On Thursday, March 02, 2006 11:47 AM Tomislav Parcina wrote:
sox: Failed reading fpm-calm-river.mp3: Do not understand format
type: mp3
Have I done anything wrong?
Well your sox does not understand mp3 since the support is not compiled in.
Compile your own suitable version of sox.
Hi All
I have an Asterisk box using a Sirrix card sitting between our PSTN and
an ISDN pbx. Calls from the PSTN are forwarded to the PBX ok.
Calls from the PBX are having problems - the digits being passed are
being garbled. The numbers from the PBX are totally incorrect and
sometimes too
Hello list,
We need to find the Asterisk/VoIP user from Costa Rica for small testing.
Please contact me off-list
Cheers,
Madhawa
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Hello friends,
Can I run two asterisks running simultaneously on the same machine? I want
one to run v1.0.2 for h323 ( which is an old and running production system )
and one for sip implementation. I wonder how it can be done since they will
want access to the same ports and ip addresses.
You could run a virtual machine. I'd try xen, uml, and vmware in that
order (vmware would be the easiest/quickest to setup, but is more of a
resource-hog than xen or uml). Assign a separate ip to the virtual
server, setup asterisk, and you're all set.
BTW, just curious but why can't you run one
Hi all,
I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump.
Error Text Start=
[res_crypto.so] = (Cryptographic
Hi,
Thanks for your replies.
I am going to have many DID's and I have to provide each of them this feature.
So I cannot solve this problem with a dedicated DID having G711. Is
there a way
to change codecs in the middle of the call? Please tell me what else can I do
here?
Quoting Darrick
Hi All,
I was able to insert some extensions in Mysql DB and use them successfully. In
Mysql extensions table the priority column is of type tinyint and when I give
's' value for it, it is not accepting that value as it takes only tinyints.
Please tell how can I make that column accept values
Hi all, I m in a trouble using festival voices in asterisk. I am not able to change the default male voice of festival. Although i downloaded the us1 female voice and it iw working good in festival's CLI but it is not coming when i am usinf Festival in asterisk. I changed the
is possible to define a parameter to, hangup the line on silent? or ping
dead or something?
because all line have busy after the pc hangup :(
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Hi
all,
I have a simple and
maybe also stupid question: if i'm in coversation on a Zap channel and the
remote party send me a DTMF, could I capture it?
Thanks
all
Giordano
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I use PICO (nano for CentOS). Works great.
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http://mail.yahoo.com
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I solved this issue by replacing the router (Netgear RP 614) with a
newer model (Netgear DG 834 B). It seems the old router had occasionally
problems to forward the UDP-ports to Asterisk. However, I'm glad
everything works now! Thank you for your help!
Regards, Lius
I'm experiencing a
Hi Sina,
a detailed list of the steps you took could help.
Did you follow the suggestions in README.udev, also
a 'make linux26' did some magic for me.
kr,
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender
Sent: maandag 6 maart 2006
asterisk tends to not work well with mp3's that have
ID3 tags
--- Zach A [EMAIL PROTECTED] wrote:
Hi,
The 3 MP3 files which are installed with asterisk,
what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
___
Johann wrote:
In Asterisk the Agent / Queue setup is kinda different than most
people may expect. You can use a Queue without using Agents and
Agents can be used without Queues. Agents however extend normal
channels with the ability to login/logout/pause that is not available
on
+++I am out of the office until Tuesday, March 7th attending training, I
will be returning calls and emails at that time+++
+++Thank You+++
Cory Andrews
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct - 716.250.3402
Oh No! Here we go again
Cory you should know better.
--- Cory Andrews [EMAIL PROTECTED] wrote:
+++I am out of the office until Tuesday, March 7th
attending training, I
will be returning calls and emails at that time+++
+++Thank You+++
Cory Andrews
++
Oh No! Here we go again
Cory you should know better.
--- Cory Andrews [EMAIL PROTECTED] wrote:
+++I am out of the office until Tuesday, March 7th
attending training, I
will be returning calls and emails at that time+++
+++Thank You+++
Cory Andrews
++
Hi all,
May I have to patch asterisk-1.2.x with this patch
http://bugs.digium.com/bug_view_page.php?bug_id=0002859
to configure an outbound sip proxy in sip.conf ?
