Thank you very much! But how could I integerate the VXML or "Voice XML"
into asterisk? Give me a hint please!
john
- Original Message -
From: "Chris Albertson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>
Sent: Thursday, March 13, 2003 8:49 AM
Subject: Re: [Asterisk-
- Original Message -
From: "Chris Albertson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>
Sent: Thursday, March 13, 2003 8:49 AM
Subject: Re: [Asterisk-Users] Build a complex IVR?
>
> What is "complex"? If all you need is a menus tree with static
> menus (that is,
As far as I know, there is not currently a way to do this. I've toyed
with the idea of adding support myself, but 1) its probably a bit over
my head and 2) I really don't have that much *spare* time anyway.
But its a feature I'd love to see!
Anyone want to collaborate on this?
-BAK
On Thu, 2003
> Just out of curiosity, early on in the discussion someone had suggested that
> upgrades to iax stream include version information. Did this wind up in the
> specification?
Yes, to the degree that there is a specification. As I've been working on
IAX I've also been working using xml2rfc on prepa
> I agree with the earlier post that said keep the name IAX.
I'm in this camp. It's short, sweet and meaningful. I even like the '2'. And
I just can't stand any more acronyms...
Just out of curiosity, early on in the discussion someone had suggested that
upgrades to iax stream include version inf
I don't think you understood my reply, I also don't use
Asterisk as a UA no calls are registered to Asterisk.
Asterisk just needs to be able reach the phone.
For this you need a "peer" statement for each phone.
That never takes calls. And a way for Asterisk to reach
the phone when it has a MWI me
On Thursday 13 March 2003 22:01, Mark Spencer wrote:
> Can somebody look at the RTP packets with "ethereal" and tell me if they
> notice any difference between what we send and what we receive? Perhaps
> we're starting out with values that are too high or something?
What i see here is that aste
Another suggestion - perhaps there would be a way to detect the status
of the PRI itself, perhaps monitoring and/or returning the performance
registers on the PRI interface.
Rationale: if the PRI is seeing lots of errors (or an LOS), then we
could fail-over to a physical line (FXO) or another PRI
I am trying to automatically put people into a MeetMe room when they
call each other. This way it is easy to add others as needed. I need
to know if the channel is busy first to know how to handle it. If it
is busy put them in the MeetMe room the called channel is in
automatically. If it
I'm trying to set up a call back for a few of our employees and encountering
an unexpected behavior.
When qcall tries to call when it picks up the line (POTS lines via ZHone
channel bank and quad t1 digium card) it detects
-- Qcall initiating call to ZAP/g2/x on Zap/1-1
(/var/spool/a
Yes, it is a zap channel. That would be great!
Wouldn't it make more sense if ZapIsAvail is true to continue at
priority+1? That way if it isn't available (busy) it works like the
Dial app and goes to priority+101.
Dave
On Thursday, March 13, 2003, at 01:59 PM, James Golovich wrote:
What t
I still do not think you understand my question...
I am using 'ser' for my SIP gatekeeper.
I am using asterisk for VOICEMAIL ONLY.
The only time incoming calls touch asterisk is when someone
does not answer their phone, and SER (the SIP gatekeeper)
redirects the caller to asterisk/port 5110 to t
According to Jim Archer:
>
> Well, the line has two pairs on it, on the red/green pair and the
> blk/yellow pair. I am not sure which pins those correspond to on the
> connector so I'm sure your right (it seems the inner pins are one pair and
> the outer pins another).
I once heard a phone wi
Hi,
is there a way to log errors to syslog ?
Currently logger.conf is logging everything in /var/log/asterisk/messages
but I would like to see the errors in my /dev/tty12 console on my server.
[logfiles]
console => notice,warning,error
messages => notice,warning,error
Thanks,
Jean-Pierre Denis
What type of channel are you trying to do this for? I was thinking of
writing an app ZapIsAvail, that would check if a certain zap channel is in
use and if so continue on with priority+1, otherwise jump to priority+101
James
On Thu, 13 Mar 2003 [EMAIL PROTECTED] wrote:
> In the extensions.conf
According to Ajit Kallingal:
> Hello All,
> Can the current Asterisk be integrated with mySQL to query a database ?
> I am looking at a typical IVR scenario where the user punches a "product
> code" and the database query will determine if the product is available or
> not. The reply would be numb
In the extensions.conf file, is there a way to detect if it is on hook
or off hook (without dialing it)?
