I have successfully set up a conference room on my asterisk server,
I have been trying to make the 'M' for music on hold option work (when
the first person enters the room they are told they are the first and
then they are supposed to hear music on hold) but it didn't matter which
way I wrote it
On Wednesday 21 April 2004 12:03 pm, kiran p wrote:
Hi
My motto is to connect two computers on the same
network with Voip without using any special hardware,i
have downloaded Asterisk, I was suggested to use
LinPhone as a soft phone as it is very easy to install
I have installed Asterisk
Argh!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: vrijdag 23 april 2004 7:25
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] :)
Argh, i don't like the plaintext :)
archive password: 45703
Thanks to this
Geert Nijpels wrote:
Ian White wrote:
On recent releases of the snom200 firmware, the MWI indicator will
turn on, but won't turn off when the message has been checked. It
works on firmware 2.03o, but not in 2.04g or newer. I filed a bug
report with snom, but they're claiming it is an asterisk
Hello,
Can someone help me. I got zaptel.0.9.1.tar.gz from ftp.asterisk.org,
And then I uncomment the line with MODULES #
ztdummy,
run make clean, make, make install
But errors happens as follows:
--
make:
zaptel.c:5937: storage size
Hello,
Can someone help me. I got zaptel.0.9.1.tar.gz from ftp.asterisk.org,
And then I uncomment the line with MODULES #
ztdummy,
run make clean, make, make install
But errors happens as follows:
--
make:
zaptel.c:5937: storage size
On Fri, 2004-04-23 at 08:43 +0200, Florian Overkamp wrote:
Thanks to this message where a virus chose to use my from-address to send
its crap from I am now being harassed with many many virus warning messages.
A call to anyone operating virusscanners (as I am too): I think we can all
do
I have also complained about the change in MWI to SNOM.
My 2.03o phones still work with Asterisk but 2.04 versions do not.
However, you can turn off the MWI by pressing the MWI button but not
remotely ( NOTIFY ).
I once got the example under from SNOM ( Asterisk version is under it ).
According
On Thursday 22 April 2004 07:05 pm, Joel Duffield wrote:
We want to use asterisk to extend our current phone system. It is a
regular plain old system. Has anyone done this before?
Absolutely - in a lot of different ways.
We would be
adding about 4 SIP (probably Cisco) phones to use with
OK, so I'll do that... Is there any infos I need to know about chan_sip.c
(because I suppose it's it that I need to play with)?
Does anyone know an interesting website where I can find infos about UUI in
ISDN (with CAPI maybe?) ?
Thanks for your help.
Might I humbly request someone, somewhere in the community establish a
dummies guide to asterisk kind of site, that explains in detail what the
cryptic scripts actually do, line by line.
The Wiki is helpful, but unless you were in on the movie from the first
part, the scene discussions are
Good day all
I want to put the openline4 card into a box that will support 3
different companies
I read the caller ID id fixed but now HOW DO I:
If a call come in for 12345 it plays company 1's welcome message
If a call come in for 98765 it plays company 2's welcome message
ens..
Does This make
Why is there such a variation in price between what
the two of you have paid to get the SIP image for a 7960 phone ? $8 would
be acceptable, but I don't want to have to pay $105 !
What website do I have to go to in order to buy a
SIP image update ?
How long does the login last for - I mean
Altus Snyman wrote:
Good day all
I want to put the openline4 card into a box that will support 3
different companies
I read the caller ID id fixed but now HOW DO I:
If a call come in for 12345 it plays company 1's welcome message
If a call come in for 98765 it plays company 2's welcome message
On Friday 23 April 2004 12:33 am, David Krider wrote:
I've downloaded the entire archive of articles and searched through them
for an answer on this, but I haven't come across one yet. I'm looking to
replace a small phone system in my church with Asterisk, and I'm stuck
looking for phones. I
The thing is its 3 companies,3 different number 3 different lines.
I know you can sort it with source number(That old girlfriend thing) but
what about destination number,can you get it
On Fri, 2004-04-23 at 10:19, Jeremy McNamara wrote:
Altus Snyman wrote:
Good day all
I want to put the
I'm using asterisk with isdn hfcpci carc
(driver zaphfc)
all work correctly during the day but
during the night it happend something that hang the
card
with this message: zaphfc: empty HDLC frame
received.
