On Mon, 2005-05-02 at 21:21 -0500, Matthew Boehm wrote:
Here is the current scenario:
Someone calls my DID. It comes in on PRI. I do Answer() then Wait(1) to
see if there is a fax tone. If there is, goto fax context and do the fax
thing.
If no fax tone, then Dial(SIP/myphone,30). If I
On Mon, 2005-05-02 at 12:40 -0700, Sean Kennedy wrote:
Adam Goryachev wrote:
On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote:
The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model)
phones can all do what he wants. ie, have multiple lines with blinking
red lights when a
On Tue, 3 May 2005, James Lin wrote:
I am trying to configure an E100P channel bank card.
The * will be connected to a PSTN switch with an E1 line.
I am a bit confused with signaling table of E100P Digium Cards.
The signaling table of E100P Digium cards is each 64K channel's ABCD
bits
trixter http://www.0xdecafbad.com wrote:
I am curious though about a companies competence when they have a
production system and it takes a week of multiple outages to chnage
something. You would think any professional company would have a test
and development network seperate from the production
I used to own a company which made fax switches and yes your devise must
answer the call to detect the fax CNG tone or wait till voicemail answers
and try to detect the CNG tone while the voicemail greeting is being played.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Pedro wrote:
What I did once was create an announcement that got played to the
receptionist announcing who the call was for based on the number that
was called. This allowed the receptionist to know which greeting to
recite.
How did you do this so that the calling party does not hear the
On Mon, 2005-05-02 at 16:54 -0400, Patrick M. Gray, Jr. wrote:
No errors, asterisk just immediately sends the other call to voicemail if
there
is already a call in progress.
Try turning on Call waiting on your cisco phone.
Cheers,
Joris
___
Hi!
I have configured:
iax.conf;
voicemail.conf
extensions.conf
everything works fine... the only things..
i do not receive any email notification when a
voicemail is left on the *.. any clues??? i think my
email server works(?).. In fact i am able to send an
email to the root (mail root
I upgraded to CVS Head last night to help fix my SCCP problems and now
my SIP installation is having issues. If I restart Asterisk, my SIP
phones may take up to an hour to register correctly so I can place calls
to them. They immediately go to voicemail as being busy. If I do a
sip reload I
Hi,
I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error
(EB113 on the display)
I am learning linux and asterisk as I go along, there might be obvious
things I should know, but bear with me.
From demsg below my 2 digium cards installed are listed (no config or
connections done
Note: forwarded message attached.
Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails !Créez votre Yahoo! Mail
---BeginMessage---
hello,
newasterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel
I would love to see this - we have about 15 snom phones in the building here.
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian
Stredicke
Sent: Tuesday, 3 May 2005 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Hmmm...
Firefox tries to open an external application but nothing happens
for IE the protocoll is unknown
Any hint?
Roman Zhovtulya wrote:
I think you should use the sip://name syntax.
I've wasted a lot of time before I figured it out myself.
Regards,
Roman
-Original
Hi,
have a setup which should not be unknown to others;
Asterisk behind wall doing NAT, and out in the wild world behind linksys
router a Polycom phone. The Polycom phone is on DMZ. It should register
with my server.
sip conf:
[4031]
type=friend
I setup my 7960 with line 1 as main, and 2 as a queue line. So if the
line is busy, asterisk queues the call and it will continue to ring on
line 2. Call waiting works too, but not as well as queueing...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I actually setup 6 registrations as separate lines. This allows me
dialout selection, like line 5 for teliax, line 6 for voipjet, etc.
I suspect you need different logons or all of your lines would ring at
once.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
So you have the receptionists voice right, then she goes for coffee and
someone else picks up, that would be odd... :)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Pratt
Sent: Monday, May 02, 2005 3:00 PM
To: Pedro; Asterisk Users Mailing
Hello,
I wanted to know if there is a way to dissplay infos from Asterisk on a
SIP phone ? Because I know Asterisk is very powerfull, so I'm nearly
sure that there is a way to do it.
