On Sun, 13 Feb 2005 22:36:45 +0100, Michiel van Baak [EMAIL PROTECTED] wrote:
On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote:
+++ Michael Devenijn [13/02/05 18:23 +0100]:
Actually I am using a supermicro board the P4SCI wonder if I can turn off
hyperthreading i dont think there is a bio
Rob at draughon.org writes
I recently obtained a Western Electric multi-line phone and am
seeking help with getting this beast working with *.
The interesting stuff in my * implementation consists of a T100P
card, a TDM400P card, and an Adtran TA750 channel bank with three
On Fri, 11 Feb 2005, Peer Oliver Schmidt wrote:
Remco Barende wrote:
I'm currently using a HFC-S card for my ISDN BRI line with bristuff.
The instability is driving me crazy however.
[..]
I have three different locations with HFC cards. I had the same stability
problems on ALL of the
Good day all
I want to know with version of spandsp works well with ether asterisk
1.0.3 or 1.0.5
Thanks
Altus
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Alberto Zuin wrote:
Yes, but I have to configure a route for each host in every host! A the
moment i have about 120 Asterisk hosts and every astersk have about
50-100 users! Is for that I want a single sip proxy that route dial.
I read more about ser, and the suggestion is to use ser for
On Sat, 2005-02-12 at 12:20 +, JunkMail wrote:
For the single card I was using with isdntool for initialization,
wich
works fine but has no support for two cards.
Can anyone tell me exactly how to initialize the ISDN system manually
???
It all starts with modprobe -v hisax
Hi,
some people report good success with the zaphfc cards, others, incl.
myself have mixed results.
I am using the debian stock kernel 2.4.27 with mixed results. Anyone
care to tell what kernel(s) you on successful zaphfc integrations?
Thanks.
--
Best regards
Peer Oliver Schmidt
PGP Key ID:
On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote:
We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN
port. Only downside is that only 1 call can be using 729 at a time. This has
been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to
On Sun, Feb 13, 2005 at 10:39:36AM -0800, Luki wrote:
The Sipuras have a ton of configurable parameters. If you understand
them (and there is no good manual, unfortunately) then you can be of
great benefit. Otherwise they'll be worthless. I particularly miss the
dial-plan, distinctive ring and
Hi,
I have * working with X-Lite and Sipura adapters, but I have one person
who is linux based, and is trying to use Linphone and Kphone. His end
works, but I get very bad echo on my end. Have any of you folks been
able to get linux based soft phones working well with *?
I'd appreciate links
On Mon, February 14, 2005 22:22, Darren Ellis said:
I'd appreciate links to howtos/docs if you have them, and/or samples of
working configs for * and the linux softphones.
I gave up trying to use linux soft clients they all seem to have some
fatal flaws or issues I could never fully get rid
Darren Ellis wrote:
Hi,
I have * working with X-Lite and Sipura adapters, but I have one
person who is linux based, and is trying to use Linphone and Kphone.
His end works, but I get very bad echo on my end. Have any of you
folks been able to get linux based soft phones working well with *?
I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13)
and noticed a setting marked 'paging' under supplementary services on the
Phone settings page on the advanced admin login. Anyone know how it might
be used? Could it be like the Snom -
exten =
hi
i am looking for some info for speech recognition for example when someone
call to my house asterisk ask for who hi want to call and he say the name
david or susan (wife) or daniela etc...
thanks
David
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David D. Faerman wrote:
hi
i am looking for some info for speech recognition for example when someone
call to my house asterisk ask for who hi want to call and he say the name
david or susan (wife) or daniela etc...
And the wife asks Who's Daniela? ;-)
--
_/_/_/_/ _/ _/
_/_/
On Mon, 2005-02-14 at 12:22, Darren Ellis wrote:
Hi,
I have * working with X-Lite and Sipura adapters, but I have one person
who is linux based, and is trying to use Linphone and Kphone. His end
works, but I get very bad echo on my end. Have any of you folks been
able to get linux
Dear friends,
I need to make a software for a listen service. A room with 6 persons, 6
lines and 6 extensions. When a people (client) call for this room
(external calls), depending of number, asterisk access a data base
searching for that number and forwarding (propably whith a PABX) to a
Hi all,
I have setup two X-Lite phones and an Asterisk box. They are all on the
same LAN and have private IP addresses assigned to them. I am able to
place a call from either phone but the moment it is picked up (trying to
be answered), it goes dead - as in no sound!
