. It allowed looking
at spans and the alarm state and bipolar errors and bit errors etc. You
could also see which channels were in use and look at signaling bits.
Anyhow this program sits in the zaptel directory. You may want to check
it out.
Don Pobanz
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incoming EM wink trunks delivered over a T1 and
are not having any issues. We are using Asterisk 1.4.18 with Zaptel
1.4.10.
Don Pobanz
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=yes in queues.conf and calls will fill in
as expected.
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search indicates
that these channel banks can deal with PRI in a drop and insert mode
only, not for termination. (I use Adtran channel banks which are not PRI
so I may be confused here).
Don Pobanz
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to be able to transfer outgoing calls ('T' option).
Don Pobanz
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but use 'AddQueueMember' and 'removeQueueMember'
instead of PauseQueueMember and UnpauseQueueMember.
Is there advantages to using pause/unpause versus add/remove?
Don Pobanz
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this but the wiki is confusing.
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
Can someone suggest an approach (preferably one currently in use)? We
are using Asterisk 1.4.17.
Don Pobanz
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From: nik600 on Tuesday, January 08, 2008 6:02 AM
I've connected some analogic phone to some fxs modules on an
analogic card.
I want to disable by default the call waiting sound.
In zapata.conf
Callwaiting = no
Don Pobanz
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Don Pobanz
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Godson Gera:
On Dec 26, 2007 11:36 PM, Don Pobanz wrote:
After upgrading from 1.2.x to 1.4.x call detail records
are not being written to
/var/log/asterisk/cdr-csv/Master.csv
In cdr_manager.conf I have
[general]
Enabled = yes
cdr_manager.conf
bank via a T1 card.
Don Pobanz
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. Could this be the issue?
Don Pobanz
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be limited like to 10 or even 100? Then even if a user does
something stupid like forwarding their calls to himself, it wouldn't
cause problems for others.
Don Pobanz
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system that I will be using for testing. If all goes well, we will move
to the 1.4 branch. I hope many others are doing the same so the
stability of 1.4 can be improved to the point where no one is concerned.
Thanks to all the developers for improving an already great product!
Don Pobanz
| || |
| A | | B | | C |
| || || |
|-| |-| |-|
I hope this helps.
Don Pobanz
Regards
Richard
like it should dial without the wait.
Could there be another part of your dial plan that starts with '16'? If
not have you reloaded extenions.conf either by restarting asterisk or
doing an 'extensions reload'?
Don Pobanz
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any problems before so I believe I am
being thorough with the upgrade procedures.
Again, has anything changed in how the zapata.conf is parsed? What about
extensions.conf when using #includes? I'm not sure what else to do. Does anyone
have any other suggestions?
Don Pobanz
to Zap or Zap to SIP or SIP to
SIP) worked fine. Outgoing calls worked fine.
Has something changed in the way configuration files are parsed? Where
does a context '(null)' come from.
Don Pobanz
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(A
talking to B) - how many PSTN lines would I need? I
think 2x.
That is correct.
Don Pobanz
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.
This was a little confusing for me also. A week or so ago, someone
pointed out that you need to include featuremap in your extensions.conf
in order to enable attended transfer, even though it appears in
features.conf. I guess blind transfer works regardless of if you include
featuremap or not.
Don Pobanz
Mandeep Singh Bhabha
Just add
include = featuremap
in extensions.conf
i think this should help.
This fixed the issue for me also. I did not realize that this was needed
to make these features work. It does not appear anywhere in
extensions.conf.sample for 1.2.18.
Don Pobanz
On Wed
/phone8SIP/phone9SIP/phone10SIP/ph
one11SIP/phone12,${SECS_TO_TIMEOUT}) ;Ring Sales Phones
exten = 567,n,Dial(SIP/phone1,${SECS_TO_TIMEOUT}) ;Send back
to Receptionist
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in distinguishing which pots line the
call came in on and want to make decisions based on that then you will
just put each line in a different context in zapata.conf. Then your dial
plan will dial phone1 for context1, phone2 for context2 and so on.
Don Pobanz
cable (1-1, 2-2, 4-4 and
5-5). For example, our local telco's 'network interface unit' to our
Digium T1 cards uses a straight through cable.
I hope this helps.
Don Pobanz
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|
| | | |
| || |
|__| |__|
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, then it would use T1 2.
This line from your Nortel system makes me think it is using an external
clock. If this is the case then you must have an atomic clock or some
other 'external' clock source at your location.
