On Tue, Oct 14, 2003 at 08:37:29AM +0930, [EMAIL PROTECTED] wrote:
I am not having any luck placing out going calls
I dial the number 08 82420173 ( our outside line )
Hi Mick,
calling your outside line would be engaged anyway, if you are using it to call out on,
no?
Call your mobile
--- Uriel Carrasquilla [EMAIL PROTECTED] wrote:
John:
are you aware of any documentation on how to configre SER to be a
front-end
to Asterisk?
I suspect it is very inexpensive to put a SER server in a hosting
facility
I think the cost is about the same as for putting a web server
at a
Hello all,
Can I execute linux command like(ls, mkdir) through the Manager interface?
I can't seem to access the manual at digium.com. I keep getting 'Forbidden'
error. Looks like they are upgrading or something.
CF
___
Asterisk-Users mailing list
Chris Albertson wrote:
This is the big problem with using Asterisk for SIP. With Asterisk
the audio data between two SIP extensions has to actualy go into
then out of the Asterisk box. This does not scale well to
thousands of users like in a university campus or a comercial
SIP service.
Thanks for all the help!
I have to postpone my asterisk ambitions as a more pressing task has
just shown up, but I'll be back in a month or two ;)
cheers!
--
Mit freundlichen Gren
Conrad Braun
Pentaprise GmbH
Im Pinderpark 5
D-90513 Zirndorf
http://www.pentaprise.de
OOppsss, I think I found the problem, I have in my dial the options
'tTm', and it seems that the m-option sends to the caller a
music-on-hold while waiting for the other party to anwer. I just removed
it ( in my test equipment ) and it give me a ring, will check it
tomorrow with the snom's
Hi Scott,
thanks for your help. The frame errors which you got i don't got.
Yes this is a slow machine, but i greenly thought that * can handle about 30
calls on it without
problems
Have you ever run 120 simultaneously calls on one machine ?
What does you mean with How many instances are you
Woody
It may have been early but it was not that early.
I was not calling the same number that I was dialling from.
This system is not in production so it is on a spare line
And I was calling our main number
So there is n issue there somewhere ???
Regards Mick
-Original
Echo has nothing to do with TCP vs. UDP. It's an analog phenomenon that
occurs where the hybrid is, where the four-wire circuit changes to a
two-wire circuit.
Anyway I have developed a way to accellerate the adaption of the echo
canceller but it's not made its way into zaptel yet. I'm in
When trying to dial out
982420173 our main number
I get the engaged signal before I finish entering the phone number
Any ideas
Regards Mick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I am still having trouble changing the sound files.
I can take a wave file out of another program and set it in the folder
and it will work
If I record a wave file in Windoze
No go
Am I missing some thing ???
Thanks for the help
Regards Mick
Hello,
Does anyone know how to set the outgoing CallerID properly when using
Snom200/SIP/CAPI/BRI?
Following doesn´t work:
exten = _0.,1,SetCallerID,526910
exten = _0.,2,Dial,CAPI/526980:${EXTEN:1}
Asterisk writes:
*CLI -- Executing SetCallerID(SIP/226-ada0, 526910) in new stack
--
Hi all,
Just a little note for the records and archives. We see many small
glitches / troubles in the mailing-list but rarely success stories ...
Here's one :
Asterisk is running perfectly fine in our setup :
Debian 3.0 stable / Athlon 1.8, 256 MB Ram / Digium E-100P / Swisscom
PRI isdn
We
Hi all:
This is a warning that appears when *
loads.
WARNING[1074494336]: File app_voicemail2.c, Line
2889 (load_module): SQL init
VoiceMail2 tries to bring up VM SQL
service?
Regards,
Gus
It looks like you are registering fine. If
you dial 12321 from another phone, does it not ring?
This is the transaction as I see it in the log that
you attached:
Phone: REGISTER
Asterisk: Proxy Authentication Required (Send me
your credentials)
Phone: REGISTER with CREDENTIALS
Asterisk:
-- Original Message --
From: James Sharp [EMAIL PROTECTED]
On Mon, 13 Oct 2003 14:01:00 -0400, Andrew Kohlsmith wrote:
I really wouldn't like to run a telecom system on Windoze in the first
place..
Last place I worked, we had to reboot our phone system
I have 2 PBX linked together with IAX using the GSM codec. This link is
over a T1
that is shared with other traffic. I know that it is problematic using
ethernet to control
QOS so I would like to hear some practical solutions from other users.