I get the error, Unknown RTP
Am Montag 14 Februar 2005 12:57 schrieb Tor Setane:
On Mon, 2005-02-14 at 12:22, Darren Ellis wrote:
Hi,
I have * working with X-Lite and Sipura adapters, but I have one person
who is linux based, and is trying to use Linphone and Kphone. His end
works, but I get very bad echo on my
(Intentional top-post, due to relative brevity of answer)
The error is a typo in the latest chan_sip.c in Stable. See my note on
Mantis bug #3557 (softins).
To fix, find line 3673 and change ast_isphonenumber(l) to !ast_isphonenumber(l)
CVS HEAD does not have the typo, so is OK.
Cheers
Tony
On February 14, 2005 01:18 am, Matt Gibson wrote:
It can receive calls both when receiving power, and when not receiving
power. It can make calls only when not receiving power from the wall. I
tried unplugging it for a good 10-15 minutes to make sure
it was off for sufficient time, but still
Hi all,
I have setup two X-Lite phones and an Asterisk box. They are all on the
same LAN and have private IP addresses assigned to them. I am able to
place a call from either phone but the moment it is picked up (trying to
be answered), it goes dead - as in no sound!
I get two errors, Unknown
On Mon, Feb 14, Craig Guy wrote:
I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13)
and noticed a setting marked 'paging' under supplementary services on the
Phone settings page on the advanced admin login. Anyone know how it might
be used? Could it be like the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
David D. Faerman wrote:
| hi i am looking for some info for speech recognition for example
| when someone call to my house asterisk ask for who hi want to call
| and he say the name david or susan (wife) or daniela etc...
|
Why not the easy
Hi,
I would like to use Realtime extentions with a four bri card, the
classic quodbri.
Normally with that card I would use * bristuffed from Klaus-Peter
Junghanns, but since that package is based on stable version there is
no Realtime at all in it (I suppose).
Did you knoww if someone has done
Thanks Mark
I am definitely interested in the budgetone 102 but am a little concerned
about the 10mbit only Ethernet ports !! From what I have read, these are
relatively new models and I like the addition of a second port to daisy
chain your PC from the same network connection, however why
Thibault Lamy wrote:
some people report good success with the zaphfc cards, others, incl.
myself have mixed results.
I am using the debian stock kernel 2.4.27 with mixed results.
We are using 2.6.10 self-built kernel on debian unstable
zapfhc works fine, we are able to send/receive calls
and
Hi there
Just a general question, has anybody experienced any problems with any
Digium telephony cards in the UK, specifically with BT (British Telecom)
lines. I just want to make sure there are no compatibility issues before
purchasing cards, (mainly TDM400P's)
Any comments would be greatly
Good day all
Anyone doing asterisk in New-Zealand?
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Altus Snyman wrote:
Good day all
Anyone doing asterisk in New-Zealand?
But of course!
The Daily Asterisk News is run out of New Zealand!
We are also local distributor for Digium gear.
We provide all of the support for products also.
Let us know if you have any questions etc.
--
Cheers,
Matt
In the vain of asterisk in new-zealand...
Anyone know of a reliable source of digium gear in singapore? Also
where to pick up IP phones, anyone any clues?
Ta
Jonathan
signature.asc
Description: This is a digitally signed message part
___
I can get you a good deal if you import the from South-Africa..Let me
know.Altus
On Mon, 2005-02-14 at 15:38, Jonathan Gill wrote:
In the vain of asterisk in new-zealand...
Anyone know of a reliable source of digium gear in singapore? Also
where to pick up IP phones, anyone any clues?
Ta
Hello,
I wanna configure Asterisk to work with iptel.org proxy. I have
already created an account in iptel.org; what steps should I do?. I
want to test the configurations using X-Lite and some help to
configure it out could be nice too.
Thx
--
-DdC
I am not much into speech recognition, but I know that a major company
only had success when they simplified the menus so as to only ask
simple yes/no-questions in this manner:
Do you have problems with your internet connection?
(yes = Do you have a black modem?)