Dave
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On Thu, 2003-03-13 at 15:31, [EMAIL PROTECTED] wrote:
> In the extensions.conf file, is there a way to detect if it is on hook
> or off hook (without dialing it)?
I do not believe so. What are you trying to accomplish?
--
Steven Critchfield <[EMAIL PROTECTED]>
_
On Thu, 2003-03-13 at 14:26, Ajit Kallingal wrote:
> Hello All,
> Can the current Asterisk be integrated with mySQL to query a database ?
> I am looking at a typical IVR scenario where the user punches a "product
> code" and the database query will determine if the product is available or
> not. T
On Thu, 2003-03-13 at 12:36, Jim Archer wrote:
> --On Thursday, March 13, 2003 11:11 AM -0800 William Walsh
> <[EMAIL PROTECTED]> wrote:
>
> > I presume the problem is in the CallerIDName that shows up when you
> > call a regular number through iconnecthere?
>
> It usually presents no caller ID
Can somebody look at the RTP packets with "ethereal" and tell me if they
notice any difference between what we send and what we receive? Perhaps
we're starting out with values that are too high or something?
Mark
On 13 Mar 2003, Matteo Brancaleoni wrote:
> I can confirm that.
> With the snom, I
Hello All,
Can the current Asterisk be integrated with mySQL to query a database ?
I am looking at a typical IVR scenario where the user punches a "product
code" and the database query will determine if the product is available or
not. The reply would be number of items available , else none.
Tha
--On Thursday, March 13, 2003 11:11 AM -0800 William Walsh
<[EMAIL PROTECTED]> wrote:
I presume the problem is in the CallerIDName that shows up when you
call a regular number through iconnecthere?
It usually presents no caller ID, resulting in an anonymous call rejection
message.
Technically
I can confirm that.
With the snom, I get no delay.
with a sip-fxs gw, I get 2 seconds delay.
Matteo
Il gio, 2003-03-13 alle 19:59, Lele Forzani ha scritto:
> On Thursday 13 March 2003 18:00, Benjamin Miller wrote:
>
> > Actually I've seen this exact issue with my Cisco 7960.
> > And it's any voi
On Thu, 2003-03-13 at 13:23, Steve Radich wrote:
> There should maybe be a few reminders sent out to posters saying this should
> have been in the dev mailing list.
>
> I would recommend though that something like a initial setup list be created
> for the questions like getting extensions working,
or Packet Telephony (Simple) Protocol
On 13 Mar 2003, Karl Putland wrote:
> What about ITP
>
> Internet/IP
> Telephony
> Protocol
>
> On Thu, 2003-03-13 at 09:40, Mark Spencer wrote:
> > > LIghtweight
> > > Voice over IP
> > > Exchange
> >
> > Or:
> >
> > Lightweight
> > Internet
> > Voice
> > E
When you get tired of SIPping you can upgrade to TASTEing... How do we go
from that to Asterisk?
Searching for taste though isn't going to return favorable VoIP results, not
that sip is great - but iax / iax2 is pretty distinct.
I do like taste though.
Steve Radich - Colocation / Virtual Dedica
I like the name IAX, only because I want to get "call piggybacking"
working and call it pigbIAX (pronounced "pig beaks"). :-)
If I were to chose a new name, TASTE is probably the one I like the
best...
On Thu, 2003-03-13 at 08:19, Mark Spencer wrote:
> What do you all think of renaming IAX2 a
What about ITP
Internet/IP
Telephony
Protocol
On Thu, 2003-03-13 at 09:40, Mark Spencer wrote:
> > LIghtweight
> > Voice over IP
> > Exchange
>
> Or:
>
> Lightweight
> Internet
> Voice
> Exchange
>
> Mark
>
> ___
> Asterisk-Users mailing list
> [EMA
There should maybe be a few reminders sent out to posters saying this should
have been in the dev mailing list.
I would recommend though that something like a initial setup list be created
for the questions like getting extensions working, setting up initial VoIP,
config voicemail, initial channel
On Thu, 2003-03-13 at 08:55, Jim Archer wrote:
> Still, I can't get it to work. Maybe its specific to my area. I think I am
> giving up on iconnect for a while. Thanks.
I presume the problem is in the CallerIDName that shows up when you
call a regular number through iconnecthere?
If so, that
Out of band DTMF for SIP seems to be broke.