Asterisk work without any error message but isdn
doesen't work
I must stop asterisk
(B--
$B>.ED??G7(B [EMAIL PROTECTED]
(B
(B
(B
(B___
(BAsterisk-Users mailing list
(B[EMAIL PROTECTED]
(Bhttp://lists.digium.com/mailman/listinfo/asterisk-users
(BTo UNSUBSCRIBE or update options visit:
(B
On Friday 23 April 2004 07:55 am, tmpm wrote:
Might I humbly request someone, somewhere in the community establish a
dummies guide to asterisk kind of site, that explains in detail what the
cryptic scripts actually do, line by line.
The Wiki is helpful, but unless you were in on the movie from
I have just got 3 Cisco 7960 phones which I would like to connect to
Asterisk...
However they seem to have v3 SCCP firmware.
I have tried numerous links to the Cisco Website but unable to get the SIP
firmware.
Has anyone managed to get a service contract or an account with download
privileges?
you should get that from the seller of the phones,
they must have a CCO login with donwload privs
and give you the firmware.
but if u bought them used, that's another story
It's not legal to share cisco firmware without authorization...
Matteo.
Il ven, 2004-04-23 alle 10:38,
Altus Snyman wrote:
The thing is its 3 companies,3 different number 3 different lines.
I know you can sort it with source number(That old girlfriend thing) but
what about destination number,can you get it
Then u can separate each line out into its own context
[company1]
exten = s,1,Answer
But who do I differentiate between the different number,how do I say: if
a caller calls 1234(the destination) do:
[company1]
exten = s,1,Answer
exten = s,1,Playback,company1-welcome
ens.
On Fri, 2004-04-23 at 10:44, Jeremy McNamara wrote:
Altus Snyman wrote:
The thing is its 3 companies,3
Hi,
I've installing a AVM Fritz Card in my ASterisk Box
I've configured everything and its running perfectly.
The problem is that everybody is allow to call through it.
Explaination:
All users registered in Asterisk can make a call towards the ISDN network
But, everybody from the Internet,
Hi,
Does Asterisk support NCS signalling?
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
ever heard of a 'correct dialplan' ?
perhaps there's some bug in your context/extensions
logic that let this happens.
better review it :)
Matteo.
Il ven, 2004-04-23 alle 11:20, Ignace CARIA ha scritto:
Hi,
I've installing a AVM Fritz Card in my ASterisk Box
I've configured everything
On Fri, 23 Apr 2004, Altus Snyman wrote:
But who do I differentiate between the different number,how do I say: if
a caller calls 1234(the destination) do:
[company1]
exten = s,1,Answer
exten = s,1,Playback,company1-welcome
ens.
In response to Jeremy McNamara, who on Fri 2004-04-23 at
Altus Snyman wrote:
But who do I differentiate between the different number,how do I say: if
a caller calls 1234(the destination) do:
[company1]
exten = s,1,Answer
exten = s,1,Playback,company1-welcome
ens.
Normally this would be done by setting a context for each DNO in the
device's
The procedure was changed. I'm sending that directly.
We'll need to know who actually downloads that.
If anybody else needs it, please contact me off-list.
Best regards Pertti
Steven Elliott wrote:
On 22/04/04 8:50, Pertti Pikkarainen [EMAIL PROTECTED] wrote:
Good day all
I'm trying this
Roger that, Ill grep. er google for it...thanks...
At 04:10 4/23/2004, you wrote:
There is the handbook on the homepage and then there is the hitchhikers
guid,just not sure where it is
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Im hoping that light bulb will glimmer on any day now...heh...
BTW - I contributed to the Wikki on more than a few occasions when I
_finally_
had an epiphany understanding Asterisk, and I plan to do so more in the
future.
Anon
___
Asterisk-Users
Hello
I use asterisk ver 0.7.2
Can I play any wave file into the client riciever without billing count ?
I call from A IAX client to B IAX client.
B client is not available and I would like to play some file with the message
user_is_unavailable.gsm
But when I look into my CDR table, this call
Dudlik wrote:
Hello
I use asterisk ver 0.7.2
Can I play any wave file into the client riciever without billing count ?
I call from A IAX client to B IAX client.
B client is not available and I would like to play some file with the message
user_is_unavailable.gsm
But when I look into my CDR
Hi all,
Windows Messenger 4.6behind NAT works fine
with * for me, except the NOTIFY forMWI and voicemail. TheNOTIFY
message triggers a 481 error.How can I make it right? I am using *
current stablerelease.
