-Users] Question
about remote POTS lines
I have a client who asked me about a situation they have. They
have a main office and 3 remote offices. We are installing an Asterisk
server at the main office with SIP phones in the remotes. Each remote
office only has 1 person. The remote offices currently
I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a
How remote are the remote offices? Miles? States? Countries? Best of my
knowledge, the days of exchanges based on proximity to a particular CO
are over, and those numbers (assuming they are in the same area code)
often can be routed anywhere. You could also look into having a company
like
How remote are the remote offices? Miles? States? Countries? Best of my
knowledge, the days of exchanges based on proximity to a particular CO
are over, and those numbers (assuming they are in the same area code)
often can be routed anywhere. You could also look into having a company
like
List -
Non-Commercial Discussion
Sent: Monday, November 15, 2004 11:57
PM
Subject: [Asterisk-Users] Question about
remote POTS lines
I have a client who asked me about a situation they have.
They have a main office and 3 remote offices. We are installing an
Asterisk server
People,
some people like pixelFriend and Jonathan Augenstine, send me references about
asterisk documentation, I read this and ok thanks :), my project is:
I have two office of construction: office_1 and office_2
I want connect the two offices by dedicated line, and connect the two analog PBX
On Thu, 28 Oct 2004 20:05:03 -0600, Olger Merlos Valverde
[EMAIL PROTECTED] wrote:
I have two office of construction: office_1 and office_2
I want connect the two offices by dedicated line, and connect the two analog
PBX
of this offices and transfer VoIP between two offices.
Ok one time
All,
newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for call disconnect? The purpose is to be able to make routing choices based on the
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for call
disconnect?
/path/to/asterisk/docs/README.variables
Pay
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at that
point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension calling
that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension
calling that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes.
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont think it is, it
should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
-Original Message-
From: joachim [mailto:[EMAIL PROTECTED]
Sent: Friday, October 22, 2004 6:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about ISDN reason codes
Are you sure this works ? (and does it work
Yes it works. It will go to priority 2 if the call was NOT ANSWERED for
any reason (busy, number not in service, etc). You may need to add ,,g
on the Dial line to get Asterisk to go to priority two if the CALLEE
hangs up.
I do not do post call processing if the CALLER hangs up.
joachim
Aha, oke :)
I was thinking of the answered statuses. That g was not working for me last
time i checked.
But so at least its working when a call did not get answered, thats already
good news for me.
Thanks a lot...
Joachim
At 05:23 22/10/2004, you wrote:
Yes it works. It will go to priority
On Fri, 22 Oct 2004, joachim wrote:
I was thinking of the answered statuses. That g was not working for me last
time i checked.
Can you post your Dial line (and preferably the lines after that as well)?
The 'g' option should work. It does for us, but we are a bit behind HEAD.
Peter
Hi,
I may be missing something here, but I don't really understand
how asterisk supposed to handle type=user.
Suppose I have the following config (mostly taken from default sip.conf.sample):
sip.conf:
context=sip ;default context for incoming calls
...
register = [EMAIL PROTECTED]
..
I have some quetions/ideas for the Asterisk Call Queues system.
System information:
- Fedora Core 1
- Kernel 2.4.22-1.2115.nptl
- Asterisk CVS-HEAD-09/08/04-17:43:15
1. I sould like it that if a user is in the que and the expected wait
time is longer then xxx seconds or there are more then xxx
I was looking at the wiki on 'Asterisk as a voice/fax switch'
And was wondering if the extension 'fax' is global to extensions.conf
Or just to the context it is in?
The reason I ask, is that my PRI might have 5 channels that will be
scrictly
Fax, and to be functional, I need multiple 'fax'
If you use extension dedicated to fax, then you don't need to use the
fax extenstion, but just call the rxfax application directly as you
would call the answer application
exten = 123456,1,rxfax(...)
But of course you may just use different fax extensions for different
contexts.
Regards,
Subject: Re: [Asterisk-Users] Question about the 'fax' extension
If you use extension dedicated to fax, then you don't need to use the
fax extenstion, but just call the rxfax application directly as you
would call the answer application
exten = 123456,1,rxfax(...)
But of course you may just
Message-
From: mstorck [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 3:51 PM
To: asterisk-users
Subject: Re: [Asterisk-Users] Question about the 'fax' extension
If you use extension dedicated to fax, then you don't need to use the
fax extenstion, but just call the rxfax application
if it detects a faxtone?
