On every transfer the recording starts again at the moment. I
thought the "a" option would simply append to the file and keep
going. Any ideas?
Add an ExecIf that returns early if MONITOR_FILE already has a filename.
--
Trevor Peirce
AcroVoice Solutions Inc
www.a
this because, even though the number has been made available to
you, it's marked as a blocked call. Your server is honoring this and
blocking the number when it dials the next server. By using Remote
Party ID, you'll be able to carry this information forward to your next
server.
Regards,
--
Trevor
aster...@opensourcesolution.in wrote:
Not able to compile asterisk,zaptel,libpri in /usr/src
Have you tried to run make?
Without any information on what you're tried and what error you receive,
I can almost guarantee you will not receive any help on this forum.
Regards,
--
Trevor Peirce
Matt Darnell wrote:
Both their toll free and fax numbers go to a re-order message...seems
like the worst.
These days if you have your PBX, DNS, web, and email all hosted on the
same server, it doesn't take much to have your entire business appear to
be gone.
--[ UxBoD ]-- wrote:
Please how do I stop the following ???
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed
You figure out why asterisk is crashing. :)
This has nothing to do with mpg123, which is just an
for T.38 to work properly.
--
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Digital Conceptions Canada
http://www.digitalcon.ca
1-888-606-3030 / 250-391-7822
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,
--
Trevor Peirce
Digital Conceptions Canada
http://www.digitalcon.ca
1-888-606-3030 / 250-391-7822
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for caring :)
--
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Digital Conceptions Canada
http://www.digitalcon.ca
1-888-606-3030 / 250-391-7822
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. However,
this is my understanding from my previous experience with looking up
Caller Name information via CNAM/LIDB/SS7.
Regards,
--
Trevor Peirce
Digital Conceptions Canada
http://www.digitalcon.ca
1-888-606-3030 / 250 483-0386
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to make it feasible.
In Canada, we just include the name in the SS7 signaling on a per-call
basis and bypass this whole mess :)
Best regards,
--
Trevor Peirce
Digital Conceptions Canada
http://www.digitalcon.ca
1-888-606-3030 / 250 483-0386
,
--
Trevor Peirce
Digital Conceptions Canada
http://www.digitalcon.ca
1-888-606-3030 / 250 483-0386
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Michael wrote:
I want to obtained the called number (aka DID) as a variable. How is this
done?
(Upline connectivity is via SIP provider).
With verbose = 100, the called number is clearly shown as
follows Wait(SIP/[called number]-64e6, 1) for example, but I can't find
the info as to how
Sriram wrote:
2. Is there any way to block the caller id from appearing on the SIP
Phone ? my trunk is E1 PRI while i used softphones internally - i
dont want my agents to see the caller id - is their any way to block
caller ids from appearing on softphones ?
a) SetCallerPres(restircted)
Gondar Monn wrote:
Hi there!
Does anyone deal with Telus in BC ? We have some PRI lines that were
used for dialup, would like to convert them for pbx system, talked
with some technicians @ Telus, but the information given was not
clear, kind of: try this see if it works Does anyone
Sebastian wrote:
Hi,
I’m noticing MixMonitor records 5 seconds aprox less of a call.
The recording is iniciated via Queue and ends at the hungup.
(gsm format), when I listen to the audio file, has 5 seconds missing
at the end of the call.
I'm not sure if this is relevant, but in
sean darcy wrote:
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()
But it doesn't work. CALLERID(name) isn't changed:
Perhaps try this:
Paul Douglas Franklin wrote:
When calling out to another phone, they always identify themselves
correctly. But sometimes they will respond to the wrong incoming
calls. (By respond, I mean that the phone rings and if someone picks up
the receiver, the call then goes thru.) For example,
Tom Moore wrote:
Hi guys,
Does the Aastra line of phones work with dns srv records?
I'm trying to get my 8133i to do this and in the settings it asks for ip
addresses of registration and proxy servers.
Does this mean that it will not just let me put the domain name in like
other devices I
Joseph wrote:
Does anybody have an idea how to pass Off Hook caller ID to Asterisk via
Linksys ?
I'm getting caller ID type I OK but when another customer rings the phone
(when I'm on line) the CID off hook is not coming through.
I think Off-Hook CID is called CID type II, isn't it?