Thank you for advance for considering my question
___
Asterisk-Users
Good morning,
did anyone try to use asterisk behind a AVM FRITZ!Box and utilize the
Fritz!Box as the connection gateway to ISDN?
Any and all information is greatly appreciated.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
Hi all,
Maybe someone has done this before, any help would be appreciated.
I am trying to dial with multiple zap channels. I want to dial with
all channels and if the called # is busy, continue to redial with all
channels, and then once 1 gets through hangup the others and
Hello,
Need your support for my following problem.
I am using Asterisk to switch calls from softphones to our internal
telephone network which is connected to the Asterisk server via an ISDN
Fritz card. If I build up a call from a softphone to our internal network,
voice is broadcasted only
On Tue, 2005-05-03 at 05:33 -0400, Ronan Eckelberry wrote:
Hi all,
Maybe someone has done this before, any help would be appreciated.
I am trying to dial with multiple zap channels. I want to dial with
all channels and if the called # is busy, continue to redial with all
Hi All,
I've encountered the following very weird behaviour:
1. I have an Asterisk box located on the net, which is connected via SIP
to two endpoints. First endpoint is a SIPUA SPA-841 and
the other is a VERAZ softswitch.
2. When tyring to run a call from the Sipua to the VERAZ, it
Sounds kind of like what I am looking for. I can prob do an AGI that
will do this for me so I don't have to mv the file in there all the
time. Thanks for the direction on this.
-Ronan
On Tue, 2005-05-03 at 02:52 -0700, trixter http://www.0xdecafbad.com
wrote:
On Tue, 2005-05-03 at 05:33
Hi,
is there any chance to bring Skype and Asterisk User together?
Regards,
Kib
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My server is located in Sweden. And as many European countries, we use a 0 to
indicate area codes, and 00 to indicate international calls.
And, when not having any leading 0, the call is a local call.
But when dialing out through Asterisk, I can't use leading zeros! I havn't
tried international
Chris A. Icide wrote:
-BEGIN PGP SIGNED MESSAGE-
On 09:50 PM 5/2/2005, Matthew Boehm wrote:
Hold up. So you have Phone #1. And all 6 lines register with the
username of
phone1 ?
And you have phone #2; and all 6 lines register with username of
phone2?
And the phone only registers
Matthew Boehm wrote:
No, you can use the same username and secret for all 6 lines
A 7940 or 7960 will just do the right thing
That right thing being, roll over the new call to the second line if
the
first is busy, etc.
There's no way. Asterisk doesn't support multiple logins from the same SIP
Hi,
I had similar problem with today cvs. When I make call gs phones
eachother no voice both direction.
I downgraded to yesterday cvs.
On Tue, 2005-05-03 at 12:01 +0200, Nir Simionovich wrote:
Hi All,
I've encountered the following very weird behaviour:
1. I have an Asterisk box
On Tue, 2005-05-03 at 06:16 -0400, Ronan Eckelberry wrote:
Sounds kind of like what I am looking for. I can prob do an AGI that
will do this for me so I don't have to mv the file in there all the
time. Thanks for the direction on this.
-Ronan
or a shell script since its simple enough.
My * seems to jump the gun when trying to dial out. It always dials the
1st digit b4 it even picks up the channel
-Ronan
On Tue, 2005-05-03 at 04:40 -0700, trixter http://www.0xdecafbad.com
wrote:
On Tue, 2005-05-03 at 06:16 -0400, Ronan Eckelberry wrote:
Sounds kind of like what I am
On Tue, 2005-05-03 at 07:54 -0400, Ronan Eckelberry wrote:
My * seems to jump the gun when trying to dial out. It always dials the
1st digit b4 it even picks up the channel
-Ronan
sounds like you may need a wait in there. Because I dont do much with
real lines I cant really help ya
Hi All.