CLOK EXT
Don Pobanz
a hardware problem.
...
The telco told me that timing must be provided by us, but when I tried
that all hell broke loose.
Timing to a telco switch is *always* going to be provided by the telco
switch.
Don Pobanz
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the wink.
Also, I believe (but may be wrong) that the 'trunksgroups' section is
for ISDN. Switchtype is only for PRI ISDN so is not needed for inband
signaling.
Don Pobanz
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and USED
to plug into two opt. 11c but now one end is going to plug into an
asterisk
box.
Our local telco provides E M with wink so we use
signalling=em_w
in zapata.conf
As others have said, you will need to find out the details for your
setup.
Don Pobanz
When you have a bunch of analog phones that you want to
connect to asterisk, but those analog phones have no
transfer button, what are the options to allow the phones
to transfer a call?
Check out features.conf
You can specify key presses for things such as transfer.
Don Pobanz
Erick Perez
Don, I suppose that in order for this to work i need
canreinvite=no, right?
No! It doesn't matter what you have for 'canreinvite' since
'canreinvite' is a SIP attribute, not an analog phone attribute.
For analog phones, Asterisk will always be in the call path. :-)
--
Don
)
exten = s,5,BackGround(HU-welcome)
exten = s,6,BackGround(HU-welcome)
exten = t,1,Goto(Queue_main,4024631371,1)
include = desks
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time, just sending the
pre-encoded media to the SIP carrier.
You should be able to, though I couldn't tell you how.
Don Pobanz
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for the
first digit is the presence of dialtone. That was why
I was trying to reduce the dialtone volume.
Maybe this is not the correct solution. But hey, I need to
try something! The TE412P card with echo cancellers enabled
did not offer satisfactory results. (too much echo).
Don Pobanz
tones through cleanly.
Is there a way to decrease the dial tone volume?
Any other suggestions is appreciated!
Don Pobanz
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for outgoing calls. Would the Norstar pass this information to you?
and
Has anyone tried this? and if so do you forsee any problems i will run
into?
I have not done this exact thing but something similar. I don't foresee
any problems that can't be worked through.
Don Pobanz
.
exten = s,1,Voicemail(${CALLERIDNUM})
Don Pobanz
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'' between each line
or phone you want to dial. Something like:
exten = 145,1,Dial(Zap/g1/18005551212Zap/g1/18006663434)
see the wiki for details:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Don Pobanz
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is important!)
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/Asterisk+Zaptel+Installationview_comment_id=11286
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for a while!
Don Pobanz
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occasions. After
that, I got smarter and stopped doing anything with queues. ;) We will
implement our queues at a later date!
Don Pobanz
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Rich Adamson wrote:
For others to better understand the issues, did you install asterisk as
a distro or download v1.2 svn code and compile?
I downloaded the 1.2.9.1 release from the www.asterisk.org website and
compiled it.
If you installed source via svn, did you try make update to pick up
Jerry Geis wrote:
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=1,1,0,esf,b8zs
bchan=25-47
dchan=48
you don't have span 2 listed.
span=2,2,0,esf,b8zs
Don Pobanz
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immediate=no
immediate = dialtone
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.
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with
jacks. I use to work for a local phone company where we regularly did T1
installs and the only 48 we used was part of a rj48 jack. Thanks, for
not letting anything foolish get through!!! :)
Don Pobanz
-Original Message-n to a non-US dummy the following phrases I have
What is US48
Eric Bishop wrote:
Could someone explain to a non-US dummy the following phrases I have
What is US48?
I assume by US48 they mean RJ48 which is a 8 conductor modular jack with
signal from the phone company on 12 and signal to the phone company on
45.
Don Pobanz
that the Digium T1 cards use the
same algorithm for both hardware and software echo canceling so hardware
will only work better if your CPU is overloaded when doing software echo
canceling.
Don Pobanz
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(568B) has you have
pair 1 - conductors 5 4
pair 2 - conductors 1 2
pair 3 - conductors 3 6
pair 4 - conductors 7 8
http://www.lanshack.com/make-cat5E.aspx
Don Pobanz
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and if no one
picked up would then play the message 'wait-while-try-cell-phone' and
then would dial the cell phone.
Don Pobanz
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separate from the how discussion.
Don Pobanz
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if you have any erroneous files in the
/var/spool/asterisk/voicemail/default/xxx/INBOX directory.