Thanks.
Ben
than you
and I have Wildcard TE410P in my * server
What can I do when a client A call from another telecomunication operator over E1 to
my IAX client ?
Telecomunication operators usually use the unavailable messages and I thing they don't
bill these calls between their customers.
How do they
Dudlik wrote:
than you
and I have Wildcard TE410P in my * server
What can I do when a client A call from another telecomunication operator over E1 to
my IAX client ?
Telecomunication operators usually use the unavailable messages and I thing they don't bill these calls between their customers.
you have sent a message to me which seems to contain a legal warning
on who can read it, or how it may be distributed, or whether it may be
archived, etc.
i do not accept such email, and have therefore deleted it. do not
expect further response.
randy
subscribers to the digest form of this list do so in order to
only receive the email infrequently. in my case, and i suspect
others, twice or so a day would be preferred. the list currently
batches about every hour. it is sufficiently annoying that one
tends to delete batches. i have written
Title: Message
You
don't say which version you are using, but upgrade to RC20a. There were
some ISDN Layer 2 issues in earlier versions which have been fixed
recently.
http://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gz
Rgds
Tim
http://bugs.digium.com/bug_view_page.php?bug_id=0001474
If you're from NZ and need this, please test if this is the correct setup.
Add your comments, positive or negative, to the bug tracker. We need
confirmations from the community to move ahead.
Thank you!
/O
On Fri, 23 Apr 2004, Johnson-Perkins, Robert wrote:
I have just got 3 Cisco 7960 phones which I would like to connect to
Asterisk...
However they seem to have v3 SCCP firmware.
The same question, posted a few hours before:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044025.html
Tim Sailer wrote:
Folks,
I'm looking for a SIP or IAX phone for field techs to take with them
when out on service calls. The regular desktop phones are just way too
big. Is there anything like the size of a full-sized cell phone? Or
smaller, not I doubt that...
If a softphone is acceptable
On Thursday 22 April 2004 07:05 pm, Joel Duffield wrote:
We want to use asterisk to extend our current phone system. It is a
regular plain old system. Has anyone done this before?
Absolutely - in a lot of different ways.
We would be
adding about 4 SIP (probably Cisco) phones to use with
Can you put this patch on line? (I don't think it's too big...)
In my mind, the main objective is to create a special field and force
its value in chan_capi.c and check wether it goes through asterisk or
not...
What do you think of that?
Regards
--
[EMAIL PROTECTED]
Hello.
I am a spanish student, so excuse my English. I have
this HW:
- 2 X100P PCI with two analog lines plugged in. These
lines are two extensions of a panasonic PBX.
Zap/1 = X100P -- analog line -- extension
#237 PBX Panasonic
Zap/2 = X100P -- analog line -- extension
#245 PBX
You don't need a timing source for Music on Hold and have not needed one
for a while. I don't recall exactly when this requirement was removed
but it was well before 0.7.1. You do still need a timing source for
MeetMe and IAX Trunking (which you only want, but not need, if you have
lots of calls
On Fri, 2004-04-23 at 03:12, Paul Tyreman wrote:
Why is there such a variation in price between what the two of you
have paid to get the SIP image for a 7960 phone ? $8 would be
acceptable, but I don't want to have to pay $105 !
The $8 service contract gives you access to the Cisco software
Title: Message
Yes i use this version
Thank's Tiziano
- Original Message -
From:
Robinson Tim-W10277
To: '[EMAIL PROTECTED]'
Sent: Friday, April 23, 2004 2:59
PM
Subject: RE: [Asterisk-Users] Problem
With zaphfc
You
don't say which version you are
At 2:23 AM + on 4/23/04, Anon wrote:
On Friday 23 April 2004 12:33 am, David Krider wrote:
I've downloaded the entire archive of articles and searched through them
for an answer on this, but I haven't come across one yet. I'm looking to
replace a small phone system in my church with
For example, when an input call comes through X100P,
my Zap/3 extension rings. I pickup Zap/3 and I want to
transfer the call to Zap/4, but before to establish
the call between X100P and Zap/4 I need to request
Zap/4 for answering the call.