Thanks¡¡¡, Now i can blind transfer calls with # key, but in
featuresmap said
blindxfer = #1; Blind transfer
disconnect = *0 ; Disconnect
automon = *1 ; One Touch Record
atxfer = *2 ; Attended transfer
but any key with *
My 7960 doesn't behave this way. With all usernames/display names/extensions
the same, a second incoming call goes directly to voicemail. I'm on SIP
firmware 7.1... could that be part of the problem or is it likely something in
*?
Thanks!
Pat
Quoting Henry Devito [EMAIL PROTECTED]:
Hi there
i have an asterisk box running ok, and now i am trying to integrate it
with my local analog pbx
So far, i have connected the fxo port of my * to an analog extension
port of my analog pbx.
As far as i know, if a call an extension of my analog pbx on a sip phone
( i have done the right
On 4/28/05, Eugenio De Vena [EMAIL PROTECTED] wrote:
Oh boy I am getting crazy...
I installed an asterisk with J4BRI ( 3 BRI point to point ) , Snom phones
and everything works fine. Where's
the problem? Well I can not get music on hold.. Well really MusicOnHold
works, works on Queue, works
Hi;
I have Cisco 7940 with version 3.2 sccp and want to upgrade it
to sipI got the firmware P0S3-07-4-00When the phone tries to
reach the tftp server gives conf. error and in status messages
It says CFG File Not foundIs it because of version
problem or something else? Does anybody have
On Tue, 2005-05-03 at 08:25 -0400, Patrick M. Gray, Jr. wrote:
My 7960 doesn't behave this way. With all usernames/display names/extensions
the same, a second incoming call goes directly to voicemail. I'm on SIP
firmware 7.1... could that be part of the problem or is it likely something in
I had to upgrade to version 6.3 first. I then was able to install 7.4 on
afterwards.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Betül Gözlükoglu
Sent: Tue 5/3/2005 8:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] cisco7940 upgrading problem
Hi;
I
Betl Gzlkolu wrote:
I have Cisco 7940 with version 3.2 sccp and want to upgrade it to sipI
got the firmware P0S3-07-4-00When the phone tries to reach the tftp
server gives conf. error and in status messages
It says CFG File Not foundIs it because of version problem or something
else? Does
Christian,
The current snom scheme is great for most applications and would probably
work reasonably well for this user. If you read the original post, he
indicates that he would be happy with a snom if he could make it work, and I
think this is the main issue with the snom 220 - getting this
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I can
call myself and talk to asterisk :)
I have defined an extension:
exten = _0.,1,Dial(Zap/0/${EXTEN:1})
exten = _0.,2,Congestion
exten = _0.,3,Hangup
So
Hi,
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from.
Please help, Thank you.
__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best
Hi,
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from.
Please help, Thank you.
Do you Yahoo!?
Yahoo! Small Business - Try our new resources site!
Tomasz Chmielewski wrote:
exten = _0.,1,Dial(Zap/0/${EXTEN:1})
set g0 instead of 0:
exten = _0.,1,Dial(Zap/g0/${EXTEN:1})
Deti
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To
Hi,
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from.
Please help, Thank you.
Do you Yahoo!?
Make Yahoo! your home page
___
Jason Williams wrote:
The release is Asterisk
1.0.6-BRIstuffed-0.2.0-RC7k
1.0.6 had a broken hold music you need 1.0.7. and then bristuff it
or get the newest bristuff, it's updated to 0.2.0-rc8a at the moment..
___
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Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I can
call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel. Zap channels start at 1.
Deti Fliegl wrote:
Tomasz Chmielewski wrote:
exten = _0.,1,Dial(Zap/0/${EXTEN:1})
set g0 instead of 0:
exten = _0.,1,Dial(Zap/g0/${EXTEN:1})
Yes, it changed something, now I get an immediate hangup:
-- Executing Dial(SIP/201-e124, Zap/g0/98) in new stack
-- Called g0/98
-- Channel
Joseph,
Thanks for your comments. I did have the phones in a group from some testing I
was doing, which I have since removed.
