[EMAIL PROTECTED] root]# arp -an
? (172.16.8.1) at 00:04:C1:21:CC:C0 [ether] on eth0
? (172.16.8.53) at 00:04:F2:01:FA:94 [ether] on eth0
? (172.16.8.48) at 00:04:F2:01:FA:D8 [ether] on eth0
? (172.16.8.62) at 00:04:F2:01:FB:65 [ether] on eth0
? (172.16.8.60) at 00:04:F2:01:FB:20 [ether] on eth0
. In this example you
are dialing no digits.
Carey O'Shea wrote:
I'm using Dial(Zap/X/) as suggested.
However, Dial(Zap/X) does indeed work for me. So I'm curious, what's the
difference between them, and when wouldn't just Zap/X work?
On Wed, 2006-06-14 at 11:14 -0500, Eric ManxPower Wieling
Servetas, Andrew wrote:
I started with Inband, then went to rfc2833 for a while and noticed
other issues with IVR's, so now I'm back to Inband.
http://www.google.com/search?hl=enq=site%3Alists.digium.com+talkoffbtnG=Google+Search
--
Now accepting new clients in Birmingham, Atlanta,
Jean-Michel Hiver wrote:
Hi List,
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP - VoIP traffic as well? I'm toying with a
SIP gateway which has apparently a terrible call quality and would like
to know if there is any way asterisk can help
Warren wrote:
I was just told that for my forthcoming system I will be getting a data
T-1 instead of a voice T-1. Given that all of the handsets will be voip
phones, no analog at all, do I need echo cancellation? I looked at the
voip-info wiki and it seems to me that the answer should be no
If there is an Answer(), Playback(), Background(), or anything else that
answers the line before Dial(), you must have set up
/etc/asterisk/indications.conf or Asterisk will not know how to do
inband ringback.
Derek wrote:
Hi all,
I've got a fairly simple setup, a 4 port zaptel T1 card with
Asterisk does not support transcoding G723.1. If you want to transcode
G729 then you need a G729 license from Digium.
Patai Tamás wrote:
Hi,
I set a preference order of the codecs to my sip.conf
/[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context =
The problem was fixed in 1.2.0
amna saleem wrote:
No , actually I am using Asterisk-1.2.9.1
I will try the q option though
Thanks and regards,
Amna
On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
I assume you are using 1.0.x. Add the q option to the Meetme
extension. 1.0.x
Make sure you have /etc/asterisk/indications.conf set up.
People that don't know any better might tell you to use the r option
to Dial. Those people are confused. Don't do that until you have tried
everything else.
Tim Sharp wrote:
I am running on 1.2.7.1 and have an intermittent problem
Carey O'Shea wrote:
I swear Dial(Zap/X) was the first thing I tried and it didn't work, but
now it works fine... hmmm maybe I forgot to reload my extensions or
something like that.
Don't expect Dial(Zap/X) to work. Expect Dial(Zap/X/) to work.
--
Now accepting new clients in Birmingham,
Contact Digium to purchase a G729 license.
Osama Kamal wrote:
I am having a problem with asterisk transcoding GSM and G729 codecs, the
error message is below:
Jun 14 09:38:12 WARNING[18292]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/3004-fcfb(256) to
Brent Torrenga wrote:
I think [EMAIL PROTECTED] allows a user to search a directory by either first
OR last
name, right? I don't know for sure since I don't run [EMAIL PROTECTED]
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric
ManxPower Wieling
Sent: Wednesday, June 14, 2006 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No ring tone on outgoing calls
Make sure you have /etc/asterisk/indications.conf set up
I assume you are using 1.0.x. Add the q option to the Meetme
extension. 1.0.x has a known issue where enter/exit sounds cause
increasing delays.
amna saleem wrote:
Hi All!
I am facing some delay in conferencing.
Using DIAX for Voip calls ,no hardware used yet
I am using MeetMe to
There is a standalone app included with Cepstral. I think it's called
swift.