Paul Seniuk
-Original Message-
From: mstorck [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 3:51 PM
To: asterisk-users
Subject: Re: [Asterisk-Users] Question about the 'fax' extension
If you use extension dedicated to fax, then you don't need to use
Hello all,
I have a question concerning the calling number with an incoming PSTN call
through a E100P :
Here is what I see with a pri debug :
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
. Is there a way to verify its active or not?
Cheers,
Paul Seniuk
-Original Message-
From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED]
Sent: September 4, 2004 3:51 PM
To: asterisk-users
Subject: Re: [Asterisk-Users] Question on echo's for Canadian Asterisk
users ...
On Saturday 04
On Wednesday 08 September 2004 13:39, [EMAIL PROTECTED] wrote:
None of these were listed at all in the Makefile, so I added them
And tried a recompile. Still a bad echo. It is like the echo
cancellation
Is not even working. Is there a way to verify its active or not?
It's not in the makefile,
Yeah it looks to be the same setup as mine I am going to try out
Mark3 and the Aggressive Suppresor as well.
Paul Seniuk
-Original Message-
From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED]
Sent: September 8, 2004 12:34 PM
To: asterisk-users
Subject: Re: [Asterisk-Users
On Wednesday 08 September 2004 14:46, [EMAIL PROTECTED] wrote:
Yeah it looks to be the same setup as mine I am going to try out
Mark3 and the Aggressive Suppresor as well.
I'm using Mark2 and *no* agressive supressor (which surprised me, I thought I
had it in there)
-A.
Has anyone has issues with echo using a Wildcard with a PRI from a
major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom).
We are using a T1 from GT that is giving use annoying echos whenever a
SIP/IAX2 client calls a
local analog line. Calling cells phones is no issue since
On Saturday 04 September 2004 16:57, [EMAIL PROTECTED] wrote:
Has anyone has issues with echo using a Wildcard with a PRI from a
major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom).
I have a PRI with Bell Canada in Listowel, ON (519-291-).
I have echo on some calls but
Group
When I dial a phone number that should go out to the telco my local
phone rings. Does anyone have any hits ?
Thanks
Asterisk Ready.
*CLI -- Called g1/6144196143
Urgent handler
Urgent handler
-- Starting simple switch on 'Zap/2-1'
Urgent handler
Urgent handler
-- Called
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Question about dial out via Zap
Group
When I dial a phone number that should go out to the telco my local
phone rings. Does anyone have any hits ?
Thanks
Asterisk Ready.
*CLI -- Called g1/6144196143
Urgent handler
Urgent handler
M. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 23, 2004 7:32 AM
Subject: [Asterisk-Users] Question about dial out via Zap
Group
When I dial a phone number that should go out to the telco my local
phone rings. Does anyone have any hits ?
Thanks
Asterisk Ready.
*CLI
Dear all:
Doesanybody know is it possible to use the board TE405P with only one port configured as follow, or I have to use the 4 ports at the same time ?
Thanks,
Angel
ZAPTEL
span=1,1,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31
ZAPATA
[channels]context=menu-general
switchtype=euroisdn
You can use as many ports as you want. Just define how many ports
(spans) you're gonna use near the beginning of /etc/zaptel.conf.
span=1
Good luck
Bruno
Angel Diaz wrote:
Dear all:
Does anybody know is it possible to use the board TE405P with only
one port configured as follow, or I
On Wednesday 18 August 2004 10:31, Angel Diaz wrote:
Does anybody know is it possible to use the board TE405P with only one
port configured as follow, or I have to use the 4 ports at the same time ?
The four ports are independent of one another with one exception: they all
share the same
Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ?
Thanks,
Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16
-15,17-31 channel = 32-46,48-62
Kind RegardsClaus Futtrup
- Original Message -
From:
Angel
Diaz
To: [EMAIL PROTECTED]
Sent: Thursday, August 12, 2004 11:35
PM
Subject: [Asterisk-Users] Question about
TE405P
Hi all, Does somebody know how I have to setup my
: Thursday, August 12, 2004 2:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Question about TE405P
Hi all,
Does somebody know how I have to setup my TE405P ?
Is it correct my configuration below ? Otherwise, can somebody help me ?
Thanks,
Angel.
zaptel.conf
span=1,1,0,ccs,hdb3
span
:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Question about TE405P
Hi all,
Does somebody know how I have to setup my TE405P ?
Is it correct my configuration below ? Otherwise, can somebody help me ?
