Christoph Eicke wrote:
Yes, I do it in the following way. In extension.conf add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then automatically gets directed to their
voicemail where they have some options.
I hope
Tomá¹ Komárek wrote:
Hello, I have such a problem. I have an * configured as a peer connected
to the gateway to PSTN.
While calling to the switched off cell phone, the gateway sends to the *
the SIP message 180 with the SDP part, and also a lot of rtp packets
containing the operator's in
[EMAIL PROTECTED] wrote:
Thank you W, asking differently:
(Suppose) I have a very reliable hardware,motherboard,power
supply,bios,kernel,configuration, I have reliable and fast everything.
Now what is the maximum number of FXO/FXS modules? What does it depend in
asterisk/tdm cards now?
Why
Peter Svensson wrote:
On Fri, 12 Aug 2005, Bruce Ferrell wrote:
Hardware, possible. Unlikely to be cabling. It's usually a timing setting.
The blue alarm is really a very specific alarm condition normally. It
cannot quite see how it can be generated accidentally. Something along the
+#define DEFAULT_DTMF_LENGTH250 * 8
#define DEFAULT_MFV1_LENGTH60 * 8
#definePAUSE_LENGTH500 * 8
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Sherwood McGowan wrote:
I'll pass that on to my lead engineer, he was under the assumption that
rfc2833 was too unreliable. I personally don't know, but will look further
into the matter.
You need a new engineer. OOB DTMF like RFC2833 is more reliable than
inband. With inband even a tiny
George K. Konstantoulakis wrote:
Geia sou Irakli,
I would have to agree with Bryce that from the debug output the problem
seems to be with the dialed number.
Unkown Number Type Unkown Number plan point to that.
You should probably check out if you can start extensions with 3 ...
Maybe he
Ric Moseley wrote:
Also, what does the RED mean in this?
[EMAIL PROTECTED]:~]#more /proc/zaptel/*
::
/proc/zaptel/1
::
Span 1: WCFXO/0 Generic Clone Board 1
1 WCFXO/0/0 FXSKS (In use)
::
/proc/zaptel/2
::
Span 2: WCFXO/1 Generic
hugolivude wrote:
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS).
Before I implemented Asterisk, some users were using Bell services to
set-up 5 way calling: The user would set up a three way call on one
line, switch to the second line, set up another 3 way call and then
link the two lines
=${HANGUPCAUSE}) and a
Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to
see WHY the call was hungup.
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r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care
-HEAD version of Asterisk in order to have
supervised transfers.
Not quite correct. You can do supervised transfers with 1.0.x if your
phone supports it. Last I heard GS Budgetone does not support
supervised transgers.
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r: Generate
or not. Definitely in the
1.4 release.
What is the specific problem? We hav been doing supervised transfers
with 1.0.x and Polycom phones for several months.
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r: Generate a ringing tone for the calling party, passing no audio from
the called channel
this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.
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scott kerschner wrote:
Hi peoples
Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they
have configured one before for iax.
Zultys products do not support IAX. What in the world made you think
they did?
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r
VLAN..' We don't run any
VLANs. Is there some way to skip this?
In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin
VLAN Id' set to blank values.
Disable CDP on the phone.
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w/o a
T1?? (I'm not really sure how * is connected to the channel bank).
Would I have to have a T100P (whatever the new model is.. T1/E1
selectable.. blah blah) and a T1 xover cable? (If so, suddenly the deal
just got more expensive)
You need a T-1 port
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check the changelog.
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That announcement was an Aprol Fools Joke. There is no Asterisk 2.0
Justin Selleck wrote:
Is asterisk 2.0 real? Running in c#? I see references to it but cannot
find it anywhere.
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r: Generate a ringing tone for the calling party
and FXS modules on your card, then
rumor has it that you do not need the power connector.
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r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
should not be dynamic.
Asterisk has significant issues with ANY transient DNS issue. I've been
told that this issue has been addressed in CVS-HEAD, but have not
personally tested this.
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r: Generate a ringing tone for the calling party
Geoff Manning wrote:
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated and
you're also pushing VOIP data over that segment you'll have problems.
In practice most networks are not that busy, but it can happen. If
your phones, switch and NICs are VLAN
- Phone - Switch?
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Geoff Manning wrote:
Eric Wieling aka ManxPower wrote:
Are your phones on shared links to the switch?
i.e.
PC - Phone - Switch?
Actually it is a legacy PBX - Asterisk integration
Legacy Handset -- Mitel SX 200 -- Asterisk -- Switch -- Router
The calls come inbound over the internet
the CVS checkout of 1.0.x dated Aug 3 2005
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You want it to be no.