I tried switching to dtmfmode=inband this
works fine for local phones, but any phones
over non LAN link, can not enter digits
without duplicates showing up, this is most
sever for the user name prompt in voice mail
main.
Is anyone working on getting out o
Before you prepend the "I", does IAX2 really depend on IP?
For example couldn't you send TASTE over ATM, FDDI, X25 or a
simple serial cable? My bet is that most VoIP would work better
if it weren't for the IP part. Yes I do have use for other
transports.
--- John Todd <[EMAIL PROTECTED]> wrote
Mark and List:
I agree with the earlier post that said keep the name IAX. No we don't
need to go through a progression like Algol60 -> Pascal -> Modula2 ->
Modula3...
To our multi-lingual listers - do IAX or TASTE have any non-English
complications? Remember how much fun General Motors had with
On Thursday 13 March 2003 18:00, Benjamin Miller wrote:
> Actually I've seen this exact issue with my Cisco 7960.
> And it's any voice prompt I dial. I loose just the very first .5
> seconds of the audio for whatever reason.
> So the sip users hear "eedian Mail" and "nk you for calling".
> Any on
I'm in favor of "TASTE" myself, though Mark's take on "LIVE" has the
all-important "I", to establish the use of this protocol over IP
networks, which is an important part of the protocol and conceptual
structure, yes?
Perhaps "ITASTE" with the "I" standing for the obvious "Internet".
JT
Wha
I still like my original idea, SVP - Streamlined Voice Protocol.
I think the trick to naming IAX2, is to encourage the thought of
streamlined and efficiency. If people want a feature-rich and bloated
protocol, they'll run H.323, SIP, etc .. IAX is all about performance and
resource utilization
TASTE... tastes good for me.
but have you thinked 'bout LISP ? ;-)
Lightweight
Internet
Signalling
Protocol
or SIVEX? Simple Internet Voice EXchange
WONVE : Working Over Nat Voice Exchange ;-)
think that any combination could be done...
matteo
Il gio, 2003-03-13 alle 17:40, Mark Spencer ha s
Denon mentioned to me:
SVP -- Streamlined Voice Protocol
Jeremy
Mark Spencer wrote:
LIghtweight
Voice over IP
Exchange
Or:
Lightweight
Internet
Voice
Exchange
Mark
Title: Message
Actually I've seen this exact issue with my Cisco
7960.
And
it's any voice prompt I dial. I loose just the very first .5 seconds of
the audio for whatever reason.
So the
sip users hear "eedian Mail" and "nk you for calling".
Any
one else dealing with this?
Any
ideas?
Morning,
Actually the phone will be running SIP, but from what I have read,
most people use the call manager for the services_url stuff. I read
the url mentioned below, and wrote some simple xml, but the phone
services emulator doesn't like it. My guess is that just doesn't work
correctl
What is "complex"? If all you need is a menus tree with static
menus (that is, the options are fixed) then the menus system
built-into Asterisk would work.
But if you want to serve up dynamic content, like custom menus
for each caller based on products they may have purchced from you
or what acc
> LIghtweight
> Voice over IP
> Exchange
Or:
Lightweight
Internet
Voice
Exchange
Mark
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At 09:19 13-3-2003 -0600, you wrote:
What do you all think of renaming IAX2 as:
Telephony Authentication, Signalling, and Transport Exchange (TASTE)
"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?
Sounds good. It slightly reminds me of a local frat-club,
[deeper]
exten => s,1,Playback,you-re-in-the-deepest-menu
exten => s,2,Goto,options|s|1
[options]
exten => s,1,Background,prompt-1-deeper-2-back
exten => 1,1,Goto,deeper|s|1
exten => 2,1,Goto,sales|s|1
[sales]
exten =>
s,1,Background,prompt-1-information-2-connect-or-stay-on-the-line-0-operator-*-
TEA : Telephony data Exchange with Authentication
goes well with SIP also :)
or,
RSTP: Really Simple Telephony Protocol
(gets out that unlike h323 and SIP, this is designed to be
simple, but way to easy to confuse with RTSP, or others).
-SteveK
On Thu, Mar 13, 2003 at 10:50
Still, I can't get it to work. Maybe its specific to my area. I think I am
giving up on iconnect for a while. Thanks.
--On Thursday, March 13, 2003 8:45 AM -0600 Mark Spencer
<[EMAIL PROTECTED]> wrote:
Mark - I beg to differ. Generally callerid works with Deltathree; but
sometimes they seem
Please disregard last
Will RTFM
Hallo
What soft clients can all be used
with asterisk?
