On Thu, 16 Sep 2004 16:12:10 -0700 (PDT),
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Could anyone with any experience with * over a non-PRI T1 help this newbie?
I have a fractional T1 that is working fine through a channel bank, but I
can't get any response on * using a T400P. My analog line
Seems to be alot of these questions on the mailing list recently. AUSTEL
is the old name for the ACA, A-tick is the correct term for certification.
It's only illegal if you connect to a carrier network without A-tick
(you can get consent from them to connect without A-tick).
The ACA has plently
I am developing a sip user agent i am having a problem
with my Callee..When i call from SJphone to my user
agent with Asterisk as the Sip Proxy, it does not
recognize by Ringing and Call answer messages.
___
Do you Yahoo!?
Declare Yourself -
[EMAIL PROTECTED] writes:
7) Provide the resulting sound files as a free download from your
website so that others don't have to do the same thing.
In fact, a library of multiple language versions of the standard texts
would be a cool thing for us to build. And, suddenly, the phrase My
ATT plugin is quite good
I've listened to them all and to me, the Rhetorical stuff stands out.
Probably why they think it should always be sold. Maybe if more people
badgered them...
Well, boyo, I can't do a welsh accent.
But I used to live down't pit sor if tha wants a good Yorkshire
I am upgrading from an X100P to a TE405P (T1/E1/QuadSpan) card. The
asterisk server is an IBM xSeries 300 running Fedora Core 1.
uname -a reports:
Linux hermes 2.4.22-1.2199.nptl #1 Wed Aug 4 12:21:48 EDT 2004 i686
The system has an Adaptec U160 SCSI card in it.
/proc/pci:
SCSI storage
My wife's got an appropriate Southern England (Wimbledon) accent and I'm
sure she would try her hand. Does anyone have a comprehensive list of the
words that need to be said? Matt, do you have them if your wife's done a
set for French users?
Mark, if you have the kit maybe you could chop up the
Found it. It's a Micronet-specific error message. So much for
standards... :-/
Evert Meulie wrote:
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: Evertsip:[EMAIL PROTECTED] ext. IP];tag=as6e18534e
To: sip:[dialled [EMAIL PROTECTED] server of VoIP provider]
Call-ID: [EMAIL PROTECTED]
rant
Especially when asked to press pound!
Pound! This is a pound £ not this #
rant-end
Mark, I would be happy to help and am actively seeking a suitable female,
and my father speaks taff !
D
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
are the scsi and te405p irq shared?
te405p hates shared irqs...
matteo.
--
Matteo Brancaleoni
System Administrator
[EMAIL PROTECTED]
EspiA Srl - e*solution provider
Via Pascoli, 37
20129 Milano - Italy
SIP:[EMAIL
Good day all,
I switched my SIP FWD account to IAX and connect my * in IAX. Working
great, but I face one problem: I have an iaxtel account and try to call
from there (iaxcomm) my FWD Iax # by 17009xx. It's ringing but no
termination on my *. Calling 1700612 or 17009SIP FWD# is working
On 17 Sep 2004 at 8:22, Bill Seddon wrote:
My wife's got an appropriate Southern England (Wimbledon) accent and
I'm sure she would try her hand. Does anyone have a comprehensive
list of the words that need to be said? Matt, do you have them if
your wife's done a set for French users?
You
Hello Jeremy,
thanks for your remark..
this is what i get out of it ...
--
#0 0x41f57c50 in oh323_new (i=0x80f8f50, state=0,
host=0x449e5147 213.xxx.202.xxx) at chan_h323.c:625
625 chan_h323.c: No such file or directory.
in chan_h323.c
#0 0x41f57c50 in oh323_new
Matteo Brancaleoni wrote:
are the scsi and te405p irq shared?
te405p hates shared irqs...
The single time that I had the machine up and running with the te405gp
and the adaptec in (which lasted the whole of 3 minutes before it
crashed), procinfo claimed that the adaptec was on IRQ 11, by
Hello!
I would like to call a number (e.g.35), and when i press a secret code
(12345), it should jump to my voicebox menu.
On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found
something about Silently Wait for DTMF Input.
In my case it wouldn`t be silence. It woudl just play
Good day all
I'm totally lost with this caller id,so can someone please help me
We are using a openline 4 card so in my vpb.conf I added callerid = on
And we are using sip as protocol so in sip.conf.