Panitaxx wrote:
yes. overlapdial=yes.
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r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your
someshwarak wrote:
Hi * users,
I want to know if STUN suport is available with Asterisk.
Kindly let me know. I have posted this also in DEV list but none replied to
me.
Short Answer: No.
Longer Answer: No, and most people that think they need STUN don't
actually need it.
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Peter Svensson wrote:
On Mon, 8 Aug 2005, Kib Eki wrote:
Hi,
we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our
old PBX. So now we could migrate to the * server.
But, there are two things we can't live with:
1. A call from the outside to the old PBX is
Anish Basu wrote:
Hi,
For some reason, my AGI perl scripts cannot write to the CLI console using
standard error. I ran the agi-test.agi test script that came with asterisk
and verified that the problem was not with the code. Asterisk is always
started with 4 or more v's, yet this the CLI
Tzafrir Cohen wrote:
On Mon, Aug 08, 2005 at 03:50:02PM -0500, Eric Wieling aka ManxPower wrote:
Anish Basu wrote:
Hi,
For some reason, my AGI perl scripts cannot write to the CLI console using
standard error. I ran the agi-test.agi test script that came with asterisk
and verified
of the other feature codes.
Call parking and # transfer work though, so I'm guessing they're simply
not implemented yet, as of 1.0.8.
They will never be put into 1.0.x since 1.0.x does NOT get new features.
It's bug fix only.
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an extension?
You have come across one of the few times you want immediate=yes. When
the phone is picked up Asterisk will try exten = s in the context that
channel is in.
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Chris Coulthurst wrote:
I have two 300s and 4 500s. The 300s talk the same language, but have a
lousy screen. The other thing to consider is, while it does have the
'monitor only' speaker, the volume is horrible. Cranked up to its
highest setting, you can't hear voicemail with ANY
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is
that the default ring for calls from people in the phone direcory is a
silent ring. One of the entries in the phone directory config file
for is the ring type for that entry.
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Only terrorists use the r option to Dial
not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. r makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.
This needs to be in the info for show application dial
--Eric
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a ringing tone no matter what the carrier sends
back. Then the Dial timeout can expire and the caller can be sent back
to the user's mailbox (assuming the cell carrier didn't answer the call
and send it to the cell phones voicemail).
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r: Generate
Eric Wieling aka ManxPower wrote:
Robert Goodyear wrote:
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
device requires the 'r' option or not.
You almost never want it enabled on a trunk line, only for terminal
devices.
Almost nothing generates inband ringing. That has nothing to do with r.
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r: Generate a ringing tone for the calling party
[3983]: chan_zap.c:4717 zt_indicate: Don't know
how to set condition 16 on channel Zap/1-1
-- Started music on hold, class 'default', on Zap/1-1
-- Stopped music on hold on Zap/1-1
-- Hungup 'Zap/1-1'
It's a warning, not an error. You don't have /etc/asterisk/indications.conf
--
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:[EMAIL PROTECTED]/7771
[10.0.0.115]
type=peer
context=default
secret=
username=7771
fromdomain=10.0.0.115
canreinvite=yes
dtmfmode=RFC2833
qualify=yes
host=10.0.0.115
insecure=very
fromuser=7771
Remove the qualify=yes and Asterisk will stop sending the options packets.
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[EMAIL PROTECTED] wrote:
Dear Christ and helpers!
Thanks for the information about the r option in asterisk. However, I have
tried it and it doesn't work. Moreover. My Sip phone is ringing just fine
when I call the exteral PSTN phone, but not hear the ring from the IAX
phone. When dial the
Adam Holt wrote:
On 4/8/05 3:41 pm, Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Thursday 04 August 2005 10:32, C F wrote:
If you are trying to send an SMS to a US cell phone then you can use
the email of the phone, like this:
Verizon subscriber: [EMAIL PROTECTED] (where xx is the
Adam Holt wrote:
On 4/8/05 3:41 pm, Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Thursday 04 August 2005 10:32, C F wrote:
If you are trying to send an SMS to a US cell phone then you can use
the email of the phone, like this:
Verizon subscriber: [EMAIL PROTECTED] (where xx is the
is the
console output when a call is recieved. what am i missing here?
Make sure you have a /etc/asterisk/indications.conf
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Only terrorists use the r option to Dial.