Doug Lytle wrote:
Doug Crompton wrote:
Ok Thanks. I just registered 'Diane' also. She seemed to have the best
voice. I am curious if you added the Cepstral app or used the festival
method described in the Cepstral
It's called hotline or Private Line Auto Ringdown (PLAR).
SIP: It's a function of the phone, look for hotline in phone docs
Zap: immediate=yes, runs exten = when phone is picked up
Cisco and others: Look up PLAR
[EMAIL PROTECTED] wrote:
Basically, I am looking to set up an extension which will
When the customer calls in, have the exten = run an AGI to create a
.call file (set the creation time of the file a few seconds in the
future to be sure Asterisk doesn't call immediatly). See sample.call
in the Asterisk source and the wiki
Jean-Michel Hiver wrote:
Tigran Kocharyan a écrit
The number of taps the EC has to deal with is the delay on the PSTN
side. I can't imagine echo is more than a few ms in modern TDM
networks. This latency has NOTHING to do with VoIP latency, since the
echo must be canceled BEFORE it gets to the VoIP side of things.
Brian Swan wrote:
My wife
Sounds like EM Wink to me.
Steve Edwards wrote:
I have what my SBC/ATT representative calls a Supertrunk, but he can't
tell me the specifics I need to know to configure Asterisk to work with it.
By fiddling about, I've come up with the almost working configuration
below. This works except
Jason Lixfeld wrote:
If you have to use it, make sure you only use the mpg123 bundled with
the asterisk distribution. mpg123 from any other source (yes, evem the
developer's website) will yield major issues.
mpg123 is NOT bundled with Asteirsk. make mpg123 will DOWNLOAD the
mpg123 source
Pietro U wrote:
i have a problem, if i dial [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] i can
call my doamin users without any registration in the asterisk. how to
block this?
You can start by not permitting it. By default Asterisk will allow any
client to connect. You control this by using
This is a feature you enable/disable on the phone. Call waiting.
Jeremy Koski wrote:
When I pick up the handset to make a call, SIP 483 BUSY is returned
to the server and the below statement works. However, when I'm in a
call, the BUSY status is not returned on my Cisco 7960 phones.
Is
Doug Crompton wrote:
Ok well I am not crazy! This seems like such an important issue I am not
sure why it has lasted for so long. DTMF is the backbone of everything we
do here. Without it we would not have calls!! At least get the DTMF stuff
right. I feel a little guilty complaining since this
Erick Baum wrote:
We setup a company with 50 of these phones and had my client not been as
understanding as they were, that could have put me out of business.
What an unbelievable nightmare. This was about 8 months ago when the
firmware was so bad the phone was a better paper weight than
I don't think you can do what you want. The Zap custom calling features
work very much like Centrex service in the USA.
Henry Margies wrote:
Hi all,
I have some problems configuring the right behaviour of Zaptel devices
in asterisk. Especially with three way call, call waiting and call
Stephen Bosch wrote:
Hi:
I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with
kewlstart signalling.
When an outside caller calls the server, the TDM-400 goes off-hook and
provides a ringing tone to the caller. If the caller hangs up before the
receiving party answers the
DTMF problems happen at the point where the PSTN call is converted to
VoIP. EXCEPT where you are using inband DTMF and ulaw or alaw codec.
inband DTMF does not work with any other codec.
Mr. Jones wrote:
Hi Folks,
I'm trying to test out Asterisk overall.
I'm having some problems with DTMF.
Start out by using a valid Dial line:
Dial(Zap/G3/${exten},120,gM(connected));
You don't put commas beween the dial options
Julian Lyndon-Smith wrote:
svn trunk 31497
For the life of me, I can't get this :) I want to be able to catch the
situation where the calling party hangs up *before*
Then you have a typo.
Akpome Akpoguma wrote:
I used it on a 4 digit extension
From: C F [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
my $dialstat = $AGI-get_variable('DIALSTATUS');
Hi,
I have tried that as well.