Don Pobanz
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it is as Daniel said something to do
with MOH. I am running 1.2.6.
Don Pobanz
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Justin Tunney wrote:
Are you guys using native music on hold, or MP3 music on hold?
I believe I am using MP3. My musiconhold.conf file looks like this
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
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Thanks Matt Roth!
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channel and work down when sending
calls to them. This will minimize 'glare', when both parties try to
initiate a call on the same channels at the same time.
Don Pobanz
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is repeated.
The solution is to have one end of the circuit supply the clock and the
other end derive the clock from the incoming signal.
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analog lines to the phone company it will not work since only 1 A/D
conversion is allowed!
We aren't doing any IDSN. It ?may? be possible.
Don Pobanz
Nico Giefing
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modem communications, however this does not give the same dialup
speed connections as to an ISP. Sorry for any confusion.
Can Asterisk serve as an access server/gateway to the internet?
???
Please share your experience.
Thank you.
Andy
On 3/28/06, Don Pobanz [EMAIL PROTECTED] wrote:
Nico
Charles Marcus wrote:
Is Asterisk capable of allowing for the recording of calls on a per
extension basis?
Yes, I use
exten =
51,1,Set(CALLFILENAME=/var/log/calls/${EXTEN}-${CALLERIDNUM}-${TIMESTAMP})
exten = 51,n,Monitor(wav,${CALLFILENAME},m)
exten = 51,n,Dial(Zap/10)
Don Pobanz
/iax_NDSXS1-NDSXS2/)
where is your disa extension of your other * system.
Don Pobanz
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]
snip
exten = 784,1,macro(stdexten,${EXTEN},${PobanzD},124,8311385,4623687)
snip
[macro-stdexten]
exten = s,1,Background(transfer)
exten = s,2,Dial(${ARG2},18,t)
exten = s,3,Goto(s-${DIALSTATUS},1)
snip
Don Pobanz
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)
exten = 5551235,4,Voicemail,us5551235
exten = 5551235,104,Voicemail,b5551235
exten = o,1,dial(secretary2)
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Damon Estep wrote:
Thanks Matt,
PRI signalling means that calls and answered and dialed (aka signalled) by asterisk, the goal is to maintain the signalling between the two nortel boxes.
I have gathered that raw point to point circuit emulation is not possible on asterisk...
To connect
. If this is true then you will need fxoks in your pbx instead of
the fxsks.
Don Pobanz
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and entering 3 invalid Authenticate values,
I get the congestion tone. I would expect to
hear 'one zero three'.
I am running SVN-branch-1.2-r7231 which was
downloaded on November 30, 2005.
Is this a bug?
Don Pobanz
aki toku wrote:
I'm Japanese. Sorry,English is not so understood,Please let me
Dan Elder wrote:
Is there a setting other than fxsks,fxsgs,fxsls that I should use for these
lines? (i.e. something like fxsem?)
Try the following in your zaptel.conf
span=1,1,0,esf,b8zs
em=1-24
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)
There isn’t any activity on this PRI (that I’m aware of) so I don’t
think it’s truly congested.
Has the telco 'turned up' the trunks on the PRI? If not, you would have
zero trunks available and so would always get congestion.
Don Pobanz
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into
voicemail for that user, the voicemail box will be created.
Don Pobanz
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just a few minutes to cut over.
We do not have voicemail on our backup server, but since our upgrades
happen after hours it has not been a problem. We have a channel bank
dedicated for testing any changes. It seems to work for us.
Don Pobanz
in
this context begin with a different digit.
Only have
1 option 1
2 option 2
3 option 3
and not additional lines like
301 extension 301
Extensions in the same context would begin with a 4, 5, 6, 7, 8, 9.
Don Pobanz
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,esf,b8zs
em=73-96
This says use the timing from span 1 if span 1 is up, span 2 if span 1
is down, span 3 if span 1 2 are down, span 4 if spans 1,2,3 are down.
Don Pobanz
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be able to trace their timing back to a stratum 1 clock
(very accurate clock). Two stratum 1 clocks will look like they are
timed together even if they aren't. If the telcos can trace their timing
back to a stratum one clock, you won't have any timing problems with them.
Don Pobanz
Ps. When I
3.x is the minimum needed to compile
asterisk. add this to the fact that before last week, gcc 2.95 happily
compiled asterisk without problems.
That would explain it. I am using gcc 2.96.
Thanks for getting me pointed in the right direction.