Thanks,
Angel.
zaptel.conf
span=1,1,0,ccs,hdb3
span=2,0,1,ccs,hdb3
span=3,0,1
HI all
I have a little question, and since there is a alot of Cisco Gurus
somebody might be able to help me.
I think It is an easy problem.
My PSTN proviver strips the first digit in the callerid on all incoming
calls.
So when the call reaches my Asterisk I am missing a 0 in the CLID
I
Hello,
I have a one-way audio problem. If any one can give me a clue on how to
solve it, I'd highly appreciate.
My configuration is:
Both Asterisk server and a SIP phone run within a LAN. Asterisk:
CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp
14262. The Linux box
On Wed, 21 Jul 2004, Michael Wang wrote:
How do I change configuration of Asterisk so that phone B can use
aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address?
sounds like * is using reinvite to get itself out of the loop and let the
phones send RTP directly between
Hello,
I have a one-way audio problem. If any one can give me a clue on how to
solve it, I'd highly appreciate.
My configuration is:
Both Asterisk server and a SIP phone run within a LAN. Asterisk:
CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp
14262. The Linux box
Michael Wang wrote:
Hello,
I have a one-way audio problem. If any one can give me a clue on how to
solve it, I'd highly appreciate.
My configuration is:
Both Asterisk server and a SIP phone run within a LAN. Asterisk:
CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp
Excuse for my bad English, but I have a problem after the compilation of
Asterisk-addons:
when I try to execute the simbolic link
ln -s /usr/lib/mysql/libmysqlclient.so /usr/lib/libmysqlclient.so
this produce an error because the source object libmysqlclient.so is not present in
any
I know this is a little off list but I can't think of a better place to
ask this question.
I upgrade the phone to 7.1 and it installed the Universal Application
Loader. Now I'm getting Protocol Application Invalid after it reads tftp
SIP(MAC).cnf
Any ideas?
Again sorry this is off topic
On 08/07/2004 at 08:21 Hall, Eric M. wrote:
I know this is a little off list but I can't think of a better place to
ask this question.
I upgrade the phone to 7.1 and it installed the Universal Application
Loader. Now I'm getting Protocol Application Invalid after it reads tftp
SIP(MAC).cnf
You the MAN!!! I drop the P of the P0S3-07-1-00 Everything is golden
now!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell
Sent: Thursday, July 08, 2004 9:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Question about Cisco IP Phone
I have2 X100P
card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area
codes to match mine.
When I try to dial
out I get
app_dial.c:554
dial_exec: Unable to create channel of type 'Zap'
A zap show channels
: [Asterisk-Users] Question
about x100P and zap
I have2 X100P card and configured everything based on
configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match mine.
When I try to dial out I get
app_dial.c:554
I did as you stated however I get the same error. Here is
my config file. Did I miss something?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about
:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about x100P and zap
Your $EXTENs need to
be changed to ${EXTEN}. Youll also need to include any substr #s within
the brackets (ie. ${EXTEN:1}).
-wade
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent
You might want to check to make sure your signaling is correct in
Zapata.conf and zaptel.conf. They should be fxs_ks and fxsks
respectively.
If you run zttool from your zaptel source directly, does it tell you
that you have an active and green/OK alarm X100P?
-wade
Title: Message
I was wondering if
this is possible. I have a situation where I am connecting to a third
party voicemail system from asterisk. I know this does not make since to
everyone, but it has to be this way. Basically - I have an application
that runs on the Asterisk system and when
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 25, 2004 5:30 AM
To: [EMAIL PROTECTED]
Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs
Send Asterisk-Users mailing list submissions to
[EMAIL PROTECTED]
Hello
I am very new in this area, just start reading about Asterisk and VoIP 2 days
ago. I am very interested in this product but not really getting good
information on. Will appreciate of someone an answer these question in detail
or direct me to right documents:
1. How to setup and use this
Question number one is overwhelming... How do you want to use this
product. That will dictate how you set it up.
In any case, there is extensive documentation on this, please go there
first and formulate more specific questions that we can help you with.
Try here for starts:
.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 10, 2004 11:22 AM
Subject: [Asterisk-Users] Question about Asterisk and its use
Hello
I am very new in this area, just start reading about Asterisk and VoIP 2
days
ago. I am very interested in this product
like the rest of us did.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 10, 2004 11:22 AM
Subject: [Asterisk-Users] Question about Asterisk and its use
Hello
I am very new in this area, just start reading about Asterisk and VoIP 2
days
ago. I am very
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
Hello,
Somebody has an example with all data loaded in the
base for prepaid?
or an example of a base that this
working?...