___
Hi Thomas-
Yes, I can run 120 channels receiving IVR calls on one machine. The calls I
get are short, so this is sort of a worst case IVR setup. I forgot to
mention that I use Raw PCM (G.711 uLaw) for best quality - I think this may
have a lower load factor than using ADPCM or one of the other
My main question lies in the interworking between iptel's SER and Asteriks.
Not only on the configuration side, but also on the network side (here I
mean: can both run on the same server, or do they need to have different IP
addresses, ...).
My 10 cents:
Make sure that you run the two SIP
you can also do an insmod -N wcfxo (the -N checks only the numeric part
the of the module and not the extra stuff)
David J Carter wrote:
Thanks Rich,
I am re-installing the base SuSE Linux system again and will try to install
everything without doing any updates. I can't remember any
Hi all!
Please I need someone that have already done an H.323 - SIP gateway to help me
with some problems. I can stablish calls from a SIP telephone to a H.323, but I
can't do vice versa... (problems with port 1719- when the gatekeeper tries to
contact with asterisk at this port, it is
Dunca:
I am not sure I understand your statemnet.
SIP devices (UA) on the other side of the Internet behnid a NAT communicate
to * on the public Internet. Then this Asterisk connects to other Asterisks
(via IAX) that can be behind Firwalls (or NATS). am I understanding
correctly?
this is
Hi All
I receive thatwarning message:
WARNING[49159]: File chan_sip.c, Line 2220
(__transmit_response): Unable to determine sequence number from
''
What is it?
There is some documentation with all error
messages?
thanks
miklos
Hi Marcel,
Great news. Thanks for posting your success story
john brown
chagres technologies, inc
On Tue, Oct 14, 2003 at 01:02:17PM +0200, Marcel Prisi wrote:
Hi all,
Just a little note for the records and archives. We see many small
glitches / troubles in the mailing-list but rarely
It's an application and not a cli command, put it in extensions.conf
[default]
exten = s,1,System(ls /tmp/log)
regards
Martin
On Tue, 14 Oct 2003, Chee Foong wrote:
Hello mate,
I tried that, i get No such command 'System(ls)'. I can't even make it work
on CLI.
I am able to execute linux
Thanks
Chris,
I have now
re installed Linux, (Red Hat this time) and all seems to be working well. I can
receive calls and make calls via the FXO card.
My next
area is to try and talk to some Multitech boxes via H323.
Thanks all
for your help, I will no doubt be back asking
Where does the 4 wire change to a 2 wire?
On Tue, 2003-10-14 at 03:30, Mark Spencer wrote:
Echo has nothing to do with TCP vs. UDP. It's an analog phenomenon that
occurs where the hybrid is, where the four-wire circuit changes to a
two-wire circuit.
--
Sample configs and more:
On Tuesday, October 14, 2003 8:32 AM, Eric Wieling
[SMTP:[EMAIL PROTECTED] wrote:
Where does the 4 wire change to a 2 wire?
4 to 2 wires happen anywhere the signal goes from digital to a cable
pair. First the digital signal is converted to an analog 2 wire for
transmit and from 2 wire analog
Hi Martin!
here is:
s=Tue, 14 Oct 2003 17:55:00 GMT, sip:[EMAIL PROTECTED];expires=3600
Expires: 159
Content-Length: 0
9 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.microcity.com.br SIP/2.0
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From:
-- Original Message --
From: Steven Critchfield [EMAIL PROTECTED]
I understand I used to do the same. Sometimes up to 2 times a week.
But Asterisk has problems with Zombie lines. And guess what you have
to do to get them unlocked?
If you are getting
Where does the 4 wire change to a 2 wire?
with just a x100p that would be at the telco in their switching gear
where they do the a/d to route the call
but if you have a channel bank withn T1 to * then you have the a/d at the
telco switch
and then again more a/d when coming off channel banks fxo
What you want sounds pretty doable, with the exception of distinctive
ringing detection - reading the list recently I think someone had a patch to
allow caller ID to be received if the incoming call was a distinctive
ringing call but I don't think Asterisk differentiates or distinguishes
between
Hi,
I've just read the postings regarding the interworking between * and SER.
