yes we got them and they caused huge problems. The echo training would
cause the line to mute and you would hear something like a dtmf tone briefly
and then you would be connected and talking again. This might happen once
or 50 times during a call. I spoke to Digium and they say there
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