Thanks for the suggestion.
Any other thoughts.
1) What version of Asterisk are you using?
2) Can you get any other dialplan variables?
3) Are you running the Dial app inside your AGI or before you run
Derek Lee-Wo wrote:
there that might benefit you in a situation like this. Go look through
your zaptel source
tree for fxotune and see if it cant possibly correct some of the
problem you're having.
Thanks for this suggestion. I ran the test and activated the settings
with fxotune -sand I'll
Mark Phillips wrote:
Don't ya just love living in this technological backwater they call the
USA? DECT technology was released almost 20 years ago. In most of the
world it's been and gone.
I believe that DECT is approved for use here. Either that or Staples et
al are selling loads of illegal
If you want to roam between GSM and WiFi while on a call, the GSM
carrier is going to have to support it.
Dean Collins wrote:
Mark, how ignorant are you?
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Monday, 22 May 2006
You would normally get Congestion tone. Show us the dialplan for
outgoing calls.
Michael Knill wrote:
I get MOH when the caller hangs up. Is there any way I can just get Busy
tone.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Don't specify the remote side by name, specify it by IP address. If
asterisk experiences even 1 dns failure it will not try again until a
reload/restart/whatever.
[EMAIL PROTECTED] wrote:
I'm going to try and lay out all the relevant information I have here in
this one post. I can provide
Nitin Gupta wrote:
Hi,
Is there anyway in Asterisk to know that outgoing call has been forwarded
to voicemail by the callee system?
Some of my users don't want to connect the call if its forwarded to callee
voicemail, so I am wondering if theres anyway to identify this in Asterisk
and drop the
Urban wrote:
May 21 11:03:10 WARNING[12188]: channel.c:2049 ast_indicate: Unable to
handle indication 3 for 'SIP/XX-c52d'
You don't have a /etc/asterisk/indications.conf
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Are any of these FCC licensed for use in the USA. DECT in the USA is
VERY new.
Michael Graves wrote:
HmmmI have a 480i-CT. Does this mean that I might be able to add
third party DECT handsets? Or just the matching Aastra handsets?
Michael
--Original Message Text---
*From:* Dovid Bender
Aaron Paxson wrote:
Thanks Cosmin!!
I didn't realize that the dialplans ran in sequential order. I'll try
that. thanks!
We originally had dialplans run in random order, but people found it too
confusing.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and
R is not a valid Dial option. r is the option you wanted. HOWEVER,
if you are not hearing ringback, r will almost never fixes the issue.
Make sure you have a /etc/asterisk/indications.conf In some situations
if you do not have that file you will not hear ringback.
Sebastian Kayser wrote:
I would need admin access to the SIPura, the NAT router/Firwall, and the
Asterisk server. To see my posts to the mailing lists do a Google
search of site: lists.digium.com ManxPower (no quotes, of course)
Eric Lyons wrote:
I'm still unable to get my SPA-1001 to work behind NAT with an
Paul Dugas wrote:
I have a SPA-3000 that is failing to hanging up pretty often; almost
every day now. The weird thing is that an almost identically configured
(same FW, different HW rev) second unit right next to it isn't having
the same problem. Swap the lines and the problem stays with the
Doug Lytle wrote:
Dave Morrow wrote:
Hi all. I was reading a sample config someone had posted relating to
call forwarding, and in it, they use a Dial command with components
that I cannot find any reference to.
Can someone point me to a reference which could explain the difference
between
AR Tarzi wrote:
Unless reinviting works, wouldn't that add up to what he's experiencing ?
client - asterisk - service provider.. makes that 180k each connection
so 4 of them would give 800k or so.
What I can't understand is: if only g723 is allowed, and Asterisk only
allows it as passthrough,
Hall, Eric M. wrote:
I did not get this back from the list so I'm not sure if this hit the
list last week or not so I'm sending it again. Sorry if this is a
duplicate post!