(no = Do you have problems
Hi Altus
What sort of price are you able to get? Im only looking for prob 2
(cheap) ip phones right now, maybe more later if all goes well... And as
this is personal stuff, im on a tight budget.
Ta
Jonathan
On Mon, 2005-02-14 at 15:40 +0200, Altus Snyman wrote:
I can get you a good deal if
On Monday 14 February 2005 13:00, Brett, Gary wrote:
Thanks Mark
I am definitely interested in the budgetone 102 but am a little concerned
about the 10mbit only Ethernet ports !! From what I have read, these are
relatively new models and I like the addition of a second port to daisy
chain
Hi,
There are several people on the UK mailing list (I am one) that have
purchased the TDM400P FXO and are having problems with disconnect.
Basically the cards are great (sound quality etc) but give some issues with
detecting a UK remote hang-up. Mainly an issue within IVR, MeetMe and VM.
There
On Mon, 14 Feb 2005 14:11:15 +
Bob Goddard [EMAIL PROTECTED] wrote:
On Monday 14 February 2005 13:00, Brett, Gary wrote:
Thanks Mark
I am definitely interested in the budgetone 102 but am a
little concerned
about the 10mbit only Ethernet ports !! From what I have
read, these are
relatively
On February 14, 2005 09:23 am, Robert Webb wrote:
On Mon, 14 Feb 2005 14:11:15 +
Bob Goddard [EMAIL PROTECTED] wrote:
On Monday 14 February 2005 13:00, Brett, Gary wrote:
Thanks Mark
I am definitely interested in the budgetone 102 but am a
little concerned
about the 10mbit only
[EMAIL PROTECTED] wrote:
Folks,
I recently obtained a Western Electric multi-line phone and am seeking
help with getting this beast working with *.
The interesting stuff in my * implementation consists of a T100P
card, a TDM400P card, and an Adtran TA750 channel bank with three
On Mon, 2005-02-14 at 22:29 +1100, Duane wrote:
I gave up trying to use linux soft clients they all seem to have some
fatal flaws or issues I could never fully get rid of
While I'd second that, Gnomemeeting is still pretty good and by far the
best softphone I've used on Linux. Currently, it
And middle posting is almost as bad. :-)
But.. To the point...
If you would have read what you were replying to, you
would have noticed they did mention why weren't they
100Mbits connections on the 102 models for daisy
chaining
to a PC.
Robert
SNIP
Yes but failing to trim is even worse. :-)
-A.
Eric Wieling wrote:
joachim wrote:
Yes,
It's untested and unfinished and touches the core of asterisk. (maybe
causing massive amounts of deadlocks).
So? That's what CVS-HEAD is there for.
Adding in experimental patches willy-nilly, especially ones that have the
potential to cause huge
Brett, Gary wrote:
Hi there
Just a general question, has anybody experienced any problems with any Digium
telephony cards in the UK, specifically with BT (British Telecom) lines. I just
want to make sure there are no compatibility issues before purchasing cards,
(mainly TDM400P's)
Any comments
I'm having some problems getting meetme to work now that I have upgraded to
.5 I am able to conference calls but every time I try to manage the
conference through meetme it just says No users in this conference Any
ideas why it doesn't see the conference call?
Thanks for any help!
Jason
This
On Feb 13, 2005, at 4:43 PM, John Novack wrote:
I use JFAX which I think is also known as Efax.
If you are open to a new fax number anywhere else in the US from your
home Zip code, then it is free.
Otherwise there is a quarterly fee.
AFAIK, you can't port an existing number to them, but I
I wouldn't recommend the grandstreams, I had very bad experience using
the grandstream 102, It kep locking up on me. The buttons are very bad
buttons. The sound quality is just as bad.
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On Feb 13, 2005, at 7:50 PM, Rich Adamson wrote:
Can't offer any clue on the above either. Based on Steve Underwood's
comments earlier (relative to outbound fax now fails on the TDM when
it was working earlier), it would almost sound like a timing issue of
some sort that is associated with calls
I your case the problem is with the grandstream, the GS will not
display callerID correctly, take out the name from the callerid string
like this:
exten = ${EXTEN},PRI,SetCallerID(${CALLERIDNUM})
On Fri, 11 Feb 2005 23:46:13 -0800, Robert L Mathews
[EMAIL PROTECTED] wrote:
Nicol?s Gudi?o
On Feb 14, 2005, at 5:39 AM, Nicolas Bougues wrote:
On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote:
We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1
WAN
port. Only downside is that only 1 call can be using 729 at a time.