No each time a call comes in from the outside I dont se the number where
its coming from on my
Luis Vazquez wrote:
Hello all
We have tested for a mounth or two an asterisk PBX using one T1 channel
bank with 24 fxs and one TDM400P digium card with 4 FXO modules.
This worked with minor problems, the most notorious being some sporadic
static noice or failure in the first FXO module on the
Dont feel bad. Here in Newfoundland H is silent in some places. I have a
fried who's name is either Ellen or Helen :-) who's from Hawks pond or OX
pond depending on who says it
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Davies
Sent: Friday,
[EMAIL PROTECTED] wrote:
Hello!
I would like to call a number (e.g.35), and when i press a secret code
(12345), it should jump to my voicebox menu.
On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found
something about Silently Wait for DTMF Input.
In my case it wouldn`t be
Yep. If your wife will do the voice work I'll chop the file.
Saarff Wimbuldon eh? Don't say as hour I never do nuffink for ya!
Don't forget to translate the relevant words. We don't have zee or
pound in English (of course, being a Brit you already knew that)
Mark
Bill Seddon said:
My wife's
rant
Especially when asked to press pound!
Pound! This is a pound £ not this #
rant-end
Mark, I would be happy to help and am actively seeking a suitable female,
and my father speaks taff !
English gentleman seeks female for oral project?
Hmmm...!
Linus
On Thu, Sep 16, 2004 at 01:17:12PM +0200, Pawlowski Julian wrote:
Hello,
I just noticed an error in the current version of Klaus-Peter Junghanns
bristuff package, especially in the HFC module.
Everytime I try to unload the HFC module with modprobe -r I got a
kernel panic and the complete
hi,
i read a lot of papers about answering tone whenn i call outside and
inbound with sip phones.
in my case, there is no dial tone wenn i do a call outside or if
someone calls the sip phone.
how can i configure/play a ring/dial tone till the endpoint accepts the
call?
thx!
thomas
On Friday 20 August 2004 09:54, Massimo De Nadal wrote:
I've asked Grandstream tech support about attended transfer.
They told me that in about a month there will be available a firmware
upgrade that supports attended transfer natively.
maxx
any news?
--
Maurizio Marini GSM
English gentleman seeks female for oral project?
Hmmm...!
Linus
nice one :)
SJ
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To UNSUBSCRIBE or update options visit:
Hi everyone!
I'd like to create the following: a user picks up the phone (gets a dial
tone), dials '0' for an 'outside' line, gets a second (different?)
dialtone, and is able to enter an external phone number.
How do I implement this in extensions.conf...?
Regards,
Evert
I'd like to create the following: a user picks up the phone
(gets a dial tone), dials '0' for an 'outside' line, gets a
second (different?) dialtone, and is able to enter an
external phone number.
Klaus-Peter Junghanns has something like this on his page:
On Thursday 19 August 2004 23:29, administrator tootai wrote:
I made one. Can be found at
http://ftp2.tootai.net/spandsp-0.0.1k-whole.tar.gz The 3 headers files
are included, made a short readme file for installation and modify the
Makefile.patch (remove the dtmftotext). Comments welcome.
On Friday 17 September 2004 11:43, Evert Meulie wrote:
How do I implement this in extensions.conf...?
maybe this may help...
http://lists.digium.com/pipermail/asterisk-users/2004-February/036737.html
--
Maurizio Marini GSM +39-335-8259739
Work: +39-0721-855285 Fax +39-0721-859609
Home:
Spandsp
I cannnot get the command patch Makefile.patch to work when trying to use
rxfax etc.
The app_rxfax.c and txfax are in the apps dir but the command just sits
there and doesn't do anything.
It's a standard cvs install, has anyone got a working patch makefile I can
grab ?
Cheers
d
Ignore me it worked, although grabbing the lastest cvs and doing make clean
; make install in zaptel error'd
Odd
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Davies
Sent: 17 September 2004 11:34
To: 'Asterisk Users Mailing List -
thanx andrew
first of all
your messages are in Plain Text
format!
i hv monitored Asterisk both managerAPI console and
Asterisk main console to see wht is actually going on .when a new incoming
connection comes.
when the phone is ringing.it gives
starting simple swithc on 'ZAP/1-1'
and
Isn't it possible to use T.38 for interconnecting hardware gates
supporting T.38 with asterisk using SIP REINVITE?