---
Has anyone had problems with a
Try:
exten = _XX,2,VoiceMail([EMAIL PROTECTED],j)
and
exten = _XX,110,VoiceMail([EMAIL PROTECTED],j)
Álvaro Palma wrote:
I've the following dialplan.
exten = _XX,hint,SIP/${EXTEN}
exten = _XX,1,Dial(SIP/${EXTEN},10,j)
exten = _XX,2,VoiceMail([EMAIL PROTECTED],u|j)
exten = _XX,3,Hangup()
r almost never fixes ringback issues. r will provide ringback when
the caller should hear something ELSE, such as onhook audio the
cellular user you are calling is not in the calling area
Confirm you have /etc/asterisk/indications.conf Not having it can cause
ringback issues in some
If you want dynamic features you must upgrade to Asterisk 1.2.x
Josué Conti wrote:
Eric, thank you very much. But It could help in this case me?
Regards
Josué
2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Josué Conti wrote:
Dinesh, very
You need the span= BEFORE any channel lines. You also need any options
before the channel lines.
Ex:
span=1,1,0,ccs,hdb3,crc4
loadzone=sg
defaultzone=sg
bchan=1-15
bchan=17-31
dchan=16
azyuky wrote:
I know and it's crazy isn't it because this is what i have in my zaptel.conf
--
Now
The problem is called talkoff. Search the mailing list archives.
Also post your sip.conf and zaptel.conf (sans passwords, of course)
Stefan Agethen wrote:
Stefan Agethen schrieb:
I got a Problem with DTMF occuring channelwide in my SIP/ZAPPhones.
I want to post the Solution for this, so
http://www.asterisk.org/hardware
Erick Perez wrote:
Does Asterisk support a Brooktrout TR1000 ?
http://www.cantata.com/products/tr1000/
It seems that they have linux drivers.
I have one around and was wondering if it works with *.
Thanks,
--
Now accepting new clients in Birmingham,
What makes you say that?
Nabeel Jafferali wrote:
That list is obviously not complete ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric ManxPower Wieling
Sent: May 11, 2006 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I don't see anything obviously wrong with your configs.
You don't want relaxdtmf. That can cause the problem, not fix it.
In sip.conf [general] put context=INVALID It's easy to configure
Asterisk to accept calls from unauthenticated devices. Putting that
context line in [general] will make
Josué Conti wrote:
Dinesh, very obliged for the attention. I am using version 1.0.9 of
asterisk
and it is really all good with this version, only this case of atxfer that
it does not function. The function DYNAMIC_FEATURE = to atxfer in my [
globals ] of extensions.conf functions in version
It happens by default. I don't know what you would ask BellSouth.
Lenwood S. Sawyer III wrote:
How do you do it on a PRI and what do I ask my provider (Bellsouth) for
if they permit it?
Eric ManxPower Wieling wrote:
Tim Litwiller wrote:
Not, on your question - but you brought up something
Leo Ann Boon wrote:
Antonio Almodóvar wrote:
I've tried modifying parameters in the bios and I didn't managed to
change the irq.
Does anyone have a machine like mine?
Have anyone changed the irq in order to not sharing irq's?
You can't change interrupt for the card in the BIOS. Disable
ESF/B8ZS is a T-1 config.
azyuky wrote:
Hi Josue and list,
This is part of my dmesg log. Also one other thing, I don't know why the red
light
on span/port 1 of my card keep flashing.. is it normal that way? Or it's
supposed to
mean something? Obviously I have no experience with ISDN before
Tim Litwiller wrote:
Not, on your question - but you brought up something I would really like
to do and I was told it wasn't possible.
how do you do the transfer to cell phone with the hook flash.
Martin Roy wrote:
I doubt it's possible but I'll ask just in case there's a legal way
to do
Joe wrote:
I was just looking through the Wiki for some info on how to retain the
original caller's callerid when make transfers to internal extensions, and I
came across the parameter below:
useincomingcalleridonzaptransfer=yes
There is nothing in the zapata.conf file from vers. 1.2.7 so I am
Frank Garcia wrote:
I'm trying to get asterisk to pass through a call without requiring
supervision on the line.