As these persons seem quite knowledgeable on this, I would like to have
their advise on my planned installation:
- I have broadband cable access
- I plan to install a SIP-aware router
- I plan to install a Linux server
I think the way to run SER and * on the same box is to have
Asterisk listen to SIP only on the loopback interface 127.0.0.1
that way no SIP clients will ever connect to Asterisk.
Configure SER to use all the interfaces. SER will connect to
SIP clients over the external interface and will notice
Does Anyone have a breakdown on what each option means in manager.conf
system,call,log,verbose,command,agent,user
I want to make a user who does not get a ton of events in
the socket and is just for sending a query and getting that 1 reply
I dont want to keep restarting my pbx to figure it
Hello,
I have seen the Mitel 5055 SIP phone mentioned a few times on the list, does
anyone have any wonderful or horrible things to say about it? We are
thinking about using them because they have many more programmable buttons
than the Snom200 phones and are about $70 cheaper.
Thanks,
MATT---
I don't see any warnings in your trace.
regards
Martin
On Tue, 14 Oct 2003, listas iPfone wrote:
Hi Martin!
here is:
s=Tue, 14 Oct 2003 17:55:00 GMT, sip:[EMAIL PROTECTED];expires=3600
Expires: 159
Content-Length: 0
9 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
I have asterisk and SER running on the same machine perfectly. As far as
I am concerned the best way to do it is to have two ip addresses for the
same ethernet interface. That way you can bind asterisk to one IP
address and SER to the other. This way you don't have to use
non-standard ports for
I am trying to figure out the correct syntax for the CLI command SIP SHOW CHANNELS.
I have tried
SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such
as:
-- Zap/15-1 is ringing
-- Zap/15-1 answered SIP/206-4299
asterisk*CLI sip show channel
I'd like to find out more about it too.. Are the buttons programmable as
line appearances and speed dials too? Does anyone have one working as a
standalone SIP phone with no Mitel PBX/other hardware in the mix?
I've worked extensively on SX2000's and have to say I love those systems
(anyone need
Use tabulator button for asterisk to help you guess the name.
regards
Martin
On Tue, 14 Oct 2003, Walker Haddock wrote:
I am trying to figure out the correct syntax for the CLI command SIP SHOW
CHANNELS. I have tried
SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is
On Tuesday 14 October 2003 13:24, Anthony Minessale wrote:
Does Anyone have a breakdown on what each option means in
manager.conf
system,call,log,verbose,command,agent,user
I want to make a user who does not get a ton of events in
the socket and is just for sending a query and getting that
Can someone please point me to a Cisco reseller who can re-certify a
7960 an put it under a maintenance agreement?
Paul Mahler
[EMAIL PROTECTED]
phone: 650-207-9855
fax: 877-408-0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel
Batista
Sent:
From -
Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by
sccrmxc11.comcast.net (sccrmxc11) with ESMTP id
20031014185753s1100nos46e; Tue, 14 Oct 2003 18:57:53 +
Received: from rwcrmhc12.comcast.net (localhost[127.0.0.1]) by
comcast.net
(rwcrmhc12) with ESMTP id
Walker:
sip show channel refers to a Call ID:
noc2pbx2*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
Format
172.16.254.620341522910 3607139911@ 00101/3 0ms ms ALAW
1 active SIP channel(s)
Then, you could see the details:
If you do make config in the zaptel then it's going to be loaded during
bootup. Otherwise it's not being loaded unless you do 'modprobe wcusb'
regards
Martin
On 14 Oct 2003, tom wrote:
From -
Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by
sccrmxc11.comcast.net (sccrmxc11)
On Tue, Oct 14, 2003 at 01:52:14PM -0500, Martin Pycko wrote:
Use tabulator button for asterisk to help you guess the name.
I have been trying that. I think that may have been implemented after I did the CSV.
I'm at:
Asterisk CVS-09/24/03-20:51:12 built by [EMAIL PROTECTED] on a i686 running
JanM wrote:
Hello,
Does anyone know how to set the outgoing CallerID properly when using
Snom200/SIP/CAPI/BRI?
Following doesn´t work:
exten = _0.,1,SetCallerID,526910
exten = _0.,2,Dial,CAPI/526980:${EXTEN:1}
Asterisk writes:
*CLI -- Executing SetCallerID(SIP/226-ada0, 526910)
On Tue, Oct 14, 2003 at 04:20:27PM -0300, CW_ASN - Gus wrote:
Walker:
sip show channel refers to a Call ID:
noc2pbx2*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
Format
172.16.254.620341522910 3607139911@ 00101/3 0ms ms
Is it common on this list to experience long posting delays?