I'm not shure but but think its's might be possible because after
reinvite traffic goes directly from one gate to anotger, not over
Asterisk
--
Sincerely,
Elman Efendiyev
[EMAIL
[EMAIL PROTECTED] wrote:
Isn't it possible to use T.38 for interconnecting hardware gates
supporting T.38 with asterisk using SIP REINVITE?
I'm not shure but but think its's might be possible because after
reinvite traffic goes directly from one gate to anotger, not over
Asterisk
We've
And what about using same codecs for asterisk and endpoints? Lets say
G.729. Yes, it needs license but while G.729 is industry standart
de-facto I thing most of us need to use it anyway
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
Hi
I have reason to believe that I have errors in my
configuration because when I make a call I can see the H323 call executed ok but
not being processed by Zap. I am using R2 signaling (which I know is
incomplete but should I not see it when I debug Zap channel?). I think there is
a
Hi
I have reason to believe that I have errors in my
configuration because when I make a call I can see the H323 call executed ok but
not being processed by Zap. I am using R2 signaling (which I know is
incomplete but should I not see it when I debug Zap channel?). I think there is
a
exten = _9NXX,1,Dial(Zap/G1/${EXTEN})
Zap/g1 = hunts for the first available channel in group 1
Zap/G1 = hunts for the first available channel in reverse order in group 1
Is it possible, code wise, configuration wise, at all - to reverse the
order in which the available zap channels are used
Hi
I have reason to believe that I have errors in my
configuration because when I make a call I can see the H323 call executed ok but
not being processed by Zap. I am using R2 signaling (which I know is
incomplete but should I not see it when I debug Zap channel?). I think there is
a
[EMAIL PROTECTED] wrote:
Hello!
I would like to call a number (e.g.35), and when i press a secret code
(12345), it should jump to my voicebox menu.
On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i
found something about Silently Wait for DTMF Input.
In my case it
On the Asterisk side your firewall shall allow UDP port 5060 for SIP
and some UDP ports for RTP (default 1-2 can be changed at
/etc/asterisk/rtp.conf). Your sip.conf shall have Qualify=yes and
Nat=yes.
On the telephone side, as long as your firewall allows outgoing
traffic on 5060 and on
Maurizio Marini wrote:
On Friday 17 September 2004 11:43, Evert Meulie wrote:
How do I implement this in extensions.conf...?
maybe this may help...
http://lists.digium.com/pipermail/asterisk-users/2004-February/036737.html
Thanks! That works like a charm! The only thing I'd like to do now is
NOT
On Fri, 17 Sep 2004 [EMAIL PROTECTED] wrote:
What you want is an extension 12345 in the same context as the
extension 35 that will be used when you dial 12345 while background is
playing the message.
in the 12345 extension you do the normal 'voicemailmain' with the skip
password
NOTICE[98310]: chan_sip.c:6638 handle_response:
Failed to authenticate on INVITE to
'sip:[EMAIL PROTECTED];tag=as0f1d3429'
sip.conf
register =
1234:[EMAIL PROTECTED]
extension.conf
--
;; Own extensions;exten =
Hi Matt Riddell
Here you have the write permission for all
drwxrwxrwx2 root root 4096 Sep 14 16:09 incoming
And about your question; How long does it time out for? It stay there
without hang up until I switch off the fax machine.
I have installed tiff-v3.6.0
Hi everyone!
The following: Any calls coming in on extension 12121212 should get a
message telling them to dial the extension of the person they are trying
to reach, and then press #.
The call should then go to the entered extension.
This is as far as I got...
See, that's just the thing. I didn't. It just worked! I did some
limited packet traces, and it seemed to be working from a SUBSCRIBE.
I don't know of any commands in Asterisk to see what's happening in
more detail at higher layers.
- Jeff
On Fri, 17 Sep 2004 12:23:13 +1000, Paul Hales
On Fri, 17 Sep 2004 [EMAIL PROTECTED] wrote:
Hi,
This may work:
[working]
exten = 39,1,Answer()
exten = 39,2,GoTo(working-busy,s,1)
[working-busy]
exten = s,1,Background(tt-allbusy)
exten = s,2,Voicemail(35)
exten = s,3,Hangup()
exten = 123,1,VoicemailMain,s35
This works very well,
Hello!
After settting up my voicebox, i need to figure out how i can get my
numpad of kphone working.If i call the number with a normal phone, i can navigate
through the menu
just fine.