Regards
Harry
___
On 00:07, Tue 07 Mar 06, Adrian Carter wrote:
Im just curious, How would one use 'agents' without a queue. Is this
what you are essentially doing using Local/XXX@ dial strings??
kindda.
Maybe an example makes it a bit more clear.
Say you have 10 desks, all with a phone on them.
Users dont have
Yeah, "Hot Desking" but ok.. if you'll indulge me further, why
would the likes of AMP use astdb to implement that combined with some
clunky macros?
Im after that exact solution, but have various issues on occasion with
the AMP implementation of 'user login/logoff'. I'd love for you to
I just sent an email to one of his coworkers to disable that stuff.
Rich
From: Dovid Bender [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6
kernel
Date: Mon, 6 Mar 2006 05:18:13 -0800 (PST)
To: Asterisk Users Mailing
billy wrote:
i have AAH connected to pstn via digium TDM01B
had been testing it on telewest line (UK cable company) with very
little issues.
now moved to a BT line and had several that i anticipated from
infomation on this list.
the one that has caught me out is low volume from the caller
Ive been trying for quite some time now to make hints work correctly, so
that I may use the BLF (busy lamp field) features of the Snom and
Grandstream models that support it.
My problem is NOT a subscription problem.
I have a running Asterisk system, everything is as it should be, hints are
in
Sorry for my ignorance but what are 'HINTS'?
Thanks
Mimmus
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the time.
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Message: 23
Date: Mon, 6 Mar 2006 10:12:02 -0300
From: Pablo Allietti [EMAIL
Hi friend,
I am running asterisk in production and it is being used by many people using
h323. I cannot afford to change all their configurations. Also, the newer
asterisk dosenot support inband for h323 properly. Thats why I want two
asterisks one for backward compatibility and one for sip
Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card. I
connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is
made to a wrong number.
Is there anybody who had the same problem and solved it?
Title: Re: Problem with libpri?
In addition, I have created a possibly larger dump of the issue, as below. Can someone help me determine what the problem is? Is there more information that I can provide? I am running libpri 1.2.2, zaptel 1.2.4 and asterisk 1.2.5:
gdb dump:
Program received
Try 'show hints' in the console...
Or read http://www.voip-info.org/wiki-Asterisk+standard+extensions
It's Asterisk way of knowing the state of a phone so that phones may
subscribe to this information and make small led light up if a phone is
busy, and flash if it's ringing.
// Per
Sorry for
Hi
Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk interface ?
I would like to send and receive calls from/to my asterisk extensions from PSTN by spa3000 fxo.
Thanks in advance.
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte
Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk
interface ?
I would like to send and receive calls from/to my asterisk extensions from
PSTN by spa3000 fxo.
Go to www.voxilla.com and look for a setup wizard. Also, lots of other
good references/user-experiences at
No, some IP 501's have the inline cable and some have the power jack.
-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 05, 2006 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom 501 power over
I have the following python AGI script.
I know it's been abstracted, but it's still pretty easy to see what's happening.
self.agi.channelAnswer()
self.agi.wait(1)
self.agi.execCmd(background,enter-conf-call-number,)
self.agi.execCmd(Read,confNum|||,)
I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the
call from 3254102 to 3254104. When I try and transfer the call, I get the
following on the Asterisk console.
Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised
transfer requested, but unable to
Hi there,
I did do both of those things, yes, but it's not necessary to do the udev
permisions and rules modifications is it, since the makefile appears to do
that for you. At least it did it for me; however, just to make sure I did it
manually as well.
I'll send the output of the script command
Hi group!
How to set language for queue?
I have several queue's. In every queue, agents speaks different language. I
need to announce queue-youarenext and similar on different languages.
This is what I have in my extensions.conf and it does set language, but when
calls enters queue, it doesn't
I was just thinking, about this..
Move your Polycom Power Injecting Patch cable (Black Cable with AC
Adapter Input) into the cabling closet. You could then infuse the power
at the cabling closet and then just use a standard patch cable to patch
the phone in.
You would be looking at a line loss
Here is the compilation process of zaptel
I did edit the makefile and uncommented the #ztdummy, although, after I did
that, I get the make error of ztdummy being defined more than once.
[EMAIL PROTECTED] src]# cd zaptel-1.2.4/
[EMAIL PROTECTED] zaptel-1.2.4]# make clean
Makefile:214: target
With pen in hand, Roberto Pereyra succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
Hi
Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk
interface ?