Thanks
liaan
Finally, a question on this list I feel qualified to answer :D
I like it. Keeps the name from being confused with other
computer-related acronyms (a buddy of mine once asked why on earth I
kept referring to AIX as a VoIP thing... Surely, he said, I had to know
that it is an Operating System. It to
This feature is in development currently.
Mark
On Thu, 13 Mar 2003, Matteo Brancaleoni wrote:
> Hi, I have a mixed sip-zap evironment
> in my office. I was wondering if is
> possible to do remote call pickup
> from a sip phone, like from zap.
>
> Any hint?
>
> Matteo Brancaleoni
> [EMAIL PROTECT
You can playback a second or two of silence ...
regards
Martin
On Thu, 13 Mar 2003, T Aksoy wrote:
> Hi,
>
> We are testing a number of sip phones from different manufacturers. With one phone
> in particular, when I dial the asterisk voicemail, it misses around half a second
> from the beginni
Quoting Mark Spencer <[EMAIL PROTECTED]>:
> What do you all think of renaming IAX2 as:
>
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
>
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
>
> Mark
>
I still live "LIVE"...
LI
I didn't see any message regarding this subject
during the last days and I'm also interested in
pricing info to obtain G.723.1 and G.729 licences
for *.
If anyone has such this information, please send.
Tks,
Alejandro Olchik
Krzysztof Bujak said:
> Sorry for bothering you Jeremy.
>
> So asteri
I like it. It's better than STUN. *grin*
On Thu, Mar 13, 2003 at 09:19:11AM -0600, Mark Spencer wrote:
> What do you all think of renaming IAX2 as:
>
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
>
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP"
Hi, I have a mixed sip-zap evironment
in my office. I was wondering if is
possible to do remote call pickup
from a sip phone, like from zap.
Any hint?
Matteo Brancaleoni
[EMAIL PROTECTED]
Emmegi System Administrator
EspiA - EMMEGI Srl - e*solution provider
Uffici: Via Pascoli, 37
20129 Milano -
IAX is short and I like it. Besides if that additional '2' irritates you
then anyways in the near future when IAX2 is working fine ppl will switch
eventually to IAX2 and then we'll refer to IAX2 as IAX
Martin
On Thu, 13 Mar 2003, Mark Spencer wrote:
> What do you all think of renaming IAX2
Mark, you're a darn good coder. . ...
;-)
How about "TIP", Telephony Internet Protocol.
Or is that to close to Sip?
Ben
-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 13, 2003 10:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Proposed IAX2
Ok
Change the line in capi_chan.c( it think) to return 0 when capi not
installed.. now everything starts and loads fine.. m
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 15:00
To: [EMAIL PROTECTED]
Subject: RE:
CAPI == Common isdn API
it's not about talking to certain modems, it's a hardware
independent interface to access all kinds of isdn controllers
in the same way.
regards
kapejod
Am Don, 2003-03-13 um 15.28 schrieb Liaan van der Merwe:
> Ok... now I'm lost
> What is the idea then behind capi2.0 st
Aaa.. ok
Now it makes sense...
No I'm back to square 1.
Thanks
Cheers
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lars
Kneschke(priv.)
Sent: 13 Maart 2003 16:59
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
Liaan v
> Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
> >Oops... see. I need to learn bit more about this whole capi thing.
> >It will be a planet external isdn TA.. capi2.0 compatible.
>
> But not under Linux. The only card that support capi under linux are cards
> from AVM and Eicon.
>
> It want
What do you all think of renaming IAX2 as:
Telephony Authentication, Signalling, and Transport Exchange (TASTE)
"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?
Mark
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Asterisk-Users mailing list
[EMAIL PROTE
The capi drivers are not enough. You need to actually have CAPI hardware
working in your system.
On Thu, Mar 13, 2003 at 04:12:25PM +0200, Liaan van der Merwe wrote:
> Ok...
> Got the capi4linux.. install everything.. (no real updates done)
> How do I know whether is was installed correctly for i
Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
>Ok... now I'm lost
>What is the idea then behind capi2.0 standard??
>Is this not a "language" to talk to certain modems?
>
Capi is the interface for the software. You still need a driver which
translates the CAPI commands to the hardware commands.
The more logical breakup is asterisk-users and asterisk-dev which are both
there, and everyone who subscribed to [EMAIL PROTECTED] is now subscribed
to both the users and devel list.