Here my sip.conf:
---
[general]
dtmfmode=rfc2833;rfc2833
context=default
On 17 Sep 2004 at 8:29, Angel Diaz wrote:
Hi Matt Riddell
Here you have the write permission for all
drwxrwxrwx2 root root 4096 Sep 14 16:09 incoming And
about your question; How long does it time out for? It stay there
without hang up until I switch
I have the following setup a Norstar MICS 0X32 with 8 POTS Lines
connected to the PSTN, and one ASTERISK server connected to the Norstar
MICS VIA a PRI line.
Now here is the problem I cannot get the MICS to accept a call from the
ASTERISK SERVER when that call is for an outside line(meaning dial
We have set up an IP telephoney system hosted by Asterisk and its working
pretty well. We primarily use SIP and hardware IP phones. We have the
ability to transfer calls to another SIP phone using either the Transfer
button on the phone (these phones are Grandstream BudgeTone 100s) or using
Hi,
I'm looking into support for SS7 and I found an article
(http://www.openss7.com/news13022002.html) which says that OpenSS7
supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI
interface cards. It also says that Linux Support Inc is the primary
sponsor of Asterisk.
So was there an answer to this?
We have at least 10 T1 lines into our Cisco router and from the router, 1
Fast Ethernet going into the * server.
If we slap another nic card into the * server, will/can * do any kind of
load balancing between the two interfaces?
We are not using any Zap cards as
Linux SUpport Inc = Digium
On Fri, 17 Sep 2004 14:26:46 +0100, Alex Zeffertt
[EMAIL PROTECTED] wrote:
Hi,
I'm looking into support for SS7 and I found an article
(http://www.openss7.com/news13022002.html) which says that OpenSS7
supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7
Alex Zeffertt schrieb:
...
I'm looking into support for SS7 and I found an article
(http://www.openss7.com/news13022002.html) which says that OpenSS7
supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI
...
Hi,
I'm currently beta testing the TE410P with SS7 together with
a
I am
getting this also.
I am
trying to get Asterisk to talk similarly to BT Communicator to the BT server. I
can register but then the INVITE fails.
BT are
mixed up with theirdomains (in fact in the INVITE their software has a To:
header withnumber@domain1 and an auth URI referencing
Hi All,
When I call the stream_file function all goes well if the user doesn't
clear the call. But if I do clear the call (on the handset for
example), I get the following exception:
-- Channel 0/31, span 1 got hangup
RESULT_LINE: 200 result=-1 endpos=28000
==
On Fri, 2004-09-17 at 08:26, Alex Zeffertt wrote:
Hi,
I'm looking into support for SS7 and I found an article
(http://www.openss7.com/news13022002.html) which says that OpenSS7
supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI
interface cards. It also says that Linux
/pipermail/asterisk-users/attachments/20040917/abdd5b
46/attachment-0001.html
--
Message: 7
Date: Thu, 16 Sep 2004 11:45:21 -0400
From: Andrew Thompson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] how to get caller ID
To: Asterisk Users Mailing List - Non-Commercial
On Fri, 2004-09-17 at 08:30, Matthew Boehm wrote:
So was there an answer to this?
We have at least 10 T1 lines into our Cisco router and from the router, 1
Fast Ethernet going into the * server.
If we slap another nic card into the * server, will/can * do any kind of
load balancing between
On Fri, 2004-09-17 at 09:01, Martyn Russell wrote:
Is there any way to know if the return code (-1) is a clear or channel
failure?
Does it matter. If you receive a -1 you know the line is no longer
available and no other commands will be accepted. So all that is left
for your app is to clean
On Sat, 18 Sep 2004, vrushank wrote:
thanx andrew
first of all
your messages are in Plain Text format!
plain text format is the preferred format for this (and most?) mailing
lists. Replying to a digest and not trimming the unrelated portion before
posting isn't a good way to earn points
Roger
I'm currently beta testing the TE410P with SS7 together with
a partner, who will present SS7 support for asterisk is some
weeks, maybe some days.
GULP :)
Is this GPL effort ?, and expected to be a standard part of asterisk cvs ..?
ps. never fails to amaze what going on the bacgrod with *
On Thu, 16 Sep 2004, Christopher L. Wade wrote:
The subject says it all.
Is it possible, code wise, configuration wise, at all - to reverse the
order in which the available zap channels are used for *outgoing* calls?