Here are a couple, although you may need to make some
I'm trying to figure out how to allow an extension to register more than once.
For instance, I have all of these 4 line IP phones that I use with Asterisk and
I would like to have a persons extension (say 101) ring at all four lines so
that if the person is on the phone they can take another
I have the following in extensions.conf:
exten = 1000,1,Meetme(|dMic|)
According to the 'show application meetme' docs:
'i' - announce user join/leave (new in Asterisk 1.2)
Well, when users join the conference, Asterisk records their name, but does not
broadcast it into the conference. I
Hello,
This question is probabely recurrent, i apologize, but i haven't found a
limpid explanation (for me) in mail list, google, and hum source
code):
When use the Command DIAL to ring a group, WHERE is stored the name of
the 'winner' who pick up the call ? ($variable = ?), and, step
I'm interested in developing a new channel driver for a thrid party
telephony card for Asterisk. Is there any official document that
explains how to do this? We've been looking the doc/channel.txt and
doc/modules.txt in the source, but that's not a very complete source of
info :)
Thanks a
Giordano Grandis wrote:
Hi all,
I have a simple and maybe also stupid question: if i'm in coversation on
a Zap channel and the remote party send me a DTMF, could I capture it?
Thanks all
*Giordano *
show application Read
--
Kristian Kielhofner
I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).
This coincides with the release of v8 firmware for all Cisco phones (and
for those of you running Sergio's chan_sccp v8 works fine)
The
Not without some dialplan magic. You could have the setgroup for every
call, then use groupcount to figure out how many.
On 3/5/06, Paul Hales [EMAIL PROTECTED] wrote:
Is there a variable to read to see how many calls are currently open?
(related to channel status?)
PaulH
Hi,
I am just curious, does anyone know if I can run Asterisk on the Mac? I've
read something that it should be possible, but cant find an eventual
download page or what is supported. And also if the Zaptel driver is
supported as well as Ztdummy.
Many thanks,
Christian
http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support
It works but it's bitchy as hell to run because of root issues in OSX. I
run it on my Mini. Zaptel is not supported. You have to use an external
gateway of some kind. Zaptel development support is stalled, most likely
I used quadBri Junghanns card and I config zaptel.conf: ZAPTEL.CONF loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 ZAPATA.CONF [channels] language=it
Anyone seen this? If not I guess I'll have to post it as a bug.
Extensions.conf has this:
exten = 123,1,Meetme(|dMic|)
I dial 123, and enter my conference number. Asterisk asks me to enter my name.
At this point I hang up. If I type at the Asterisk console 'meetme list 12345'
it shows that I
Hi,
I am using Polycom 501 and I came across a problem. As soon as I have
incominglimit=1 in sip.conf, which is necessary for buddy watching, I
cannot transfer calls. On the console it tells me:
Call from user '3052' rejected due to usage limit of 1. Can someone
please tell me how to get around
Douglas Garstang wrote:
I have the following in extensions.conf:
exten = 1000,1,Meetme(|dMic|)
According to the 'show application meetme' docs:
'i' - announce user join/leave (new in Asterisk 1.2)
I use:
exten = 4299,1,Meetme(|Msicp)
Seems to work ok for me. But, I don't use the
Why do you need to have to set incominglimit=1 for buddies to work? We've not
had that requirement.
Doug.
-Original Message-
From: rivy strauss [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 9:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Buddy watch?
Hi,
Hello,
when I try to call someone via Sip, the called phone just rings about 25
seconds.
Here's my Outgoing-Context:
snip
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s,1,Answer()
exten = s,2,Playback(invalid)
exten = s,3,Hangup()
exten = h,1,Hangup()
/snip
And here's a log that shows
On Mar 6, 2006, at 8:39 AM, Colin Anderson wrote:
http://www.voip-info.org/tiki-index.php?
page=Asterisk%20MacOSX%20Support
It works but it's bitchy as hell to run because of root issues in
OSX.
I wonder what the above root issues means?
I run it on my Mini. Zaptel is not supported. You
Hi Doug. I worked it out. I had commented out chan_zap.so in modules.conf as I
didn't think I needed it. It was doing weird stuff, including not playing the
participants joining. Weird.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 9:54 AM
I've seen a lot of IP501 and I've never seen one with a power jack.