Joseph wrote:
So I don't understand, what is the point of setting timer on:
VMWI Refresh Intvl:
since it doesn't get into effect until Register Expires
This option tells the device how often to update the phone's VISUAL mail
waiting indicator. As in that icon or red flashing light on the
Tzafrir Cohen wrote:
On Tue, Jul 08, 2008 at 09:34:44PM -0700, Trevor Peirce wrote:
I was recently introduced to fail2ban. It's a nice tool that will watch
log files and when it notices too many failed authentication attempts
(SSH, FTP, Password protected web sites, asterisk) it will run
Steve Totaro wrote:
For security, how about an authentication retry setting in the sip
configuration? After X amounts of failed auth or registration
attempts, block IP for Y amount of time. It would seem fairly easy to
do using realtime with DB entries for IP blocks and expiration. Then
a
Joseph L. Casale wrote:
I'm in Alberta, thanks for the clarification. Did you guys get a Whitepages
listing by chance?
I am contacting Superpages now.
Yes, in BC it's Superpages who publishes the phone book. If I recall
their charge includes both a white and yellow pages listing.
Trevor
Joseph L. Casale wrote:
So my SIP Provider states they do not offer the service to list my numbers w/
the Whitepages.
We phoned the Whitepages and they said we can't do it, the SIP Provider must?
Either one/both of them is/are useless or I must switch SIP providers to one
that can get
undef
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Mike Hardman wrote:
Is there any way
I can tell if a call is a diversion from an external phone if it comes
in on our PRI? If so, is there also any way I can find out what number
the call was diverted from? I've done some logging with PRI intense
debug; and I cant seem to see anything
asterisk rather than direct to the
endpoint.
Thanks,
Trevor Peirce
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a bug should be
filed for this if you can reproduce it reliably...
Thanks,
Trevor Peirce
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should block that.
Did something change around the release of 1.4.18 that would have
changed the behaviour? I thought that when ChanSpy, MixMonitor, and the
like are enabled on a channel it would be prevented from reinviting the
audio to bypass asterisk.
Thanks,
Trevor Peirce
candidate.
HTH,
Trevor Peirce
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Mindaugas Kezys wrote:
Hello,
Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
Yes I also saw this and had to revert. Calls to the IVR seemed to be
fine, but as soon as two peers call each other it crashes as the call
progresses (never connects). I haven't had a chance to
on your line you might
be able to accomplish this. I have never tried to have asterisk
recognize distinctive rings and this wiki page doesn't look too
promising, but take a look:
http://www.voip-info.org/wiki-Asterisk+ZAP+channels#DetectingDistinctiveRingonIncomingCalls
Best regards,
Trevor
this, but I just wanted to respond that you're not the only one being
bothered by it.
Best regards,
Trevor Peirce
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asterisk
Fons van der Beek wrote:
I've overwritten the indications.conf with the one from the
sourcecode, stil no luck
Perhaps somebody knows what the correct value for indications.conf is
when using the dutch xs4all as sip carrier??
A simple way for you to test your indications.conf as far as the
shadowym wrote:
I guess someone has to say it.
Have you considered Aastra?
You can argue about quality/features/functionality but I have set up
both and the Aastra are definitely easier to configure and they reboot
quicker.
Nobody ever complains about the quality of sound or
Fons van der Beek wrote:
After implementing the described test for indications.conf
The CLI outputted:
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in
new stack
-- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488,
ring) in new stack
--
Gleim, Jason wrote:
second T1 (Zap/g1). We have a number of DIDs that come in on that T1 and
I need them all transparently bridged for the time being.
[custom-nortel]
exten = _N.,1,Dial(Zap/g1/${EXTEN})
Anything that comes in will go right back out again.
Best regards,
Trevor Peirce
Nitesh Divecha wrote:
Everything is working fine but the only problem is voice mail greetings
for Busy and Unavailable is not played. By default only Temp Greetings
voice mail greetings is played. I am passing the correct parameters for
Busy = 'b', Unavailable = 'u' and default goes to Not
Yes I had to whitelist their mail servers because their reverse DNS
disappeared sometime last week...
Armin Schindler wrote:
Hello,
sorry for beeing off-topic here. But can anyone confirm that
there is a problem reverse resolving lists.digium.com (216.207.245.17) ?
Because of this problem,
Abdul wrote:
routes.pl
$dgw = 'SIP/5556'; #A-Z carrier
$opt = 'L(6:1)';
$AGI-set_variable(routecall-destination, $dgw);
$AGI-set_variable(routecall-args, $opt);
Extnenitons.conf
[testwell]
exten = _x.,1,Set(TIMEOUT(absolute)=3660)
exten = _x.,2,AGI(routes.pl)
exten =
(6:1)
Best regards,
Trevor Peirce
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list and asking for assistance there.