This has
been confirmed with Linksys.
nope, it uses an callinfo header:
http://lists.digium.com/pipermail/asterisk-users/2005-January/086462.html
On Mon, 14 Feb 2005 19:41:23 +0800, Craig Guy [EMAIL PROTECTED] wrote:
I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13)
and noticed a setting marked 'paging'
Greetings,
I have a problem making a call from Asterisk to Cisco H323 PSTN gateway
using H323 channel. I can call but there are no sound in both way. If I call
H323 gateway directly from SJPhone I have no problem with sound.
Any advice are welcome.
Thanks in advance.
Any more ideas on my below mail? If a user is registered with SER and
leaves a voicemail message with asterisk (by using rewritehostport
etc in ser.cfg), then how is the user supposed to listen to the
message afterwards? Is there any other way other than the MWI method??
Thnaksm
Aisling.
hi,
since my latest libpri update i get these messages:
!! Unable to handle ROSE operation 36
!! Unable to handle ROSE operation 30
i searched through ITU X.219 and X.229 but can't find any values for the
Remote Operations Service Elements.
are these AOC-E messages?
regards
frank
Hi there
Just a general question, has anybody experienced any problems
with any Digium telephony cards in the UK, specifically with
BT (British Telecom) lines. I just want to make sure there
are no compatibility issues before purchasing cards, (mainly
TDM400P's)
Any comments would
daniela is affear but shhh
- Original Message -
From: Bill Maidment [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 14, 2005 8:54 AM
Subject: Re: [Asterisk-Users] speech recognition
David D. Faerman
Mark Eissler wrote:
While eFax, and similar services, are some sort of a solution to at
least half the problem, I just think using these services is a kludge.
I don't agree.
Inbound faxes sent to my E-mail as TIFF are the best solution. No wasted
paper, ink or toner. It it needs to be printed
Hi,
I am implementing T.38, and finding a problem getting boxes that work
with T.38 for testing. A lot (maybe most) ATAs now claim to support
T.38, but I'm finding a lot of these lie. I have one box here that just
crashes when it hears a fax tone. :-)
I'm looking for boxes known to implement
Hi Jason,
The web meetme wont control a conference until someone dials in to it
(eg you cant have a web interface setup then wait for someone to dial in
afterwards).
If you are unable to use the amp extension based conference rooms set up
one of your own by editing the conf file and see if you
Does anyone know if this has been implemented? I have been around the sites and
haven't really found much. I know there was an old patch that would make it
work
but it doesn't do anything but break the application now.
.o---o.
Brian
Bob, Thanks for your reply, im not sure what top posting is, but I have been
on holiday and am simply replying to a response that was given to my
original question, If you could explain to me how I go about continuing the
thread it would be much appreciated, with regards to your reply, I am indeed
You have to add the include statement in the context thet you want the
parking (park, and pickup) to be available. # will only work with a t
(for the called), and/or a T (for the caller) in the dial command.
On Sun, 13 Feb 2005 00:28:30 -0500, Robert Webb [EMAIL PROTECTED] wrote:
I am trying to
Hi gentleman
I've configured SER to forward every call starting with sip uri request
1 to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it
call to my other SIP Provider outside my network, sending username and password
for authentication.
I've read at
Good day list,
I am feeling extra stupid this Monday morning and am hoping
someone can come to the rescue.
I am trying to use the ztmonitor utility on my wildfire 4 FXO
card. and have read the following from the wiki.
*Wiki start
If you set this to yes, use
Sorry issue solved.
I had to RTFM better I just needed to increase the gain
higher my magic number ended up being 15.5
Sorry to bug 8000 ppl.
~ron
-Original Message-
From: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Sent: Monday, February 14, 2005 11:18 AM
To:
Cisco and Asterisk are not behind firewall.
Where can I check for settings noH245Tuneling and noFastStart in Asterisk
H323?