Any thoughts?
This is the default unless you do something to answer the line.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
David L. West wrote:
There's definitely a ) missing in this line! A good text editor with
Thanks( I finally spotted it after fortifying myself with a good Thai
dinner). I've been using gEdit, so should probably start shopping for
something vim-ish.
For what little coding I do, I use
Alexander Lopez wrote:
The line build out value is a power level that is set based on the
distance from the Device to the T-1 service provider's gateway. If the
device is close by, the gateway requires less power and the line build
out value is lower; if the device is far away, the gateway
Tom Engleward wrote:
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
If Asterisk has a DNS lookup failure it will
never
retry that lookup.
Never meaning until the next reload command is
issued, or until the next restart command is
issued,
or until the next time the OS reboots
Mark Ackroyd wrote:
There was a bug in various versions of Asterisk when outbound calls were
placed using spool files and then could not detect DTMF from the called
party. Without more details, including the version of Asterisk you are
running, it will be difficult to suggest anything to you
Andrew Kohlsmith wrote:
On Friday 05 May 2006 08:47, Eric ManxPower Wieling wrote:
Any one of those will work.
I disagree; this is NOT an Asterisk DNS issue. I specify my hosts by IP and
this still happens (in fact, it happened this morning).
Then you are experiencing a different problem
Wes Baehr wrote:
Simply generate or record a 5-second sample of ringing. Then use
Background() to play that ringing file - if someone presses 9, they will be
routed accordingly, or otherwise sent to your default extension.
Or use the Ringing app, along with the WaitExten app.
--
Now
Wes Baehr wrote:
Simply generate or record a 5-second sample of ringing. Then use
Background() to play that ringing file - if someone presses 9, they will be
routed accordingly, or otherwise sent to your default extension.
Or use the Ringing app, along with the WaitExten app. Or you could use
Are you specifying the remote Asterisk box by IP or by hostname. If by
hostname, then specify it by IP. Asterisk's DNS lookup support has issues.
2) What is your qualify= set to. Set it to yes (2000), or don't set
it at all. Also look at the qualify smoothing options in iax.conf.sample.
Tom Engleward wrote:
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Are you specifying the remote Asterisk box by IP or
by hostname. If by
hostname, then specify it by IP. Asterisk's DNS
lookup support has issues.
In the trunk peer details in AMP I'd set host= to a
hostname. I've
Colin Anderson wrote:
That is not a good metric for call completion. Shitty quality will not be
counted in that metric...
Ah, true dat. However, if quality was crappy believe me my users would let
me know. They are salespeople and wholly intolerant of anything that keeps
them from yipping on
Tom Engleward wrote:
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
If Asterisk has a DNS lookup failure it will never
retry that lookup.
Never meaning until the next reload command is
issued, or until the next restart command is issued,
or until the next time the OS reboots, or until
Sounds like you are using an inband codec and not using ulaw or alaw.
If you are not using alaw or ulaw, then you need to use either INFO or
RFC2844 DTMF. Just remember that both Asterisk and the phone device
MUST be using the same DTMF mode.
Adam Hatia wrote:
I have exactly the same
John Joseph wrote:
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
John Joseph wrote:
Hi
I want to submit a bug about auto-dial , but
I
am not sure on which project the auto-dial comes,
how
to know about which project , auto-dial comes
Define auto-dial.
Thanks Eric
Actually this is only the case with analog Zap channels and
loopstart/groundstart T-1 Zap channels. EM Wink and PRI Zap channels
do not have this problem because the line can tell Asterisk when the far
end answers.
[EMAIL PROTECTED] wrote:
Hello Yusuf,
This is a normal use of zap channels
This is a known problem with the 1.0.x version of Asterisk. If you do
not use join/part sounds it won't happen. I think the q option to
MeetMe stops the join/part sounds. This issue has been fixed in 1.2.x.