Best regards,
Trevor Peirce
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Anthony Chapellier wrote:
Sorry, I'm doing a mail test since I was not able to send any mails to
the mailing list for about a week...
Tell me about it. I've just given up on numerous posts because they'd
vanish into cyberspace. Doubt this will show up since now even replies
are being
Andrew Ott wrote:
ZAPATA.conf
===
;911 group
group = 2
restrictcid=yes
signalling = e911
channel = 25-26
===
...
I've tried it with either one of those ${EXTEN} which just does 911, and the
${CALLERID(ani)} both have the same result no number
Hello,
Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and
HPEC 9.00.003?
In particular, with a hardware configuration similar to:
Module 0: Installed -- AUTO FXO (FCC mode)
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Not
Matthew Fredrickson wrote:
Trevor Peirce wrote:
Hello,
Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and
HPEC 9.00.003?
First of all... Why are you using zaptel 1.4.5 with asterisk 1.2? That
is a red flag in itself.
References:
http
Steve Totaro wrote:
Just to make sure you saw this reply quoted from Tzafrir, seems to
really know his stuff:
/
If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll
need: ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h ln -s
zaptel/tonezone.h /usr/include/tonezone.h
Tzafrir Cohen wrote:
If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll
need:
ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h
ln -s zaptel/tonezone.h /usr/include/tonezone.h
Note: I'm not sure exactly what happens if you run a 'make install' of
zaptel 1.2 on top
John C. Wolosuk Jr. wrote:
ok this is a wired problem. when i use X-Lite - after i register with
asterisk X-lite sends a subscribe/notify request to asterisk to
determine if the account has any messages waiting.
if i create a sip.conf account using:
user 12345 with a voicemail box 12345 -
voiplist wrote:
I have always been able to block toll-free numbers by catching them
with a line similar to this for each DID I have on my system:
exten = 5554441212/_888NXX,n,Playback(GoAway)
Where 15554441212 is one of the DIDs that rings into our Asterisk box.
The problem with this
Trevor Peirce wrote:
[macro-blocktollfree]
exten = s,1,GotoIf($[${MACRO_EXTEN:3} = 800]?goaway)
exten = s,n,GotoIf($[${MACRO_EXTEN:3} = 888]?goaway)
You'll probably have better results using ${CALLERID(num)} here...
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voiplist wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
This
OCOSA ListAcct wrote:
I apologize if this question has already been answered / asked. I was
searching on Google and nothing I do will work. All that happens is that
the phones ring for 00:01:15 then voicemail kicks in.
I wonder if this is your phone deciding it has been ringing for long
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
reply. The reply showed up but the new thread still seems to be MIA. No
bounce or anything (and I have no filtering on this account). Weird...
Maybe I'll
C F wrote:
This is the postmaster at the list and I am notifying you that your
message failed.
Over the past two days my new posts seem to have silently been dropped.
I wonder if I can reply to an existing thread...
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,Wait(1)
exten = s,n,Answer
exten = s,n,Background(opengreeting)
exten = s,n,Dial(SIP/ht1SIP/gxp3,20)
Regards,
Trevor Peirce
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Zeeshan Zakaria wrote:
I've got a 778 DID for vancouver, but don't know if it will be a local
call if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to
get 778 DID than 604?
Yes they are both the same calling area. The 778 area code is an
Hello,
I'm at a loss for a way to find the serial number of a Digium analog
card without physically removing it from the server. The only time I
have physical access to this particular installation is during business
hours and that's obviously a bad time to be taking a server down.
It
Adam KOSA wrote:
this is what's most likely as i have no experience in asterisk
configs. I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten =
Rob Schall wrote:
We have the caller id with name option enabled with our provider,
however, our polycom 501 phones will only display the number of the
incoming call. Is there a way to see the callerid name from the cli when
the call is coming in (like a print in the dial plan)? I'm not sure if
Ray Wadkins wrote:
I had the bright idea to set up a virtual extension that would just
ring, virtually. Then we could use call pickup to snag the call at an
extension and be able to open the door. Unfortunately, I can't figure
out how to get that to work. Wait(30) and Answer(3) don't
Matt wrote:
Community,
I have put up www.voip-wiki.us http://www.voip-wiki.us
My apologies to our fellow Asteristians outside the us... this was the
only easy domain available.
What's wrong with voip-info.org ?