Mark
On Thu, 13 Mar 2003, Roy Sigurd Karlsbakk wrote:
> join both
>
> On Thursday 13 March 2003 12:57, Michael Bi
> Mark - I beg to differ. Generally callerid works with Deltathree; but
> sometimes they seem to reject it / mess it up.
Nevermind, I was thinking this was incoming Caller*ID.
Mark
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Ok... now I'm lost
What is the idea then behind capi2.0 standard??
Is this not a "language" to talk to certain modems?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lars
Kneschke(priv.)
Sent: 13 Maart 2003 15:39
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk
On Thursday 13 March 2003 14:38, Lars Kneschke(priv.) wrote:
> It want work with external TA. If you need a external box, there will be
> USB ISDN card from AVM available, which will be supported by capi.
You _don't_ want to work with isdn4linux.
It's horrible when it comes to latency and echo.
--
Ok...
Got the capi4linux.. install everything.. (no real updates done)
How do I know whether is was installed correctly for it still says capi not
installed...
Thanks
Ps: the following modules are all loaded
- capi
- capidrv
- capifs
- kernelcapi
- capiutil
About every hour I see the yellow alarms on all or a number of channels of
my PRI which is connected to the dutch telephony network, Asterisk keeps
on working fine
Here's an example where channel 1-24 went into alarm:
WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected
ala
Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
>Oops... see. I need to learn bit more about this whole capi thing.
>It will be a planet external isdn TA.. capi2.0 compatible.
But not under Linux. The only card that support capi under linux are cards
from AVM and Eicon.
It want work with extern
this comes all the time, and has been there since I started using CAPI. I
don't know what it might mean ...
On Thursday 13 March 2003 13:52, Klaus-Peter Junghanns wrote:
> Morning Roy,
>
> capi says that the send queue is full. what did you stick in there? ;-)
>
> regards
> kapejod
>
> Am Don, 20
Ok.
So... my guess.. stuff the external isdn idea and get internal card.??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 15:00
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
H
That what I thought...
Mmmm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Bielicki
Sent: 13 Maart 2003 15:01
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_capi version 0.1.0 released
You are making a basic mistake here. External TA's ar
At 13:16 13-3-2003 +0100, you wrote:
join both
But then, if you have a question that has an overlap, should you post and
crosspost to both lists ? And comments that return on that question ?
Best regards,
Florian
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OK, let me rephrase that.
MFC/R2 isn't "a" protocol. Its a whole family - one for almost every
country, and two for Colombia. :-)
Which variant do you need? That is, which country are you trying to install something in?
Regards,
Steve
Claudio Aznar wrote:
The most common R2 version, MFC R2.
Oops... see. I need to learn bit more about this whole capi thing.
It will be a planet external isdn TA.. capi2.0 compatible.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sander Striker
Sent: 13 Maart 2003 14:36
To: [EMAIL PROTECTED]
Subject: RE: [Asteri
Hi,
this is funny :)
you expect it to work with no card
i dont think that there are external TAs with capi 2.0
linux drivers. isdn4linux wont help much...you want
capi4linux, but without a card it's not much fun ;-)
regards
kapejod
Am Don, 2003-03-13 um 11.11 schrieb Liaan van der Merwe:
>
You are making a basic mistake here. External TA's are treated by linux like
modems. isdn4linux faq tells you that explicitly.
It won't work.
On Thursday 13 March 2003 12:19, Liaan van der Merwe shaped the electrons to
say:
> I just installed all the latest isdn4linux stuff..
> Still same error
Morning Roy,
capi says that the send queue is full. what did you stick in there? ;-)
regards
kapejod
Am Don, 2003-03-13 um 12.35 schrieb Roy Sigurd Karlsbakk:
> hi
>
> I keep getting these errors all the time:
>
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> DATA_B3_
I send example XML files a week or so ago, as well as
an impotent line for you dhcp server.
Mike Reiling wrote:
Anyone know if it is possible to load your own XML scripts on to the
phone, bypassing the Cisco CallManager? I am still waiting for my
phone to arrive, but I have been playing with C
Message waiting indication work fine you
just need to set up DDNS for phones Asterisks
Needs to know how to reach the phones!
[2114]
type=peer
username=2114
insecure=yes
canreinvite=no
context=default
mailbox=2114
host=SIP003094C274B3.bna01.isdn.net
You will nee
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Liaan van der
> Merwe
> Sent: Thursday, March 13, 2003 1:20 PM
> I just installed all the latest isdn4linux stuff..