Can someone Check something out for me here. I have a PBX behind an
asterisk system connnected via T1. The PBX is not seeing the caller ID or
ANI coming across from asterisk. I am setting it explicitly using :
extensions.conf fragment
[macro-dialswitch3]
; ARG1 Called Number, Arg2 Caller ID
I'm having trouble configuring these services the
way I want. Basically I prefer using iLBC before GSM, however Iaxtel only want
to talk GSM. It _seems_ that Voicepulse prefers GSM also, because even if I put
ILBC before GSM in the "allow=" part of iax.conf, if GSM is mentioned,
Voicepulse
I am working with one now. As far as I have gotten is that I can
authenticate against asterisk but then can't make a call. I will be
spending more time on this this week. Too many projects.
I don't know what price you're looking at but Mitel just dropped the
price of these phones to be pretty
Hi,
I've found that neither Michael Manousos patch nor ztdummy driver
do not fix musiconhold sound interruption problem up to acceptable quality
level. Sound is choppy here anyway.
It is my understanding (please correct me if I'm wrong) that if I have
a Digium card in my asterisk machine, these
Hi!
I have done that but it doesn't work because I need also the port 1720 to make
the comunication. Port 1719 is only used to the RAS messages and 1720 is used
to
make the communication.
Thanks a lot for your help
Regards,
Mireia
Quoting CW_ASN - Gus [EMAIL PROTECTED]:
Or, if you must use
With the musiconhold and SIP-SIP call it turnes out that you need to
disable silence supporesion on your phones/gateways since the timing is
taken from the coming stream (but only for musiconhold AFAIK)
regards
Martin
On Tue, 14 Oct 2003, Michael Ulitskiy wrote:
Hi,
I've found that neither
What steps would have to happen, in order to take an already-connected
call and move both parties into a conference room? i.e. do both parties
have to be parked first, or can one or both of them just be immediately
transferred to a MeetMe extension?
No. I also run machines with pure VoIP and there is not a single problem
with music on hold.
I don't think an X100P card will help. Anything you gain from the
ztdummy driver will be the same as what you can gain from an X100P, FWIW
the card is just a $10 winmodem.
-Original Message-
Ok folks I have another question. So far I have gotten my IAXTEL number and I have
been able to make calls from my asterisk system to any IAXTEL number and even to FWD
numbers. I also got FWD to work and I now can get calls to my main system. It's great
when these things work. But when I
You must use GSM with iaxtel and Voicepulse for now... I talked to the guy
from voicepulse and they said ilbc might be turned sometime in the future.
But not sure.
bkw
On Tue, 14 Oct 2003, Stig Hess wrote:
I'm having trouble configuring these services the way I want. Basically I
prefer using
Has anyone tested using SIP endpoints (Possibly the ATA-186)
with a connection that has at least 200ms, if not more, of latency? We are
trying to get some stuff setup in Australia and wanted
to know if this would be feasable, are there any added delays? Echos?
Hello all,
I've got a T100P connected to an Adtran TA750 with a T1 crossover...
This connects to a patch panel with phone ports. The Adtran is fully
populated with FXS cards.
All I get on any phone port is a fast clicking noise... No dialtone.
Asterisk 'sees' the card, (the channels show
Hi,
si posible on SIP phones to have the dial tone after 9 like on the FXS card?
I set ignorepat = 9 on my extensions.conf...
Best regards,
Chris HARIGA
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello,
While I've been searching for SIP hardphones and trying to pick between all
of the different features that are available for them I had the idea that we
(The Asterisk community) should create a label that hard-phone manufacturers
can put on their products that meet basic requirements to
Where did you get your 5055? I've tried to order it from 2 separate vendors
and they both tell me it will be at least a month before they get any in.
MATT---
-Original Message-
From: Barry Porch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 4:48 PM
To: [EMAIL PROTECTED]
Voicepulse _has_ ilbc turned on, but it will only
work if I disallow GSM. So I wondered if there was some way to turn on the
codecs for every connection...
Stig
- Original Message -
From:
Brian West
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 11:08
PM
Is your problem related to the settings under iax.conf? I have commented
out the whole of [iaxtel2] and left in [iaxtel].