The phone reports call_waiting : 1 when I telnet in and check the config. Is
this the correct setting or should it be set to 0?
With the group config removed it still
I am a small customer myself and I have had some great service from
them. Granted, I have a had a couple of techs who were not particularly
polite or customer oriented in the past, however, they have been very
good with helping me. You have to realize these guys are growing so
fast and taking on
On 5/2/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote:
On Mon, 2005-05-02 at 00:06 -0400, skamp wrote:
asterisk runs great on BSD if you follow the sirections, and the card i
believe does work
That is debateable :) First which bsd?
[snip]
Here is the translation table
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel. Zap channels start at
Michael Welter wrote:
Pedro wrote:
What I did once was create an announcement that got played to the
receptionist announcing who the call was for based on the number that
was called. This allowed the receptionist to know which greeting to
recite.
How did you do this so that the calling party does
Julien Goodwin wrote:
Then why haven't you sent a backtrace? If I can see why it's crashing
then I can fix it.
Thanks,
Julien
chan_sccp project lead
The general consensus was that I needed to be running HEAD to make this
work properly. I upraded last night to HEAD and my SCCP stuff seems to
Daniel Nyström a écrit :
My server is located in Sweden. And as many European countries, we use a 0 to indicate area codes, and 00 to indicate international calls.
And, when not having any leading 0, the call is a local call.
But when dialing out through Asterisk, I can't use leading zeros! I
I found the necessary documents on www.voip-info.org and create
XMLDefault.cnf.xml it starts downloading but now it says Protocol Application
Invalid ...It tries periodically and gives that error...
Do you have any idea what should I do now?
-Original Message-
From: Kristof Hardy
Hi Michael,
you mean we should focus more on the usability (GUI) than the protocol
stuff? Maybe we should put the GUI programmer for a couple of days on
the receiptionists place and make sure he will have a lot of stress? :-)
Anyway, also for this usability stuff comments are welcome...
CS
Eric Wieling aka ManxPower wrote:
Correct. Asterisk does not support multiple logins from the same SIP
username. HOWEVER, if you configure the same SIP username/secret on
all 6 lines of a Cisco (or Polycom) the PHONE will NOT register using
the same username/secret more than once. It will
To help identify the source of the delays, I built a new system this
weekend from scratch. When that is complete, I'll use it to compare
the differences in motherboards, OS distro's, and maybe kernel versions.
Very good Rich, the results of that work will be very interesting.
And now for
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the
software and drivers, I am wondering if anybody knows where I could
downloaded from.
Have you tried http://www.mediatrix.com/ ?
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Asterisk-Users mailing list
Hi Marty -
The complaint from the users is that calls cut out, kinda like when
you have spotty cell coverage. Doesn't seem to matter whether the call
is incoming or outgoing, although it might be true that my users hear
the remote party cut out, while the remote party doesn't notice the
same
from
I'm trying to setup a multi-tenant configuration of * and have the
following question:
In extensions.conf, there is a [global] section that I would normally
use to define global variables for my single tenant setups. Now, is
there a way to have something like global variables on a per tenant
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain
the software and
drivers, I am wondering if
anybody knows where I could downloaded from.
The firmware is not openly available. Mediatrix approach is to charge
customers for every release they generate, and
I currently have Asterisk, an X100P and a Grandstream 100 for
playing around with. I'm interested in another IP phone for
daily use. The Grandstream is fine but feels a little cheap
and won't be acceptable by my wife (very important). What
I want is a phone with lots of buttons and an LCD screen
Rich Adamson wrote:
To help identify the source of the delays, I built a new system this
weekend from scratch. When that is complete, I'll use it to compare
the differences in motherboards, OS distro's, and maybe kernel versions.
Very good Rich, the results of that work will be very interesting.
Joseph wrote:
Is call waiting turned on?
What do you see if you telnet to the phone, and do a show config?
Are the lines all set the same?