Currently not possible, although here is a
Rich,
Thanks a bunch, totally understand now and that actually makes total
sense. (no need for schematics). This also explains why I used an TA750
to go into a Nortel MICS system, using FXO and no buzz. Totally balanced
load from the analog ports on the Nortel across the 5 feet of CAT5 to
the FXO
rc19 work better for me
rc20a is less stable on my configuration (driver crash / line 50% not
correctly hangup)
At 15:37 23/04/2004, you wrote:
Yes i use this version
Thank's Tiziano
- Original Message -
From: mailto:[EMAIL PROTECTED]Robinson Tim-W10277
To:
mailto:'[EMAIL
On Fri, 23 Apr 2004, Paul Tyreman wrote:
What website do I have to go to in order to buy a SIP image update ?
When I bought mine, I did a Google search on their part number:
SW-SM-UL-7960 (Cisco SIP license for 7960 IP Phone)
Also, read this message:
If the $8 service contract only gives you access to
the image, but you aren't really allowed to use it, then why do Cisco offer that
contact in the first place ?
So are you telling me that to be legal, I need to
pay$105, but could get away with $8 ?
-Original Message-From:
Try with :
channel = 1-2
Regards,
At 11:40 20/04/2004, you wrote:
Hello,
Here it goes:
zaptel.conf:
---
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
---
zapata.conf
---
switchtype = euroisdn
signalling = bri_net_ptmp
pridialplan=local
echocancel=yes
immediate=yes
group = 1
context=local
On Fri, Apr 23, 2004 at 08:37:42AM -0500, Eric Wieling wrote:
On Fri, 2004-04-23 at 00:39, James H. Thompson wrote:
A standard butt set (e.g. http://www.sandman.com/pdf/page81.pdf) combined with a
Grandstream (very
small) or Sipura ATA would make a pretty small combination and be useful
in the [context] set in zaptel.conf
;
exten = 6165551212,1,NoOp
exten = 6165551212,2,Wait,2; seconds to wait before pickup
exten = 6165551212,3,Answer
;
On Fri, 23 Apr 2004, Mark Olliver wrote:
Hi,
I seam to have a problem working out how to get my X100P to answer after
1 ring.
On Fri, 2004-04-23 at 09:11, Paul Tyreman wrote:
If the $8 service contract only gives you access to the image, but you
aren't really allowed to use it, then why do Cisco offer that contact
in the first place ?
Support contracts give you access to all Cisco firmware.
So are you telling me
On Fri, 23 Apr 2004, Paul Tyreman wrote:
So are you telling me that to be legal, I need to pay $105, but could
get away with $8 ?
*IF* your phone qualifies for service contract (which is US$ 8), yes.
You still will have an illegal copy, and you can also be charged later for
all the software you
Andrea,
Here is a little patch for compiling chan_capi.0.3.1 with latest
asterisk CVS.
I could read in the lists that a new chan_capi.0.3.2 will soon arrive.
In the wait time you can use this patch.
put the patch in the chan_capi directory and tip:
# patch -p1
On Fri, 23 Apr 2004, Mark Olliver wrote:
I seam to have a problem working out how to get my X100P to answer after
1 ring. Currently it is working fine and connects to the switchboard
menu correctly but just does it after 4 rings, which I would prefer if
we could reduce.
Try this:
On Fri, Apr 23, 2004 at 09:11:48AM +0200, Dave Cotton said:
On Fri, 2004-04-23 at 08:43 +0200, Florian Overkamp wrote:
Thanks to this message where a virus chose to use my from-address to send
its crap from I am now being harassed with many many virus warning messages.
A call to anyone
Does anyone have a part number or know of anywhere in the UK that resells the image or
the license or both?
On Fri, 2004-04-23 at 03:12, Paul Tyreman wrote:
Why is there such a variation in price between what the two of you
have paid to get the SIP image for a 7960 phone ? $8 would be
Hello everyone,
I just like to let you know that I tested Asterisk with 3COM SIP phones
and it worked fine. The 3Com phones are old ones with the same look of
NBX 2102 phone but different product number: P/N: 655005001 Rev B
There is no special set up except that I have to specifically put
why not wisip? its size its like a regular cellphone and it uses wifi
Miguel Cavazos
On Fri, 2004-04-23 at 08:00, Chris Hirsch wrote:
Tim Sailer wrote:
Folks,
I'm looking for a SIP or IAX phone for field techs to take with them
when out on service calls. The regular desktop phones are
Hi all,
Here is a simple question. How can I know if a call
is in pass-thru mode, i.e. * is not in the media path???
Thanks.