I have a problem with my setup in that I have got it to register by
putting the register command in the iax.conf, but when I call my own
number I just get silence. Output
-
Hi,
I've found that neither Michael Manousos patch nor ztdummy driver do not
fix musiconhold sound interruption problem up to acceptable quality
level. Sound is choppy here anyway. It is my understanding (please
correct me if I'm wrong) that if I have
a Digium card in my
No I am using a Cisco 7940
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R.
Besch
Sent: Tuesday, 14 October 2003 11:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] dialling out
Mick,
If you're using the Grandstream,
Concept sounds good, not sure whether this was intentional or not but the
logo just says WiFi Certified if I don't give it more than a quick
glance. Something like this should be distinguishable without actually
reading it from all other similar certification or conformance marks. Eg:
how many
Hi,
my asterisk experiences with isdn cards supported by i4l
are not very good, but with avm a1 and capi everything
works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20,
german ISDN).
Now I want to connect a T1. Should I use an AVM T1-B
for approx 6000 EUR or is it ok to use one of Eicon's
Martin,
Thanks a lot.
The problem was a turned on silence suppression on cisco ata 186.
Now it seems to work perfectly.
Thanks to everybody else too.
Michael
On Tuesday 14 October 2003 05:04 pm, Martin Pycko wrote:
With the musiconhold and SIP-SIP call it turnes out that you need to
disable
Somebody keeps saying it is bad form to respond to one's own postings, but
I am going to do it here ...
Further experimentation and I discovered that this change to
iax.conf make the problem go away:
;
; Trust Caller*ID Coming from iaxtel.com
;
;[iaxtel]
;type=friend
;context=default
;auth=rsa
I have Asterisk up and running and it is working great with my SIP phones.
However, I have some Skinny-protocol Cisco 7960s. Does Asterisk support
the Skinny protocol? I've seen some references to Skinny in the software.
If so, should I stick with Skinny with the 7960 or convert to SIP? If
On Wed, 15 Oct 2003, Roger Schreiter wrote:
Now I want to connect a T1. Should I use an AVM T1-B
for approx 6000 EUR or is it ok to use one of Eicon's cheaper
Diva Server BRI S2M cards?
Why not use Digium hardware like the Wildcard T100P or Wildcard TE410P?
Regards,
Jac
--
Jac Kersing
I am having problems with early dialing and
chan_phone. In extensions.conf Ihave:
exten = _41.,1,Dial,IAX
If I dialvia a SIP or ZAP channels it works
fine.With chan_phone it start dialing right after the 3rd number.
If tried different combinations like (41., ... or
_41X., ) and still
Hi,
I would like to know if is possible to setup my Asterisk to detect if the
phone lines from FXO cards are in use. We use the parallel phones on the
same lines...
Best regards,
Chris HARIGA
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Uriel -
1) Please stop top-posting.
2) I'm afraid I don't have any data on specifics of creating a
front-end. I know how to do it, but my time these days is spent
writing lots of other projects that I have been doing. :-) I would
suggest you get SER and set it up - it's quite easy, and
I found the answer: One can disallow a codec within
each context of iax.conf.
Stig
- Original Message -
From:
Stig Hess
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 11:43
PM
Subject: Re: [Asterisk-Users] Iaxtel and
Voicepulse
Voicepulse _has_
Marcel Prisi a écrit :
I have some news ... after a bit of tweaking, the following seems to
work with Swisscom :
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
I still have a problem : Incoming calls from PRI work well, but
outgoing call don't :
I dial from a Grandstream 101
--
Jason Piterak wrote:
Hello all,
I've got a T100P connected to an Adtran TA750 with a T1 crossover...
This connects to a patch panel with phone ports. The Adtran is fully
populated with FXS cards.
All I get on any phone port is a fast clicking noise... No dialtone.