-
-- Executing Dial(SIP/msn-069a, H323/[EMAIL PROTECTED]:1720) in new stack
-- Called [EMAIL PROTECTED]:1720
-- H323/peer:1720 is making progress passing it to
Hi there,
The settings are in oh323.conf
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
Hi Gary,
Aren't those all tied to service providers now?
Regards,
Steve
Gary Carr wrote:
We use the PAP-2NA with fax machines and have not had any problems.
Gary
Hi,
I am implementing T.38, and finding a problem getting boxes that work
with T.38 for testing. A lot (maybe most) ATAs now claim to
The X101P works but I dont think it would be acceptable in a
commercial environment. The audio levels are too low and there is too
much echo (or speech break-up with the aggressive cancellation set
on).
Saying that hang-up detection works and CLID works with some source
code changes.
Anybody got
No, the PAP2's are. The PAP2-NA is for any provider.
Gary
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 14, 2005 11:33 AM
Subject: Re: [Asterisk-Users] ATA
Could anyone shed any light on how SER and/or Asterisk (stable branch)
has held up for them in that last while?
Are you using SER and/or * in a production environment? Do you ever
restart the software or reboot the system? How many users are
utilizing the system? How many calls per
On Sun, 13 Feb 2005, Malcolm Taylor wrote:
I'd be grateful if someone could point me in the right direction.
I have a Broadvoice trunk attached to Asterisk which I use for frequent
calls to the UK using the following in extensions.conf
exten = _0[1-68].,1,Ringing
exten =
noH245Tunneling instead of noH245Tuneling
typedef struct call_options {
charcid_num[80];
charcid_name[80];
int noFastStart;
int noH245Tunneling;
int noSilenceSuppression;
unsigned
Quoting Gary Carr [EMAIL PROTECTED]:
You might want to tell that to these guys:
http://www.voipsupply.com/product_info.php?products_id=317
regards,
Paul
No, the PAP2's are. The PAP2-NA is for any provider.
Gary
- Original Message -
From: Steve Underwood [EMAIL
Maxemail.com is out there too. $14.95/yr if you don't care about the
number, or $6/month if you do. Not a bad deal for the service.
Outbound is still the most difficult, but there are print-fax drivers
out there. Packetel has (or used to have) a $4/month option as well,
iirc
-Original
Hi,
is there someone who speaks in Italian?
I'll try to explain in english my problem, but if there is someone who speaks
italian i think it would be better for me.
I'd like to use asterisk only as IVR and call diverting. I have only one
phone line, and no other phones, all the calls arrive at
No, I am using H323 driver
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Liu
Sent: Monday, February 14, 2005 11:36 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk-H323
Hi
On Mon, 2005-02-14 at 13:08 +0100, Jens Kübler wrote:
Maybe you wanna check out the softphone zip4x5 made by Zultys.
It's the software which is used by the same hardphone.
Howdy,
Do you use this product and do you have any relationship with Zultys?
It looks interesting, but it is documented
Dear Xu, my name is Marco Castillo, I'm in Guatemala, Central America, and I
have recently succesfully installed a TE110P here in Guatemala. There are
many implementations of a E1 or T1, but I think that the great majority can
be configured via the zaptel drivers. I will suggest you to buy a card
On Mon, 14 Feb 2005, Frank Sautter wrote:
since my latest libpri update i get these messages:
!! Unable to handle ROSE operation 36
!! Unable to handle ROSE operation 30
i searched through ITU X.219 and X.229 but can't find any values for the
Remote Operations Service Elements.
are
Hi all,
How can I configured H323 EPs or OH323 EPs to get them authenticated
through GNUGK???
Many thanks
Ben
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I have 3 of the Black ones. I think the are junk. They work, and I
actually found a manual online for it. I ran into a weird problem last
week. After I did a Reset to Factory. All the phones were getting th
same IP address from the DHCP server, I found that the MAC address on
the phones were
Darren Ellis wrote:
Hi,
I have * working with X-Lite and Sipura adapters, but I have one
person who is linux based, and is trying to use Linphone and Kphone.
His end works, but I get very bad echo on my end. Have any of you
folks been able to get linux based soft phones working well with *?