Sorry to barge in like this, here it's working like you're telling.
If I do a show config, I see call_waiting: 1 and all the line's are set
to the same SIP
Betl Gzlkolu wrote:
I found the necessary documents on www.voip-info.org and create XMLDefault.cnf.xml it
starts downloading but now it says Protocol Application Invalid ...It tries
periodically and gives that error...
Do you have any idea what should I do now?
Can you post your
Neil Cherry wrote:
But the reality is that we need a phone that is not overly
complicated, handles 2 calls, caller ID info, reasonably
priced, not ugly looking (the 7920's are ugly) and not much
else. So a compromise to something in between is called for.
Neil,
The Cisco 7912G is a great phone or
- Original Message -
From: Neil Cherry [EMAIL PROTECTED]
What are your recommendations for a slightly fancy home phone?
Cisco 7940
Great phone. Great feel of quality (solid).
And you can put a logo on it.
___
Mobilcom
http://www.mobilcom.net
This winds up being very user specific. I love Polycom, others love
SNOM others love Cisco... Etc...
The Polycom IP300 is nice but only one row LED. IP500 is a great phone
and has 3 line LED.
Personally, I love the whole upright look and the phones have good
speaker phone.
$0.02.
Cheers,
Ah... the on going saga of upgrading Cisco phones to SIP ... have a look at
this
recent Cisco authored document:
http://www.tubby.org/cisco-voip/Cisco_7940_and_7960_firmware_upgrade_matrix.pdf
and see if you can get your head around it...
I'm planning on writing a Cisco phone upgrading HOWTO
I accidentally posted this on the biz list,
but it probably belongs here.
---
Folks,
Has anyone successfully got a Libretel DID forwarded
to a Voicepulse Connect number for incoming calls?
I have signed up for an overseas DID with Libretel. My
main VOIP number is a
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some reasons I have an issue with the quality. The voice is a bit
scratchy. I have tried iLBC and SPEEX, but it didn't make any
Hi all,
I am trying to make a call from an X-Pro with only the G.729 codec enabled to another with both G.711 and g.729. The Asterisk version is 1.0.3 and canreeinvite is set to Yes. What happens is that I got an 403 - Forbidden response and setting verbose 10 in Asterisk I can see the message:
Neil Cherry wrote:
What are your recommendations for a slightly fancy home phone?
The Sipura SPA-841 is a nice compromise between the Ciscos and the
Grandstreams.
-jbn
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sometime, * gives out tons of :
chan_oh323.c:2143 oh323_write: warning : OH323/L1648 invalid
frame size for G.729( 2 bytes)
chan_oh323.c:2143 oh323_write: warning : OH323/L1648 invalid frame
size for G.729( 12 bytes)
is there anything wrong ?
how to fix it?
Mario
Rich Adamson wrote:
Any suggestions on the above would be greatly appreciated!!!
Try using a non-redhat kernel. i.e. one from kernel.org.
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Hi Julio. It would be nice if you show the extensions.conf that
handles that kind of calls. You can do something like this:
[macro-analogpbx]
exten = s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes
from other Zap ch
exten = s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3,
I tried that and it didn't work. Then I decided to use a different phone
line. I had not thought about this before, it just didn't occur to me.
And everything worked fine. The phone line that doesn't work is my ADSL
line. Wall to splitter, one side going to ADSL router the other going
into a
Greetings to all!
Sorry for the numerous postings. but How could I slim my Asterisk PBX.
Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't
have any special hardware. Please, could I hope your various suggetions in
this regards. brief me your idea.
It seems this is a redundant question (or at least problem) within
this group, but I'm unable to find a solution/combination.
I have 3 30 button VIP phones running behind 3 different firewalls/servers (NAT)
My asterisk server is running great with a public IP address (no NAT)
The 30 button
Kumara Jayaweera wrote:
Sorry for the numerous postings. but How could I slim my Asterisk PBX.
Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't
have any special hardware. Please, could I hope your various suggetions in
this regards. brief me your idea.