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Rich Adamson wrote:
All of these postings about ringing two (or more) phones is well known
and fairly well understood by everyone. The issue that everyone seems
to want to ignore in the postings is the busy lamp field functionality
of key systems (not pbx's). I'm not the OP and I've been around
Many people seem to want this feature. I think they are just
confused. I've never actually heard of a good reason to let multiple
devices register with the same username/secret. Most of the time they
want a call to ring on multiple devices and they are trying to make
a device == extension,
I ring multiple phones ALL THE TIME without needing duplicate
username/secrets. The following line wrapped, but you can still see
what's happening. When someone dials extension 3400 the devices with
SIP the three SIP usernames (we set them to MAC-[a|b|c|d] where the
letter indicates which
I found your original message:
Can somebody please help here. At least respond and call me a moron.
I have tried everything. I finally gave up and installed [EMAIL PROTECTED]
from the iso and I am back to the exact same problem. Everything seems
to work but my extensions are all busy. I used
Tim King wrote:
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
Once is enough. Perhaps you did not provide enough information for
anyone to help
mohammad wrote:
I have problem with incoming Sip call to users behind Nat.I set the following
for my users behind Nat:
nat=yes
canreinvite=no
qualify=yes
--
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Mark Twain
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, but I'll take it out and test again today.]
Are there any obvious problems with that digitmap? Anything else that I
should take a look at?
I've had significant issues with using T. You have to be careful to
have NO overlap. You have overlap. 911 and 9411 will also match 9xT.
--
Eric
start to dial.
where does this delay come from?
has it to do with 'overlapdial=yes'?
This is normal behaviour if you use '.' in your extensions.conf. Use '!'
instead and Asterisk will start dialing immediately.
But not on Asterisk 1.0.x
--Eric
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locally generated
by the PAP2-NA.
If the channel is answered Asterisk has to do inband (audio) tones.
--Eric
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. I use mutt.
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a remote system which successfully connected to my asterisk server i'm getting the error as call failed: 499 Not acceptable here .
That reequently means that you have a codec problem. In sip.conf put
disallow=all and allow=ulaw. If that works, you can experiment.
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have an /etc/asterisk/indications.conf ?
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more flexible method to handle
the various results of a Dial.
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Asterisk (1.0.x, I think) with
lots of custom dialplan, script, and other things.
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Kevin P. Fleming wrote:
Eric Wieling aka ManxPower wrote:
You do realize that host= only applies for calls from Asterisk to the
TNT, right? You need permit/deny to match for inbound connections.
Your Asterisk server is open to anyone that claims to be the user tnt.
That's not true
doesn't
say much about the digitmap and refers you to the MGCP RFC digitmap
handling.
--Eric
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.
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Dean Collins wrote:
I've just gotten off a skype conference call and it pisses me off that
the quality of skype is higher than my asterisk calls.
Is there such a thing as a super high bandwidth codec?
Asterisk does not support wideband codecs as far as I know. Most
telephony gear expects
(s${CALLERIDNUM})
exten = 22999,2,Wait(3)
exten = 22999,3,Hangup
Don't put extra spaces in extensions.conf
exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})
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Asterisk
://www.ilbcfreeware.org/
iLBC does not seem to support any kind of wideband mode, so it will not
be any clearer than plain old G711 ulaw/alaw.
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ones on eBay are clone cards.
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Steve Underwood wrote:
Eric Wieling aka ManxPower wrote:
Do yu have a link for wideband-ilbc info?
It is described on the GIPS site, along with the narrow band ilbc. The
wideband one is not offered to the world on a royalty free basis, as the
narrow band one is. I have never looked at how
It seems that exten = fax does not work in a macro.
Asterisk detects the fax, since it complains about no fax extension, but
I have an exten = fax in the macro.
Has anyone else experienced this?
--Eric
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server is open to anyone that claims to be the user tnt.
--Eric
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models can't.
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Eric Rees wrote:
We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX
sounds better then the land line.
Using UDP or using TCP? Might want to confirm by using tcpdump.
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Lists wrote:
On Friday 22 July 2005 18:18, Andrea Cristofanini - Gedam Europe Srl wrote:
We use Supermicro and we have NO problem at all :-)
Ditto, SuperMicro is solid. One of the best price/performance/quality servers
you can buy.
I'm still in therapy after my first (and last)
do this
in sip.cfg (the polycom config file) and set the dialplan.digitmap option.
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with a workaround or solution to this?
The IAXy (both versions) does not support DNS name resolution.
Would it be accurate to add ...and never will support DNS?
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Matthew Boehm wrote:
Is there anyone out there with news on VAD/Silence support in Asterisk?