According to Polycom they all use the cable.
Possibly it was an IP500? -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: Douglas
On Mar 6, 2006, at 6:46 AM, [EMAIL PROTECTED] wrote:
Hi friend,
I am running asterisk in production and it is being used by many
people using h323. I cannot afford to change all their configurations.
Also, the newer asterisk dosenot support inband for h323 properly.
Thats why I want two
On Mar 6, 2006, at 6:46 AM, Giorgio Incantalupo wrote:
Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card.
I connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is
made to a wrong number.
Hi,
While I am talking, if somebody call me, it is ringing at the background
and I cannot hear well current peer.
Is there anyway to cancel new call notify?
regards,
- bs
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While I'm asking about the Polycom ip500, the answers for all phones
where mic/handset/headset levels are adjustable would be of interest
to many I'm sure.
For the ip500, the default value for the handset seems to be
voice.gain.tx.analog.handset=3
I've noticed that echo all but goes away when
Hello!
i'm trying to set up transfer without using the respective
asterisk-function but with the built-in phone functions. my goal is to
have the first callleg billed to the caller and the second callleg to the
callee, who is responsible for the forward(and i can't bill a unknown
caller anyways)
Hi,
if I dial normal with the dial comman I have in my cdr file the peer-name as
source and the CALLERID (number and name) as I have set it in the dialplan.
Now Iam using call files and Iam using in the file for example:
Callerid: name 333
333 will be used for the field src AND the
Wilson Pickett wrote:
While I'm asking about the Polycom ip500, the answers for all phones
where mic/handset/headset levels are adjustable would be of interest
to many I'm sure.
For the ip500, the default value for the handset seems to be
voice.gain.tx.analog.handset=3
I've noticed that echo
I have a hardwareFXO/FXS which handle my voip calls, and they support
G723 internally. Asterisk hands off these calls just fine, and everything
works, as long as I don't wantPBX menues available... The
problem is, once I want it to return messages, it will only return them as
GSM... which
On 04-Mar-2006, Pete Barnwell wrote:
Emacs...
On Sat, 2006-03-04 at 01:35 +0100, adibar wrote:
Vim forever ;-)
http://unix.rulez.org/~calver/pictures/curves.jpg
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I'd like to set up a sort-of follow-me: on a call to a given extension,
I'd like to simultaneously call several different numbers, play them all a
prompt upon answering, and monitor for DTMF digit 1. I know how to get
Dial() to dial multiple numbers, and I know how to play prompts and
monitor for
All - I've a new system, that since it's been in production, has seen a
few issues, that look like they should be fixed by upgrading asterisk @
home to the latest version. I was curious if anybody out there can tell
me their experiences with this, and what to expect.
Thanks,
Rolf Brusletto
I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).
I have a service contract for my 7960 but I don't see 8.x SIP firmware for
it at
With pen in hand, Rolf Brusletto succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
All - I've a new system, that since it's been in production, has seen a
few issues, that look like they should be fixed by upgrading asterisk @
home
On Mon, 2006-03-06 at 12:38, Nabeel Jafferali wrote:
I have a service contract for my 7960 but I don't see 8.x SIP firmware for
it at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.
I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the
7960.
You have to
On 13:38, Mon 06 Mar 06, Nabeel Jafferali wrote:
I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).
I have a service contract for my 7960 but I don't see 8.x SIP firmware for
it
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I
Thanks Michiel. I haven't tried chan_sccp in awhile. This weekend, I installed 1.2.5 with the latest sccp. Asterisk no longer cores when the 12 SP, however, there is no audio in either direction. There is one way audio if I dialout from the device, but internal call to call does not work, nor does
On 14:23, Mon 06 Mar 06, Ryan Laginski wrote:
Thanks Michiel. I haven't tried chan_sccp in awhile. This weekend, I
installed 1.2.5 with the latest sccp. Asterisk no longer cores when the 12
That's good.
SP, however, there is no audio in either direction. There is one way audio
if I dialout
Hi,
I have developed a custom agi and connect to it by placing a call
through a sip phone. The agi issues the STREAM FILE command from a
number of places in code to play out prerecorded messages. The problem
is if the agi tries to play a file, using the STREAM FILE command,
after the caller has
Hi All,
Have you any idea to
configure Cisco 7970 with Asterisk. Please if any of you have the phone
configured, send me any instructions.