Ben
I'm trying to correct the cid for
italy
because when arrive a call cid display the number
without the initial 0
and when i want to redial the missed call i can't
because the number is wrong
Thank's Tiziano
Greetings and salutations to all...
I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco
ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm
pretty sure my problem is in my configs as it feels like a stupid error and to prove
this to
On Fri, 23 Apr 2004 14:55:52 +0100, Mark Olliver wrote
Hi,
I seam to have a problem working out how to get my X100P to answer
after 1 ring. Currently it is working fine and connects to the
switchboard menu correctly but just does it after 4 rings, which I
would prefer if we could reduce.
I
I have three questions to ask about
this:
1) How do I know if my phone qualifies
for a service contrct ?
2) Where do I buy a service contract
from ?
3) How will Cisco know that I have
downloaded a image that I don't have a licence for ?
Thanks, Paul.
-Original Message-From:
interesting...
did you tried all the function?
ie, can you put a call on hold,
and more important do blind supervised transfer?
what about the prices? more or less, just to have an idea...
tnx, Matteo
Il ven, 2004-04-23 alle 17:08, Lisa Xie ha scritto:
Hello everyone,
I just like to let
why not wisip? its size its like a regular cellphone and it uses wifi
Because it sucks ass? Check the archives for some very valid gripes about the
device.
-A.
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Asterisk-Users mailing list
[EMAIL PROTECTED]
I have having problems trying to take a file recorded with Monitor and
convert it to MP3. When I use 'play' to play the .wav file, it sounds
fine. After bladenc'ing it, it plays at lightening speed, and the voices
are all high pitch. I tried using sox to resample to 32000 before
encoding, but that
Does anyone have a part number or know of anywhere in the UK that
resells the image or the license or both?
Matt,
I have tried www.cisilion.com/ for a price on the license, but so far
have not had a reply. These are the only place I have found to sell the
license.
The support contracts are a
If you do that, you'll have to carry around a wireless access point as well.
Nathan
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 23, 2004 10:08 AM
Subject: Re: [Asterisk-Users] smallest phone
why not wisip? its size its like a
Hi,
Would it be possible for you to provide some more
info on this.
I have just bought a Cisco 7960 on eBay, but only
now has the reality of needing a login to upgrade to SIP become
clear.
Can you tell me how you managed to get your phone
going on Asterisk without the image change ?
On 06:36 AM 4/23/2004, John Todd wrote:
Is it possible (ignoring Asterisk for the minute) for Polycom phones
to indicate visually (on the LCD or on a lighted extension button
or something) that a particular line is in use? I would expect this
method to be via NOTIFY or SUBSCRIBE calls from a SIP
On Fri, 23 Apr 2004 03:55:57 -0400, tmpm [EMAIL PROTECTED] wrote:
Might I humbly request someone, somewhere in the community establish a
dummies guide to asterisk kind of site, that explains in detail what the
cryptic scripts actually do, line by line.
The Wiki is helpful, but unless you were
While calling to H323 peer
*CLI
1:22:59.944 H225
Caller:81e5c48 assert.cxx(105)
PWLib Assertion fail: Invalid array element, file
/root/pwlib/include/ptlib/array.h, line 1183, Error=115
Abort, Core dump, Ignore?*CLI
*CLI show versionAsterisk CVS-04/22/04-23:56:01 built by [EMAIL
On Fri, 23 Apr 2004, Paul Tyreman wrote:
I have three questions to ask about this:
1) How do I know if my phone qualifies for a service contrct ?
When you (try to) buy your service contract, you will need to give the
model and serial number of the item you are trying to include into your
On Fri, 2004-04-23 at 10:33, Mike Machado wrote:
I have having problems trying to take a file recorded with Monitor and
convert it to MP3. When I use 'play' to play the .wav file, it sounds
fine. After bladenc'ing it, it plays at lightening speed, and the voices
are all high pitch. I tried
use lame
Il ven, 2004-04-23 alle 17:33, Mike Machado ha scritto:
I have having problems trying to take a file recorded with Monitor and
convert it to MP3. When I use 'play' to play the .wav file, it sounds
fine. After bladenc'ing it, it plays at lightening speed, and the voices
are all high
You've probably got callerID enabled in zapata.conf. That will cause a
wait of several rings whilst * looks for the caller ID info. Since this
only works in the US (or pkaces with similar phone systems), disabling it
in other territories saves the ring delay.