Asterisk does not support VAD. If you need VAD then don't use Asterisk.
--Eric
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Matthew Boehm wrote:
Eric Wieling aka ManxPower wrote:
Matthew Boehm wrote:
Is there anyone out there with news on VAD/Silence support in Asterisk?
Asterisk does not support VAD. If you need VAD then don't use Asterisk.
--Eric
Wow. Thanks for stating the obvious. You quoted my
). The second one points to the external IP
of my router that has port forwarding. I can roam pretty much anywhere
without any reconfiguration.
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Lists ([EMAIL PROTECTED]) wrote:
On Monday 18 July 2005 16:04, Brian Capouch wrote:
Andrew Kohlsmith wrote:
If you don't want or don't like ABE, don't use it. Nobody is cramming it
down your throat.
I have to bite my tongue when I read these conspiratorial posts about
ABE as if it
Did you remove the r option from your Dial line?
Hugo Begglo wrote:
Hello again everyone,
I'm having this same issue with Asterisk. Any ideas ?
Hugo
Cullin J. Wible wrote:
After all of your feedback and a discussion at Teliax we have fixed this
issues.
It appears that when dialing a PSTN
Kevin wrote:
Is the pager filed in the vm config still for the outcall destination or
where do you specify the number to call for the outcall?
Sorry.
You use notify= option in voicemail.conf:
3532 = 8711,Toni Hawkins,,,tz=central,notify=15045556389
-Original Message-
From: Eric
the script or perl log to see
what's going wrong?
-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED]
Sent: Friday, July 15, 2005 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VM Outcall: Rube Goldberg Edition
If you have a Polycom make SURE the cable going into the handset is
FULLY seated.
Andrew Kohlsmith wrote:
First, I would like to commend you on your excellent list post. Detailed
without burying people in details and to the point, and not running around
pointing fingers.
Now for the bad
Make sure you have /etc/asterisk/indications.conf If that fixes it,
let me know.
Michiel van Baak wrote:
On 14:12, Thu 14 Jul 05, Bill Wong wrote:
Thank you Michiel.
I tried to remove m and use r , but still not working, after I change r
to R , it is working. Anybody know why?
This is
exist.
(Check /usr/src/astersik/configs/sip.conf.sample)
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dbruce wrote:
Price: Polycom = $300USD (IP600), Cisco = $320USD(7960)
Did you include the price of a power supply and SIP firmware in the
Cisco price?
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Asterisk
supports PoE out of the box.
The IP 30x and 50x support PoE with a special cable from Polycom.
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jonny hashem wrote:
which the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.
None.
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Mark Phillips wrote:
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf
Anyone that tells you to use reinvite= is confused. The option does not
exist (check the source code if you don't believe me). reinvite= is one
of the many Asterisk Urban Myths.
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Kevin P. Fleming wrote:
Eric Wieling aka ManxPower wrote:
That's actually not correct. The IP600 supports PoE out of the box.
The IP 30x and 50x support PoE with a special cable from Polycom.
Bummer...I thought the built-in PoE chip was one of the few upgrades in
the 300-301 and 500-501
for
the phone.
There are different versions of the cable. Cisco PoE cable and a IEEE
af standard.
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or something like that on one of the Asterisk boxes
to find out what you want to know.
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Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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know whether Voiptalk have changed anything.
Is this a known bug in certain versions of Asterisk?
If I do iax2 debug, I *can* see DTMF frames being sent and acked for
each digit. Wo I can't understand why I'm not hearing all the digits.
Any ideas?
Cheers
Tony
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Eric Wieling * BTEL Consulting
.
If there is still new voicemail, then the call happens, and the person
is told how many messages they have, and have a chance to log into
voicemail. I would like to be able to detect voicemail answering and
not start playing prompts unless a human answers.
--Eric
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Eric Wieling * BTEL
.
Wrong. I saw at least two people answering you. They referred you to
the Dial command docs in the Wiki.
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Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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for Distinctive Ring,
see the Zap config file.
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Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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Andrew Sayman wrote:
Okay, so I believe I've gotten rid of the IRQ conflict, and it's still
not working.
All analog FXO ports are considered answered when the dialing is finished.
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Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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Asterisk
of the analog lines
to test this out.
Asterisk does not check for a line before dialing. Heck, it doesn't
even check for dialtone before dialing. This is a known issue. Of
course, make sure you don't have busydetect=yes or callprogress=yes.
--Eric
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Eric Wieling * BTEL Consulting * 504
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