Thanks in advance.
Diego.
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On my 4-line IP phones I can have 4 simutaneous calls come in with only the 1
registration. When a second call comes in you push line 2 and Asterisk starts
music-on-hold on line 1. What kind of IP phones are you using.
I'm trying to figure out how to allow an extension to register more than
On 17:00, Mon 06 Mar 06, Diego Mariano Velo wrote:
Hi All,
Have you any idea to configure Cisco 7970 with Asterisk. Please if any
of you have the phone configured, send me any instructions.
SIP or SCCP ?
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key:
Hi,
I saw this problem mentioned before but the user appeared to be using
the MP3 software with asterisk. I am using the native music on hold
player in asterisk 1.2 and I too have a volume problem with music on
hold. Is this controllable through the 'indications.conf'? I know this
file
On 3/2/06, Matt Riddell [NZ] [EMAIL PROTECTED] wrote:
Matt wrote:
Yup.. that's the exact problem I'm having. I really can't explain
what happens. If I don't restart asterisk it seems to happen after
about 2 days. So I restart asterisk once a day at 3am. And it still
goes down about
On Tue, 7 Mar 2006, Julien Goodwin wrote:
I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).
This coincides with the release of v8 firmware for all Cisco phones (and
for those of you
I am getting this error from call manager (4.0) and asterisk 1.2.4
I have canreinvite=yes on the call manager setup.
I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I see
this entry from call manager...
What might be
Ok, so, we've got the 7970 SIP Firmware now, but their readme is a
little sparse... Anyone have any clue as to the upgrade procedure for a
non-ccm5 system? (i.e. asterisk ;))
Aaron
Julien Goodwin wrote:
I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with
OK.
I've got the COP SIP filehow do we use this thing on the 7970?
-Darren
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On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:
I am getting this error from call manager (4.0) and asterisk 1.2.4
I have canreinvite=yes on the call manager setup.
I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I
I have installed several hundred polycom's, and I have never seen a 500/501
with a power jack.
All with the inline cable, as you mention.
Of course, if someone can provide photo evidence I will stand corrected.
PaulH
- Original Message -
From: The VoIP Connection [EMAIL PROTECTED]
To:
Can you provide a photo of this?
I am interested in seeing it!
PaulH
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 07, 2006 2:13 AM
Subject: RE:
On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote:
I have installed several hundred polycom's, and I have never seen a
500/501
with a power jack. All with the inline cable, as you mention.
Of course, if someone can provide photo evidence I will stand corrected.
I think the confusion
Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp
kernel.
I've two x100p cards connected, only one card is reconigzed by asterisk.
02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
02:02.0 Ethernet controller: Davicom Semiconductor, Inc.
Totally correct - according to me at least.
PaulH
- Original Message -
From: Ken D'Ambrosio [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:
/ I am getting this error from call manager (4.0) and asterisk 1.2.4
//
// I have canreinvite=yes on the call manager setup.
//
// I can call into the asterisk box from call manager. THat seems to work.
// When I am calling out of the box using a
Maybe this can conclude the thread. This powering arrangement works
for me:
Netgear FS108 :: Polycom injector cable :: RJ45 coupler :: patch
cable :: Polycom 501
Some notes:
1. The Polycom injector cable should be plugged into a POE port on
the switch (the Netgear FS108 switch has both
tar zxfv *.cop
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 06, 2006 4:00 PM
Subject: Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970
Ok, so,
It's just a tarball, extract it
tar zxfv *.cop
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Darren Wright [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 06, 2006
On Mon, 2006-03-06 at 15:42, Jerry Geis wrote:
here is some of the output. I am no longer the to spcifically do sip
debug but this is what I have.
along with my sip.conf snip.
The call to extension 3726 never rings. so it never gets answered.
Are you sure your sip trunk and route
On Monday 27 February 2006 19:36, Joshua M Thompson wrote:
1.2.4 now. It was on 1.2.3 but upgrading asterisk and zaptel was the
first thing I tried when we noticed the problem this morning.
So you were on 1.2.3, it worked and you went to 1.2.4 and it didn't?
-A.
On Mon, 2006-03-06 at 15:59, Mailing List wrote:
tar zxfv *.cop
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 06, 2006 4:00 PM
Subject: Re:
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