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Mark Quitoriano wrote:
Hi i have an asterisk pbx with E1 port connected to another PBX. Im
trying to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get the number 2 as it's DNIS when i change it to ZAP/1
or ZAP/g1
Patrick wrote:
Thanks for the info Trevor. Was your proof of concept also with Windows
PCs or *nix PCs? I haven't played with realtime yet so I might be in for
a bit of a learning curve.
This was just on Linux user stations with a simple bash script that send
a request to a web server.
Rob Vinson wrote:
Does anyone know if I can get Incoming caller id name and number on a
sagnoma PRI
The bigger question is if your telco is sending it to you. asterisk
generally takes care of everything automatically, provided it's
available and you've configured your PRI properly. Number
in my project.
This was a single day project with Fedora Core 5.
Best regards,
Trevor Peirce
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Vincent Tam wrote:
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect
to the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music,
users reported that they could not hear the music but can only hear
I have a TELUS PRI for a while, resold via Bell... dropped it after a
few months due to broken promises and failure to deliver /any/ of the
things we said we required when ordering.
During this time, I learned that with a TELUS PRI you cannot send name.
It's simply dropped at the switch. If
Supa wrote:
Thanks that worked, but it still tries to bridge call after dtmf, then
fails instead of moving on to next number to dial and page
So tack on a g to the end of your dial strong, to continue along the
dial plan upon disconnect.
___
Jean-Marc Salsa wrote:
exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r
mailto:SIP/[EMAIL PROTECTED],30,r)
Everything works perfectly, except when the softswitch, or the PSTN
sends back RingBack Tone.
I can see the RTP flow arriving to Asterisk,
but, it seems that Asterisk doesn't forward it
Mitch Thompson wrote:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this
Michelle Dupuis wrote:
I am trying to set callerid from a PHP script, using one of two
functions as shown below (setid1 and setid2). The first function
works great with regular names and numbers, BUT, if I call the
function with (Test,UnknownNumber), the cid number gets set to
asterisk. Why
Jerry Jones wrote:
From asterisk, you do not hear anything other than ringing as it does
not cut the audio path through until it receives the answer from the
far end, hence the steady ringing.
So instead of Dial(Zap/g1/1800xxx,,r) just do
Dial(Zap/g1/1800xxx,,) so early audio can make
Barzilai Spinak wrote:
I've seen several examples that use extensions such as;
s-BUSY
s-NOANSWER
etc...
It's more or less evident what they do, but I've searched for some
FORMAL documentation everywhere and have found nothing.
Do they work for anything else than s-? (I think I've seen other
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and moves on
sometime stoping before it finishes hello. The rest of the text reads
fine. Anyone else have this issue??
Try adding this...
Douglas Garstang wrote:
I wonder how this could actually work? If Asterisk sends an INVITE to a phone,
and the phone responds with 'Moved Temporarily', and Asterisk sends the INVITE
again, isn't the phone just going to send 'Moved Temporarily' again?
If the phone is saying to redirect the
Matthew Fredrickson wrote:
Don't know. That definitely sounds weird though. I'm laying my bets on
using the right version of fxotune helped a lot. Possibly some user
error, or something like that, or maybe you didn't have the right
version of zaptel loading on your machine (it was loading 1.2,
Ken Williams wrote:
I've spent all day today trying to fix an echo problem and I've made no
ground whatsoever.
Have you played with fxotune?
Seems to be a very nice tool that is less than well documented..
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Lee Jenkins wrote:
Does Asterisk strip off the quotes when storing the value?
You could do a 5 minute test to figure that out...
blah.agi:
SET VARIABLE testme I have quotes!
dialplan.txt:
exten = s,1,AGI(blah.agi)
exten = s,n,Set(regular=no quotes)
exten = s,n,NoOp(regular is ${regular})
Bruce Ferrell wrote:
I've been looking through everything I can find and observing the
mysql logs and I don't see password changes passing through to the DB.
Is that correct?
Works fine for me with 1.2.14.
Trevor
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Carla Schroder wrote:
On Thursday 04 January 2007 18:01, Forrest Beck wrote:
[snip] Currently I am taking music
from a CD (our campus jazz band has recorded a CD), converting to WAV,
using Audacity to convert the stereo tracks into mono, drop the gain
to -15db, then I use sox to convert to
this:
datamoredata
It's common for CSV files to escape quotes by putting two of them to
indicate it is a quote within the string, not the end of the string.
Perhaps you could accomplish what you're going for with something else,
say an underscore character?
Regards,
Trevor Peirce
Hello,
Is it possible to set up a timeout for IAX when something like the
following happens?