Asterisk 'sees' the
On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote:
I have to tell you, at the expense of offending you, that I use
MS-Outlook and the responses go to the tope of the messages. At work
I use Lotus Notes and the same thing happens. Before, I used PROFS
(on mainframes) and the same
Two comments:
1) just a $10 winmodem? is this literally true? then what
$10 card is known to work? Have you tried one? Yes it is pretty
clear that a Winmodem card could work for this application
2) I've always though a real DSP based modem card could be
re-programmed to a much
At 06:56 PM 10/14/2003, you wrote:
Two comments:
1) just a $10 winmodem? is this literally true? then what
$10 card is known to work? Have you tried one? Yes it is pretty
clear that a Winmodem card could work for this application
yes its literally true. The cards are out of production
Yes, of course. However, that would be a feature of the SIP phone, not
Asterisk, since Asterisk isn't providing the dialtone on your SIP phone,
the phone is doing that.
On Tue, 2003-10-14 at 16:28, Chris Hariga wrote:
Hi,
si posible on SIP phones to have the dial tone after 9 like on the FXS
For T-1 use a Digium card for about US$500
On Tue, 2003-10-14 at 17:02, Roger Schreiter wrote:
Hi,
my asterisk experiences with isdn cards supported by i4l
are not very good, but with avm a1 and capi everything
works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20,
german ISDN).
Now
Can anyone point me to some online documentation showing how to reset a
CP-7960 to factory default settings. I have some that are configured for
Callmanager and I want to get them back to generic default config. Any info
is appreciated.
Thanks
Cory Andrews
Is that why there is an X100P and an X101P? What design is the X101P
based on?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Tuesday, October 14, 2003 7:07 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
You shouldn't treat asterisk as a gatekeeper (because it ain't) On your
H.323 equipment, set asterisk up as a gateway.
Hi!
I have done that but it doesn't work because I need also the port 1720 to
make
the comunication. Port 1719 is only used to the RAS messages and 1720 is
used
to
make
Marcel:
Some switches with particular functionalities don't expect ton=unknown. For
example, Lucent 5ESS crap (in EuroISDN) doesn't work pretty well using
unknown.
In Siemens EWSD switch, if you have a divided PRI in two differents
Directory Numbers, you cannot send unknown.
In fact, send
Is that why there is an X100P and an X101P? What design is the X101P
based on?
AFAIK the current design uses a Tiger 320 chip which is essentially a PCI
gateway -- it provides a serial port and an 8-bit parallel interface to
anything. The single FXO card uses the serial interface, and the
-Original Message-
From: Paul Crick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 2:42 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Mitel 5055 phone
I'd like to find out more about it too.. Are the buttons programmable as
line appearances and speed dials too? Does
John:
I
don't use MSN so I can't comment. I do know that when my connections are
pure VoIP (no analog PSTN connections), the quality is better if enough
bandwidth is available.
TCP is
a protocol that gets used when you want to make sure a packet arrives at the
other end. UDP is better for
Andre:
This makes a lot of sense. I had used Asterisk in the past to play the role
of Gatekeeper for directing traffic to the appropriate Asterisk acting as a
PSTN gateway. IAX does a heck of a good job in that configuration.
However, with SIP, I have run into nothing but trouble with
2) I've always though a real DSP based modem card could be
re-programmed to a much more interesting use. Those on-board
TI DSP chips are quite powerfull computers. Easly enough to
compute any audio codec or even the front end for speech
understanding. they are flash
I would like to know if is possible to setup my Asterisk to detect if the
phone lines from FXO cards are in use. We use the parallel phones on the
same lines...
The current answer is to use line isolators -- Radio Shack sells these
things as privacy widgets. Although I am certain that the
On Tue, 14 Oct 2003 19:06:42 -0400, Jon Pounder [EMAIL PROTECTED] wrote:
At 06:56 PM 10/14/2003, you wrote:
Two comments:
1) just a $10 winmodem? is this literally true? then what
$10 card is known to work? Have you tried one? Yes it is pretty
clear that a Winmodem card could work for this
Steve:
Unless Asterisk is on the public side of the Internet, you will run into
problems if the UA (SIP phones) are behind a NAT.
In the scenario you presented, I think SER would be used for all calls
between SIP phones and they would only go to Asterisk when you need to
Gateway into the PSTN some
I think the X101P uses an Ambient/Intel HAM 56K modem chip. The older
X100 is based on Motorola. Any authoritative answers?
Andrew Kohlsmith wrote:
Is that why there is an X100P and an X101P? What design is the X101P
based on?
AFAIK the current design uses a Tiger 320 chip which is
Don't forget to reverse the FXO/FXS in the TA750. They are opposite to the
asterisk config files.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason Piterak
Sent: Tuesday, October 14, 2003 5:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
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