Don Pobanz
in this
function)
chan_agent.c:1864: `filename' undeclared (first use in this function)
make[1]: *** [chan_agent.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
Any ideas?
Don Pobanz
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. They have a number
to dial to program the ADSI menus. This number only needs to be dialed
once. After that things are pretty snappy.
I will soon start work on converting the rest of our company to
primarily ADSI phones.
Don Pobanz
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be to prefix with a digit instead of suffix with an
*. For us, all of our extensions are three digits and begin with a 5
or a 6 (5xx or 6xx). To transfer to voice mail we stick an eight in
front of the extension (85xx or 86xx). It works well for us.
Don Pobanz
with numbers that begin with
82xx or 83xx.
exten = _85xx,1,Voicemail(u${EXTEN:1})
exten = _86xx,1,Voicemail(u${EXTEN:1})
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http
of between dial
arguments. try
exten = _06.,1,Dial(IAX2/X/${EXTEN},30,rSIP/[EMAIL PROTECTED])
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to be better organized.
Don Pobanz
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Victoria Alexandru wrote:
Thanks anyone, I found the problem in rhconfig.h.
After the fix I successfully compiled zaptel.
V.
I also am trying to compile Zaptel on Mandrake 10.2beta3. I have seen
the same errors you were. What did you change in rhconfig.h?
Don Pobanz
Adding any info to the wiki
would
cause a reduction of power output.
I know this isn't a full answer but
Don Pobanz
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Jason Kawakami wrote:
So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that
should match NXXNX. Right?
wrong!
N is 2 to 9, not 1 to 9, so these are not the same. Try
[2-9][0-9][0-9][2-9][0-9]*
Don Pobanz
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the phone company will not be able to loop it. When I have talked
with the phone company I just tell them that there is not a CSU.
If CSU functionality is required, it would require a change to the driver.
Don Pobanz
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made
me figure it had to be one of my configurations.
Don Pobanz
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) or to a
Grandstream sip phone.
So, I do not know whether this is a Sipura 2000 problem or an *
problem.
Does anyone have any light to shine on the subject.
Asterisk CVS-HEAD-06/14/04-09:03:15 built by [EMAIL PROTECTED]
on a i686 running Linux
Sipura 2000 software version 1.0.33
Don Pobanz
,Voicemailmain(${CALLERIDNUM:6})
Don Pobanz
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is needed. Just plug the T1 into your Digium card and configure
your zaptel.conf and zapata.conf files accordingly.
If the trunks are analog, you would need a channel bank capable of
converting EM trunks to a T1 and plugging this into your Digium card.
Don Pobanz
-Tilghman
this noise?
Sorry I can't really help here.
Jim
Don Pobanz
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not know what 'Linux-style subscription license' means. Is it GPL?
I do know that not all open source is not the same.
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seeing is that the DNIS info is not being passed through to
asterisk. Since I get no DNIS, it shoves the call to my s
extension.
Have you verified that immediate = no in zapata.conf? If not, then *
may not be waiting for the digits before trying to find a match.
Don Pobanz
now, asterisk could not do
this.
It would probably be better to explore trunking technologies such as
low bandwidth codecs over IAX2.
Don Pobanz
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On Thursday, January 15, 2004 10:42 AM, Iain Stevenson
[SMTP:[EMAIL PROTECTED] wrote:
...
Is there any way to stop * even considering an
incoming
call on a line as a fax call?
Sure, just don't have
exten = fax.
in the same context (or included context).
Iain
--
Don Pobanz
be:
span=2,1,0,esf,b8zs
Does this happen at the same time every day? If so it does not sound
like a timing issue. If at random times, it could be.
Daniel
Don Pobanz
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) ? Will I run into problems there? I don't forsee it but I
also
didn't forsee the problem being discussed in this thread...
Yes it is possible to receive clock from one span and provide it for
the other three. That is how I am running.
Regards,
Andrew
Don Pobanz
. There have been some in the past who have had
problems getting PRI ISDN to stay up and it was due to clocking.
Don Pobanz
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to provide timing and the other
end derive timing. (The only exception is if the equipment on both ends
are tied to stratum 1 clocks. However, I would guess that this does not
apply to any of us on this list).
Don Pobanz
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the number of trunks we have.
I know this didn't exactly address your questions. For your primary
question I believe that your would need different type of channels in a
channel bank than FXOs. DPTs (Dial pulse) terminating come to mind, but
that may be wrong.
Don Pobanz
so I think my only
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