Code wise, I looked at the channel structure and it appears as though
Please do not respond to all. I don't need a copy privately mailed to me
outside of the list.
On Fri, 2004-09-17 at 09:35, Martyn Russell wrote:
On Fri, 2004-09-17 at 14:58, Steven Critchfield wrote:
On Fri, 2004-09-17 at 09:01, Martyn Russell wrote:
Is there any way to know if the return
TC wrote:
Roger
I'm currently beta testing the TE410P with SS7 together with
a partner, who will present SS7 support for asterisk is some
weeks, maybe some days.
GULP :)
Is this GPL effort ?, and expected to be a standard part of asterisk cvs ..?
ps. never fails to amaze what going on the
TC schrieb:
...
Is this GPL effort ?, and expected to be a standard part of asterisk cvs ..?
...
Hi,
I answered to the original email of this thread,
because I announced some months ago, that I will
look for any SS7 solution for asterisk. At that
time I had several solutions in mind, and I told,
...
I'm looking into support for SS7 and I found an article
(http://www.openss7.com/news13022002.html) which says that OpenSS7
supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1
PCI
...
I'm currently beta testing the TE410P with SS7 together with
a partner, who
Alex Zeffertt schrieb:
...
work the same with a TE405P instead.
Will you announce the support on this list? Will there be a website
...
Hi,
it also was tested with the TE405P - no difference.
(Any difference would be suprising.)
We have still some bugs, which prevent stable operation.
The author
Dear All,
How can I make a call transfer and line release after connected?
I've found the Transfer(Zap/..) is not working as expect Thanks for your
help!
regards,
R Wong
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On Fri, 2004-09-17 at 15:31, Steven Critchfield wrote:
If, for example, the channel fails because the PRI has been pulled out
of the back of the card, that is serious. We need to know if it is
serious or expected, and the user clearing in the middle of the call is
expected.
This isn't
James Golovich wrote:
Definitely not a asterisk-dev post, keep this stuff on asterisk-users
please.
With Zap groups you have a few ways to handle it.
Zap/g1 is group 1 starting from the beginning each time
Zap/G1 is group 1 starting from the end each time
Zap/r1 is group 1 round robin from the
Greetings,
I've been looking through the docs, but haven't located any info on how to
autodial a number with an extension.
I've got a working autodial set up where I can get asterisk to dial a
number and play a prerecorded message, and ask for confirmation of
receipt. Is it possible to get
I've asked Grandstream tech support about attended transfer.
They told me that in about a month there will be available a firmware
upgrade that supports attended transfer natively.
I never heard this, SWEET! You're not kidding right? This is something the
phone REALLY needs. Now if they could
Hi,
I'm looking for a way to do queue management for several phones. Basically I
want the phones to act as normal, except that incoming calls get queued if
the phone is busy at that moment. This can be achieved with one queue per
extension that wants this behavior (and actually works well like
On Fri, 17 Sep 2004, Jakob Borg wrote:
I'm looking for a way to do queue management for several phones. Basically I
want the phones to act as normal, except that incoming calls get queued if
the phone is busy at that moment. This can be achieved with one queue per
extension that wants this
2 new Cisco 7960 phones are requesting a CTLSEP file, seems like
I triggered the universal application loader. I want to load the
sip image 7.2
According to this Cisco information:
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps4967/products_upgr
What version firmware does your phone currently have?
on my 7940's 7960's, I've had to stair step each firmware
version, starting at 3 in order to get to 7.2
Niles
On Sep 17, 2004, at 10:30 AM, Jan Baggen wrote:
2 new Cisco 7960 phones are requesting a CTLSEP file, seems like
I triggered the
Alex Seffert wrote:
supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI
interface cards. [snip] However I cannot find these cards on the Asterisk hardware
page
Are these cards any different from the TE4xxP cards?
The T400P and E400P are the original Zaptel PRI PCI
How to load the SIP image on these phones?
create emtpy required files... that will trick it.. :)
ta
SJ
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To UNSUBSCRIBE or update
What version firmware does your phone currently have?
on my 7940's 7960's, I've had to stair step each firmware
version, starting at 3 in order to get to 7.2
Niles
i don't know, I can't use the phone menu because
it's still looking for the .tlv files.
On the display I get the error:
Protocol
Ah, this brings up an interesting point. I've noted that BT are calling #
square rather than hash. What do the other providers call it back in
Blighty?