See the Wiki under
Kumara Jayaweera wrote:
Greetings to all!
Sorry for the numerous postings. but How could I slim my Asterisk PBX.
Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't
have any special hardware. Please, could I hope your various suggetions in
this regards. brief me your idea.
[EMAIL PROTECTED] wrote:
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some reasons I have an issue with the quality. The voice is a bit
scratchy. I have tried iLBC and SPEEX,
Hi Claude. I just have giving some advices to someone with your same
problem. I assume you have the analog phone you want to call, behind
some AnalogPBX, then you have to call the analogPBX and tell him that
you want to call some analog extension. How?
Well, im just going to paste the same
TDM X100P card users:
Attached is a modified zaptel/zttest.c app called attest-mod.c. It
has been modified to report the delay in receiving 8,192 bytes
from the TDM card (instead of reporting a percentage). It works with
the digium x100p cards as well.
Drop the attachment in your zaptel
Mark Johnson wrote:
Julien Goodwin wrote:
Then why haven't you sent a backtrace? If I can see why it's crashing
then I can fix it.
Thanks,
Julien
chan_sccp project lead
The general consensus was that I needed to be running HEAD to make
this work properly. I upraded last night to HEAD and my
Thanks for all the opinions I have a lot of good examples now.
I do like the 7940G and I may borrow one from the lab and see
how well it goes. I'm also happy to have the rest of the opinions
and I've not eliminated the other vendors just slimmed it down
to fewer models. Lastly my wife informed me
WIKI is your friend.
http://www.voip-info.org/wiki-Asterisk+Slimming
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kumara
Jayaweera
Sent: Tuesday, May 03, 2005 11:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Light weight
Hi Marty -
The complaint from the users is that calls cut out,
kinda like when
you have spotty cell coverage. Doesn't seem to matter whether the
call is incoming or outgoing, although it might be true
that my users
hear the remote party cut out, while the remote party
doesn't
On Tue, 2005-05-03 at 10:02 +0200, list wrote:
Hi,
have a setup which should not be unknown to others;
Asterisk behind wall doing NAT, and out in the wild world behind linksys
router a Polycom phone. The Polycom phone is on DMZ. It should register
with my server.
sip conf:
[4031]
Hi Time,
yes, I did but nothing on their website
Cheers,
Ellafi
Time Bandit [EMAIL PROTECTED] wrote:
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from. Have you tried
Hello all,
Everyone has probably experienced this at some point in the past:
You pick up your analog phone. Rather than hearing dialtone, you are
connected with someone who has just called you. Neither you nor them
heard a ring.
Maybe it's just me, but it seems these freak incidents would
Rich Adamson wrote:
Any suggestions on the above would be greatly appreciated!!!
Try using a non-redhat kernel. i.e. one from kernel.org.
Now that I've got a distructable system, I'd like to try that.
Can you point me to some basic doc on how to do that? I'm rather
familiar with linux
Bill Coward wrote:
It seems this is a redundant question (or at least problem) within
this group, but I'm unable to find a solution/combination.
I have 3 30 button VIP phones running behind 3 different firewalls/servers (NAT)
My asterisk server is running great with a public IP address (no NAT)
Good day all
with the laster driver and latest drive asterisk I get these errors
Please help
May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes
19 z1 71 z2 36
May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes
21 z1 30 z2 121
May 3 17:43:15 pbxct
On May 3, 2005 11:25 am, Kumara Jayaweera wrote:
Sorry for the numerous postings. but How could I slim my Asterisk PBX.
Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't
have any special hardware. Please, could I hope your various suggetions in
this regards. brief me
Hi
Is there anyway of monitoring which extension is pressed on a IVR, I
need to use it for voting application.
Iqbal
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Guys,
I added some content to the Wiki on this feature. I don't think it's well
documented anywhere. Please expand upon what I put in there if you have
more details.
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx
-Corey
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Corey S. McFadden ([EMAIL PROTECTED])
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