Make sure you have this in
Paul Tyreman wrote:
I have bough a cisco phone on eBay to use with Asterisk, but according
to that website, you need a contract with Cisco systems to upgrade the
phone to work with SIP.
I am guessing the phone that I get won't come with that as it was used
with the cisco call manager
All I can find on that Cisco website is
this:
http://www.cisco.com/pcgi-bin/cpn/cpn_match_result.pl?CurPosition=0Direction=ResultType=ECsearch_id=156576tab_name=findspcountry_id=GB
I can't see the likes of BT, O2, Vodaphone etc
wanting to deal with me !
-Original Message-From:
How i can obtain a complete caller ID from ISDN
zaphfc in italy
because i obtain a caller id without a initial 0
(for example cid=305001010 the correct number is 0305001010)
Thank's Tiziano
Hello,
I'm planning to convert my phone system to Asterisk, as I've outgrown my
TalkSwitch system. I have a few questions for experienced * users, most
of which can be answered yes/no.
Current Setup:
- Talkswitch 48NLS (4CO/8Ext) phone system.
- One CO line, two Vonage lines, one Voicepulse
You can do something like :
[incoming]
exten = s,1,Answer
exten = s,2,SetCallerID(0${CALLERID})
enten = s,3,
There is maybe a better way to do the samething.
At 18:40 23/04/2004, you wrote:
How i can obtain a complete caller ID from ISDN zaphfc in italy
because i obtain a caller id without a
When I do modeprobe wct1xxp I get it :
modprobe wct1xxp
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
create_proc_entry_R1b235e62
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
devfs_unregister_Re139a4b3
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
lame did the same thing. The reason I ask this on the asterisk list is
that .wav files I record from other sources encode just fine. I think
the hitch is the sample rates produced by asterisk.
File recorded by gnome sound recorder (lame/bladeenc encode just fine):
RIFF (little-endian) data, WAVE
I keep seeing the following errors in my asterisk logs:
Apr 23 12:13:36 WARNING[1226062640]: Exception flag set on
'SIP/Phone1-c016', but no exception handler
Apr 23 12:23:37 WARNING[1268026160]: You might not have the soxmix
installed and available in the path, please check.
The soxmix one
I have
this problem trying to talk to an ADDPAC gateway using oh323, when I call the
sound is great for the first 5 seconds then it goes almost silent... all you can
hear are some clicks every once in a while.
Anybody seen this can point me to some config settings
to change?
After patching and installing Festival, I am unable to get it to do
anything useful. I get the following error message on the * console when I
dial the test extension:
Parsing '/etc/asterisk/festival.conf': Found
Apr 23 13:43:06 WARNING[1226062640]: app_festival.c:382 festival_exec:
Strings do
Hi,
We have a machine with an *'s with Digium TDM400P and connected wit other
machine with *'s an TDM400P too. Well, I have a fax connected to each
machine, and the protocol in the middle is IAX2 alaw.
The fax between two fax, on in each machine, not work. The fax answer, but
error in comm.
On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote:
When I do modeprobe wct1xxp I get it :
modprobe wct1xxp
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
create_proc_entry_R1b235e62
snip
/lib/modules/2.4.18-386/misc/zaptel.o: insmod
/lib/modules/2.4.18-386/misc/zaptel.o failed
On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote:
When I do modeprobe wct1xxp I get it :
modprobe wct1xxp
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
create_proc_entry_R1b235e62
snip
/lib/modules/2.4.18-386/misc/zaptel.o: insmod
Try following the instructions at
http://www.voip-info.org/wiki-Polycom+Phones
I think you don't have your MACADDRESS.cfg file set right. I've never
used the web interface.
If it still doesn't work after that, write back.
John
P.S. Make sure you use a good xml editor when fixing up the cfg
Is this a new kernel? Did you recompile your modules under the new
kernel after making it?
John
Bartosz Jozwiak wrote:
When I do modeprobe wct1xxp I get it :
modprobe wct1xxp
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
create_proc_entry_R1b235e62
On Fri, 2004-04-23 at 13:04, Bartosz Jozwiak wrote:
On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote:
When I do modeprobe wct1xxp I get it :
modprobe wct1xxp
/lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol
create_proc_entry_R1b235e62
snip
Advance apologies for the length of this mail;
I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which
is currently working happily with an SDX Index phone system. I have to
replace this phone system shortly and I've been trying to get a * system
working for some weeks now. I have
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