-- Executing Dial(SIP/someone,IAX2/somewhere|45) in new stack
-- Called somewhere
-- Call accepted by 1.2.3.4 (format ulaw)
-- Format for call is ulaw
nothing happens here for 15 - 30 seconds - caller
Sebastian Milioto wrote:
Hi all,
I have a public ip in Linksys RT31 (2 FXS port + 3 swtich port + 1
uplink port). I want to add behind it, a Linksys pap2 (uplink port + 2
FXS port) with private ip.
I understand that I have to configure Port forwarding or port
triggering (really I'm not sure
Miguel Cavazos wrote:
Hi guys, im using realtime and I want to show registered users or
online users on a webpage and offline users. Im taking regseconds
field to make this happend
If regseconds value is 0 then user appers offline, it regseconds is
something else then its online, but
I've seen lots about presence and Polycom phones recently. I've got one
here for evaluation but noticed other phones only seem to appear busy
when they initiate a call. If they receive a call, they still show as
available.
Is this a config problem on my part, or is that as far as presence is
Panitaxx wrote:
I have an ISDN PRI E1. I want to send an audio before answering, I am
using noanswer option in playback app but all the audio is muted
before the answer. I would like to play this audio.
I have a T1 and a few months ago my ability to playback audio before
answering ceased.
Ing. Marlo R. Beltran G wrote:
Hi,
I am about to buy ip pbx asterisk system but what ip phones do you
recommend? Are polycom ip all functional with the ip pbx system???
We just got a Polycom IP501 for testing and have thus far been
unsuccessful at getting it to regiser with asterisk.
Damon Estep wrote:
The behavior is a little weird, the * key is not recognized during the
portion of the greeting where the extension number is being played
back, after it is played back, for the duration of the greeting, the *
key is recognized and works as expected.
res =
Kevin Hanson wrote:
The Polycom 600 supports Shared Call Appearance Signaling. The
Polycom documentation states:
...
The phone supports shared call appearances (SCA) using the
SUBSCRIBE-NOTIFY method in the 'SIP Specific Event Notification'
framework (RFC 3265).
Will Asterisk support this
Eric Bullen wrote:
I hope someone can offer me some help with this. Basically, the
current CVS version of Zaptel will not compile under Fedora Core 4. I
have closely followed the directions in
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
using the versions given in the
Brian Capouch wrote:
Set(DB(family/key)=) sets the value for the key to null, but that
doesn't appear to be equivalent to removing the key entirely.
Or maybe DBDel isn't deprecated, like the other two are.
It's not deprecated. There is no code yet for a DBDel type function.
Edwin Lam wrote:
does anybody has experienece with Sipura SPA-841 phone unit?
how's its sound quality especially speaker phone? i have several
Grandstream phones and was getting fustrated about the quality
and bugs of their firmware.
As the other's have said, the speakerphone is useless. My
Philip Fleischer wrote:
With sipura (I tried this with both the 3000 and 841) set to prefer
the g726-32 codec, a call from the sipura to asterisk will use g726.
You need to enable DEPRECIATED726 or something like that in the Makefile.
___
Steve Prior wrote:
I just got a refurb Sipura SPA-2000 and was able to assign it an IP
address with DHCP and ping the device, but then I ran the firmware
upgrade utility to bring it up to spa2k-2.0.13g which seemed to
work just fine, but after it rebooted I cannot connect to its
webserver for
Nathan Goodwin wrote:
If it isn't agiast there agement, I would happy setup a resale
server for this just as you said, and probly at the prces you listed,
I will look into this abit more later today.
I would be interested in such a service providing the minimum (bi?)
monthly billing is single
Steve Prior wrote:
I've got a Sipura SPA2000 ATA basically working (I can place calls
between the
extensions plugged into each of its ports) and part of that was
setting up the
dial plan on the SPA2000 to match the one in Asterisk. This seems
like a pain
to deal with long term and I don't know
Steve Underwood wrote:
I have one weird audio log from a new HP combination printer and fax
machine that i haven't sorted out yet. These HP machines really are
total crap. I have workarounds in spandsp for several blatently wrong
things they do. I don't yet know who is at fault with this latest
bam wrote:
How or when is the voicemail name actually played?
If you do *not* record your unavail / busy messages the greet will be
played. Sort of like--
GREET is unavailable. Please leave your message after the tone...
HTH,
Trevor
___
this in asterisk to send CID Name from the
CPE side.
Bounty: $15 USD
Means: PayPal
Condition: Will be paid after verified to be valid
Regards,
Trevor Peirce
___
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