Thanks...
Julio
Somebody made run prepaid?...
- Original Message -
From:
Julio
To: [EMAIL PROTECTED]
Sent: Monday, April 19, 2004 2:38
PM
Subject: [Asterisk-Users] Question about
prepaid db
Hello,
Somebody has an example with all data loaded in
the base
Hello-
I am in the process of adding a new provider to my asterisk box (both
for outbound termination as well as inbound DID). They are going to be
delivering and receiving traffic via SIP only.
Now, in IAX via Voicepulse or others I know that I can simply have one
registration statement
: RE: [Asterisk-Users] question about CPU usage
I think Steve is referring to the following line:
export LD_ASSUME_KERNEL=2.4.1
If you put this in your command line before starting asterisk,
you will get
around the RH9 problem of leaving zombies when AGI processes quit. Other
than
Back in February I found * pinning the CPU (Slackware 9.1, Feb CVS). I
ran a strace and found that it was looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1)
I've had my asterisk running for a couple of weeks and just noticed that it
takes about 98% of the CPU time (Linux RH9). Is this what you would expect?
Is it just that the program is polling for things to do, calling sleep(0)
or something simlar so as to relinquish the machine but otherwise
On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote:
I've had my asterisk running for a couple of weeks and just noticed that it
takes about 98% of the CPU time (Linux RH9). Is this what you would expect?
Is it just that the program is polling for things to do, calling sleep(0)
or something simlar
PROTECTED] Behalf Of Steven
Critchfield
Sent: Monday, March 22, 2004 4:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] question about CPU usage
On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote:
I've had my asterisk running for a couple of weeks and just
noticed that it
takes about 98
PROTECTED] On Behalf Of Bill Hamlin
Sent: Monday, March 22, 2004 9:22 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] question about CPU usage
What is it about asterisk that makes this happen? My other
apps that wait
on a select take hardly any CPU time at all.
I didn't find anything like
Of Bill Hamlin
Sent: Monday, March 22, 2004 1:22 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] question about CPU usage
What is it about asterisk that makes this happen? My other apps that wait
on a select take hardly any CPU time at all.
I didn't find anything like ldassume using google
export LD_ASSUME_KERNEL=2.4.1
- Original Message -
From: Bill Hamlin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 22, 2004 4:22 PM
Subject: RE: [Asterisk-Users] question about CPU usage
What is it about asterisk that makes this happen? My other apps that wait
On Mon, 2004-03-22 at 15:22, Bill Hamlin wrote:
I didn't find anything like ldassume using google. Can you tell me more
about that?
It's in the RedHat 9 RELEASE NOTES.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the
Sent: Monday, March 22, 2004 4:36 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] question about CPU usage
I think Steve is referring to the following line:
export LD_ASSUME_KERNEL=2.4.1
If you put this in your command line before starting asterisk,
you will get
around the RH9
On Mon, Mar 22, 2004 at 03:49:29PM -0500, Bill Hamlin wrote:
I've had my asterisk running for a couple of weeks and just noticed that it
takes about 98% of the CPU time (Linux RH9). Is this what you would expect?
Is it just that the program is polling for things to do, calling sleep(0)
or
I've been told that MusicOnHold is *incredibly* picky about the mp3s
that it plays. I've experimented with the sample and a host of other
constant bitrate mp3s, and even some VBR ones, and I can't get any sort
of consistent workability. Even the sample doesn't work but maybe 10%
of the time.
Does anybody get any data in the 'Extension' column of the 'zap show
channels' output? I'm at a loss as to where it would be getting
any information to populate this column. I've looked in the sample
zapata.conf chan_zap.c. I've tried specifying extension=blah or
exten=blah in zapata.conf.
Thanks very much Michael.
It worked but only if I configure my cisco to use g711alaw.
If I config my cisco to use default g729r8 it created the below
Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible:
No path to translate from H323:9242(256) to H323:28967(8)
Feb 9
Anthony Law wrote:
Hi Gus,
Thanks for your reply. I have tried below and still didn't work.
exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]
and now asterisk gives out below error
Feb 6 16:12:41 WARNING[30740]: pbx.c:1773
Subject: RE: [Asterisk-Users] question for oh323 users
Hi, it seams to me that h.323 service on your cisco B could be down. You
see packets coming to this box, but did you activate h.323. Try telnet
192.168.1.3 1720 to see if it is running. If it is, then check to see
if you are allowing
Subject: Re: [Asterisk-Users] question for oh323 users
Hi,
Thanks for your reply. I am definite that my h323 is running on ciscoB
because the below scenario is working fine.