^^
You need capi4linux, not isdn4linux (although it is part of the
isdn4lin
I just installed all the latest isdn4linux stuff..
Still same error
Maybe asterisk wont work with external isdn devices.. any ideas?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 11:54
To: [EMAIL PROTECTED]
S
join both
On Thursday 13 March 2003 12:57, Michael Bielicki wrote:
> yeah what would we all carrier guys do who have completely mixed
> environments ? create a asterisk-carrier and a asterisk-oem and a
> asterisk-sysintegrator list ?
>
> On Thursday 13 March 2003 11:20, Florian Overkamp shaped the
I am very new to asterisk and SIP so these questions may be fairly
basic.
I have setup an asterisk system based on Red Hat 8.0 Linux with a few
analog cards from Digium (X101P's). I have also purchased a SNOM 200
phone and a couple of soft SIP clients for windows. Everything (so far)
is working nic
yeah what would we all carrier guys do who have completely mixed environments
? create a asterisk-carrier and a asterisk-oem and a asterisk-sysintegrator
list ?
On Thursday 13 March 2003 11:20, Florian Overkamp shaped the electrons to say:
> At 11:36 13-3-2003 +0100, you wrote:
> >Is it only me,
On Thu, 13 Mar 2003 12:20:03 +0100, Florian Overkamp wrote:
>At 11:36 13-3-2003 +0100, you wrote:
>>Is it only me, or is the asterisk mailing list growing rapidly?
>>
>>I've got an idea, although I don't know how good it is ;-)
>>
>>how about splitting the asterisk list between good old telephony
The preceding comments probably apply to direct analogue PSTN connections.
You may have problems if the line you are connecting to is from a PBX or
ISDN terminal adapter.
Iain
--On Thursday, March 13, 2003 11:49 am +0100 Klaus Darilion
<[EMAIL PROTECTED]> wrote:
Hello!
Does the X100P card
hi
I keep getting these errors all the time:
ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
DATA_B3_REQ (error=0x1102, datalen=160)
ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending
DATA_B3_RESP (error=0x1102)
ERROR[3076]: File chan_capi.c, Line 1081 (pipe
At 11:36 13-3-2003 +0100, you wrote:
Is it only me, or is the asterisk mailing list growing rapidly?
I've got an idea, although I don't know how good it is ;-)
how about splitting the asterisk list between good old telephony with channel
banks and telephones, and IP based telephony?
Hmm, I think
Hi
At 11:49 13-3-2003 +0100, you wrote:
Does the X100P card supports the european phone standards, especially
the one in Austria? Does someone have ever tried it and succeeded?
Yes it does, although I don't think its certified for use in australia yet...
Florian
_
It works in UK, DE and in Pakistan.
On Thursday 13 March 2003 10:49, Klaus Darilion shaped the electrons to say:
> Hello!
>
> Does the X100P card supports the european phone standards, especially
> the one in Austria? Does someone have ever tried it and succeeded?
>
> Regards,
> Klaus
>
>
yes! works w/o problems.
Matteo in italy ;-)
> -Messaggio originale-
> Da: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Per conto di
> Klaus Darilion
> Inviato: giovedì 13 marzo 2003 11.49
> A: [EMAIL PROTECTED]
> Oggetto: [Asterisk-Users] X100P in Europe/Austria
>
>
> Hello!
>
> D
Hello!
Does the X100P card supports the european phone standards, especially
the one in Austria? Does someone have ever tried it and succeeded?
Regards,
Klaus
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hi
Is it only me, or is the asterisk mailing list growing rapidly?
I've got an idea, although I don't know how good it is ;-)
how about splitting the asterisk list between good old telephony with channel
banks and telephones, and IP based telephony?
roy
--
Roy Sigurd Karlsbakk, Datavaktmester
On Wed, Mar 12, 2003 at 11:33:31PM -0600, Mark Spencer wrote:
> > Is anyone using MWI on 7960's when Asterisk is ONLY being used
> > for voicemail, and not for a gatekeeper?
> >
> > If anyone has MWI working successfully with 7960's, would it
> > be possible to get a dump of a successful NOTIFY mes
Hallo
To be honest.. no card as yet.
Truying to find out whether external TA will work.
I'll install latest i4l and see what happens
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 11:54
To: [EMAIL PROTECTED]
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