I have a question
for using gastman. I have set up extensions for my IAX users as
IAX2/username, and I keep getting the following
Dunno how to tell if
IAX2/username/6 is IAX2/username
I was wondering if
there is some sort of wildcard character that can be used here? The number
changes
That site is correct. You have to be authorized by Linksys to order the
product from a distributor but they will work with any VoIP service. We use
them with our * service.
Gary
Quoting Gary Carr [EMAIL PROTECTED]:
You might want to tell that to these guys:
Hi!
http://www.broadbandphone.com.au/global/pnp.htm
they are called a Kitty Ethernet Phone, seem to be available in 3 or 4
models but with identical Guts.
The only info I have found on them is Gateway Technologies, supposedly
the Chinese manufacturer website...
If the message is only sent as an email attachment
(delete=yes,attach=yes) then
the user must listen to it by playing the attached wav file on their pc.
If the message is saved on the Asterisk server then you need to provide
dial-in access to Asterisk that sends the caller to VoiceMailMain.
If this has been covered before - I appologize.
We use some Sipura SPA-2000's with the g711 codec and all seems fine
(except for the occasional failure to register errors in my asterisk
logs - but I will save that for another post).
g711 call quality is on par with our Cisco 7960's. However,
Our SER/Asterisk implementation is extremely stable if you define
stable as the ability to deliver a set of features without either
application
crashing. We are a production environment with 75 users total. Asterisk is
only used for voicemail. The only issue we have is that the audio
(greeting or
I noticed a problem this morning with our cdr logging. We have a cron
job that places a call file into the spool directory having asterisk
call itself to check to make sure its still handling incoming calls
correctly, then queries the CDR database in mysql and makes sure that
appropriate
I really appreciate your reply.
For Asterisk, are you using G729 as your codec, or something more
high-bandwidth (ulaw)?
Is there any definition of stable that you would use that would point
to SER and Asterisk not being stable?
Again, thanks for your reply.
--
Dana
On Mon, 14 Feb 2005
Could you help me with this problem? When I call H323 gateway there is no
sound in both ways.
Here is h323 debug:
- begin
-- Executing Dial(SIP/msn-6297, H323/[EMAIL PROTECTED]:1720) in new
stack
Allowed Codecs:
Table:
G.729A{sw} 1
G.729{sw} 2
On Mon, 14 Feb 2005 15:13:37 -, Patrick Lidstone (Personal E-mail) wrote:
Hi there
Just a general question, has anybody experienced any problems
with any Digium telephony cards in the UK, specifically with BT
(British Telecom) lines. I just want to make sure there are no
compatibility
Many thanks Greg!
Sometimes things are just too obvious!
Malcolm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: Monday, February 14, 2005 11:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Mon, 2005-02-14 at 10:47 -0700, Kyle Hagan wrote:
I used to use kphone and have very bad echo, I switched to sjphone and
it worked great.
It isn't too bad, but it has latency (compare it e.g. to asterisk as
softphone and you'll see what I mean) and no dial pad. So I found it
isn't really
So I've got it installed and running (?) except for one error message
and I haven't had time research it yet but I'd like to get a quick reply
or pointer to my next step to getting [EMAIL PROTECTED] working.
The error is during boot ( Linux ) and comes from ztcfg ( I think?
Memory going
On Mon, 14 Feb 2005 20:01:18 +0100, Bruno Hertz [EMAIL PROTECTED] wrote:
Another point to note is that seemingly all closed source softphones
(SJ, XLite beta and also cornfed) make connections to web servers
and transmit platform/call information. Don't know how you think about
that, but for
I applied the florz patch but my problems remain. Now I get all sorts
of weird errors on the console and I cannot make outgoing calls. Incoming
calls work as expected. I am using a single HFC-S card with BRI.
Any clue what these errors below are?
Ri = 44651 TEI msg = 3 TEI = 7f
Ri = 3800 TEI
Yea, I might be doing native bridging. The peer might do jitter buffering
(as its Asterisk), or they might have it turned off for whatever reason.
Also, my clients have significantly more jitter issues (Guatemala ISPs
suck), so it's possible that I might want a different jitterbuffer setup
than my
I've been fighting this for a while and have come back to the list with
some of my configuration information. I have a quicknet internet linejack
card and have been thus far unsuccessful at placing outbound calls over
the analog phone line. I can receive calls through the line jack and
route
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