pstnciscoA-ciscoBpstn
I have also eliminated access-list problem because if my access-list is
applied
Hi Gus,
Thanks for your reply. I have tried below and still didn't work.
exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]
and now asterisk gives out below error
Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel
Hi,
I am trying to forward calls from one cisco gateway to another cisco gateway
using asterisk
cisco(5300)A 192.168.1.1
asterisk 192.168.1.2
cisco(5300)B 192.168.1.3
pstn --ciscoA-asterisk --ciscoB--pstn
I have the below in my extension.conf
[default]
exten =
*My apologies if this message is posted 3 times, I was trying to sent it to
the list once before I am a list-member, the second time before I was
approved.
Can anyone point me to some resources on using hunting lines with Asterisk?
Sales support of my telco have no idea what I am trying to do.
Title: Message
Hello
all. I know * doesn't directly support recording mp3 files, but I was
wondering if anyone has created an AGI to do it indirectly. Thanks in
advance.
B. J.
On Mon, 2004-01-05 at 10:36, B. J. Bomar wrote:
Hello all. I know * doesn't directly support recording mp3 files, but
I was wondering if anyone has created an AGI to do it indirectly.
Thanks in advance.
That should be fairly trivial depending on what you want to accomplish.
If all you want
Hi,
I just setup my * with digium. I started testing
voicemail first between atas, and i am not sure why it is not prompting me any
when the call is not answered or if busy.i only get continuous
ringback andthe following message:
asterisk*CLI --
Executing Dial("SIP/6882332-1697",
extensions.conf
[sip]
;ring
exten = 5104112978,1,Dial(SIP/5104112978,20,tr)
exten = 6882332,1,Dial(SIP/6882332,15,tr)
exten = ,1,Dial(SIP/,5,tr)
;unanswered
exten = 6882332,102,Voicemail,u6882332
exten = 5104112978,102,Voicemail,u5104112978
exten =
-Original Message-
From: Jess Magnaye [mailto:[EMAIL PROTECTED]
Sent: Mon 1/5/2004 4:28 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] question re voicemail
Hi,
I just setup my * with digium. I started testing
Hello,
I have a problem. When Idial to asterisk with
H323 I do not hear ringing applecation (phone rings but i do not hear ringing
tone in handset). I have tried with Cisco 2600 H323 and Quintum
H323.
But when I connect I can hear ringing appl. What
can be wrong? Configuration is wrong?
Larry Black wrote:
[hardware]
type=friend
callerid=Hardware Phone 5
secret=phone
echocancel=yes
host=dynamic
dtmfmode=rfc2833
context=sip
My standard config for GS phones on the same LAN as the Asterisk server is..
[hardware]
type = friend
callerid = Hardware Phone 5
secret = phone
host =
Hello all,
Have a few questions.
New to asterisk , just getting setup with
1 X100P and 2 TDM400p. Redhat 9
Hope I sent this to correct list
Setting up some Aastra/Vista Powertouch 350 phones.
Things work outofbox on ADSI programming, vmail
downloads, menus etc.
Question.
We've got one of the Budgetone phones here, and we can call from any SIP
phone, or an outside line TO this phone and the conversation sounds great for
bothways, not a bad delay, no echo problem, etc. But when we pick up the
Budgetone and dial an outside line or another SIP phone the person on
On Tue, Nov 18, 2003 at 05:29:25PM -0600, Wayne Black wrote:
We've got one of the Budgetone phones here, and we can call from any SIP
phone, or an outside line TO this phone and the conversation sounds great for
bothways, not a bad delay, no echo problem, etc. But when we pick up the
] On Behalf Of Walker
Haddock
Sent: Tuesday, November 18, 2003 6:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Question about incoming/outgoing
On Tue, Nov 18, 2003 at 05:29:25PM -0600, Wayne Black wrote:
We've got one of the Budgetone phones here, and we can call from any
SIP
phone
Hi,
Here is a general question, not applying to asterisk so much, but in
the application of asterisk. I have purchased a few IAX DID's through
VoicePulse and am interested in a service provider who has the ability
to provide me with one number (reliable, as I wish to publish), and the
Hi all.
I'm going to implement some large Asterisk based solution. Maybe 4-5 PCs with 1-2
E1/T1 trunks on each.
Because some of the traffic will be sended to external VoIP provider, i has some
questions
1. Which is the lowest bandwidth consuming codec in Asterisk, which is compatible with
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