Before someone goes recording the files we'd better get the language
straight.
Mark
rant
Especially when asked to press pound!
Pound! This
I have several new Cisco 7960 7905 IP Phones for sale. Phones come w/ the
power cubes cables.
If you're interested please e-mail me off list.
Thanks,
Jon
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How to load the SIP image on these phones?
create emtpy required files... that will trick it.. :)
ta
SJ
When I create an empty CTLSEPmac.tlv files I start
looking for SEPmac.xml.cnf files. When I create this
file also (empty) I will ask for the CTLSEP again :(
No requests for SIP..
Hash!
When we had the American voices on our system a lot of people complained, not only
that it was American (no offence to Americans!) but also because of the terminology
used, e.g. `pound`.
We re-recorded all our voice files and use `hash`.
Ben Merrills
Griffin Internet
-Original
Hi!
I am trying to suppres the transmission of my CallerID when I place a
call using a .call file in /var/spool/asterisk/outgoing
Callerid:
Callerid: and
Callerid: '' made the call transmit the default number (headnumber+0)
Callerid: 1234 made the call transmit 1234
Using *31* in front of the
Most Pbx's I have worked with use hash in the uk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: 17 September 2004 16:51
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] English vs American voice files
Ah, this brings up an
I've just installed asterisk as a new phone system for our office but am
having difficulty with the queues. Specifically I need a way to
redirect our sales queue to voicemail when no one is logged in to the
queue. I see I can use the joinonempty=no setting, however this setting
doesn't work
On Thu, 2004-09-16 at 00:02, Marconi Rivello wrote:
Is there a way to enable busydetect only for inbound calls?
Or to change the busycount setting with asterisk already running?
PS: For analog lines, using X100P.
1) No and I cannot imagine any reason that this would be useful.
Busydetect
I have about 10 of these phones. All of them started with sccp firmware on
them. I used the XMLDefault.xml file to upgrade them all to SIP 3.7. The
phones should look for a CTLmac file, then a SEPmac, then SIPmac.
XMLDefault.xml should be in there somewhere.
I've never had to 'trick' my phones.
On Thursday 16 September 2004 04:27 am, Vlasis Hatzistavrou wrote:
Hello all,
We have been testing Asterisk RC2 with the native H323 channel driver.
We followed the instructions with the needed OpenH323 and PWLib versions
and everything compiled ok. Operation of the driver seems ok,
+++ Tenorio, Leandro [15/09/04 10:58 -0300]:
I've seen a lot of times, people that try to get R2 MFC to *, most of them trying to
use Dialogic Boards (BTW They 're Very expensive), none of them where succesfully,
If you want to use PCI Cards on your server, why don?t u ask to your carrier to
I just did some tinkering with my IP500s. If I send an instant message to
someone by just typing in their extension, I can see that Asterisk is
receiving the message (The Polycom sends to ext@(IP of Asterisk box)) but
Asterisk does not forward the message on the other Polycom.
If I address the
Hi,
I am new to asterisk, I would like to know how to
configure asterisk to play a recorded message
everytime a call if forwarded to it from SER. Which
configuration files should be touched? I was scanning
through the information on the internet and didn't
find what I was looking for.
Thanks.
I saw several threads requesting an Asterisk forum to
complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html
I recently created an Asterisk forum within TMCs
popular VoIP forums for everyone to use.
On Friday 17 September 2004 13:06, Tom Keating wrote:
I saw several threads requesting an Asterisk forum to complement the
email list. i.e.
And the fragmentation begins.
On the plus side, perhaps the newbies will be drawn to the glitz and glamour
of the land of refresh, emoticons and
Hi,
Bit of a puzzle this one - let's see if anybody else can shed some light...
I Ghosted my Asterisk box to build one for a colleague.
I added one HFC card to make the total two and amended zaptel.conf and
zapata.conf accordingly.
I tested it with my handsets on my ISDN lines and it worked
Put another way, if BT call it square, its almost certain nobody else in
the UK does :-)
Regards,
Steve
David Davies wrote:
Most Pbx's I have worked with use hash in the uk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: 17 September
On Fri, 2004-09-17 at 12:04, Andrew Kohlsmith wrote:
On Friday 17 September 2004 13:06, Tom Keating wrote:
I saw several threads requesting an Asterisk forum to complement the
email list. i.e.
And the fragmentation begins.
On the plus side, perhaps the newbies will be drawn to the
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