Michael George wrote:
We have recently started making more frequent use of the
Remco Barende wrote:
Found it!
It seems that Asterisk is looking at the date / time stamp of the call
file to process the call?? I was simply moving the call files hoping
it would just work (tm)
I guess that the call files created on the samba share I created carried
the time/date stamp
John Joseph wrote:
Hi
I want to submit a bug about auto-dial , but I
am not sure on which project the auto-dial comes, how
to know about which project , auto-dial comes
Define auto-dial.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Arne Morten Johansen wrote:
Hi.
How does this work?
What if this SIP/account was a member (agent) of a queue?
Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when
one of the phones is actively talking, or will the other phones continue
to ring?
You may have seen my other
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the
time. Of course his voice quality is like a morse code with dashes or
dots of connection time.
The next minute he calls me via Skype and
Have you tried switchtype=national ?
Doug Langley wrote:
Hi. recently I have been trying to setup a PRI on asterisk. Inbound
calls are working just fine but I am not able to make outbound calls.
Does anyone know what I need to change to make outbound calls work?
Right now the PRI is
Benoit Panizzon wrote:
On Friday 28 April 2006 15:32, Eric ManxPower Wieling wrote:
What does the R option do?
Indicate 'Ringing' as soon as the called party indicates 'Ringing'.
The 'r' option indicates 'Ringing' as soon as the connection is built, even if
the called party is not yet
Why not just create a .call file when the number is busy? The .call
file tries to dial the destination with the retry interval and max
attempts you specify, when the call goes thru, dial that other number.
Nathan Alberti wrote:
8
Thanks for the pointer Nathan. I slapped something together
What does the R option do?
Benoit Panizzon wrote:
After migrating from 1.2.4 to 1.2.5 I noticed that:
show application dial
does not show the 'R' option anymore. Has this become an undocumented feature
or has it gone completely?
--
Now accepting new clients in Birmingham, Atlanta,
Hans-Peter Straub wrote:
Hello all,
is it possible to make an outgoing call transferable for the dialing phones
like the 't' or T option on the Dial-Command does this for incoming calls?
The t and T option works for ANY call using Dial. Incoming or outgoing.
--
Now accepting new clients
Carlos Alberto Bernat Orozco wrote:
I'm making tests for Asterisk. I've tested with 2 users installing SJphone
and it worked fine, but when I install it over a third user with the
softphone, the phone dial for 2 seconds and a window alert goes out on the
softphone:
Busy
Call rejected: 486 Busy
Jonathan k. Creasy wrote:
Just appending the area code variable is not always going to be correct.
You will need to lookup (google local calling guide) the proper NPA for
the NXX you are dialing. For example, in Louisville, Ky if you dial
948-1592 you will actually reach 812-948-1592 instead of
Matt wrote:
Hi,
With a digium dual PRI card (dual span). Is there any reason I can't
have both PRIs being PRIMARY timing sources? They are both from
different CLECs, and as such I need them both to do their own timing.
No. However, if the two CLECs are close enough in timing, it should
Matt wrote:
Hi,
When I setup a user, I give them an extension like 570xxx. This
is fine and dandy while in one area code, but we've since gone to
other area codes.I'd like the user's to retain the ability to dial
7 digits no matter what number they have. Any thoughts on how to do
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
Matt wrote:
Eric,
Yes.. I am setting calleridnum to be their phone number. And your
example is peachy... except for the fact that it assumes I want to go
out ZAP/g1!!
My problem is I have a very intricite routing plan that routes that
Steven Totaro wrote:
Open the console with verbose turned up. Make a test call and see where
it is hanging. That will isolate the problem.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo
Sent: Thursday,
Steven Totaro wrote:
Try setting the timing to zero on both spans?
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
Digium cards only support 1 timing source per card.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
HDLC Abort errors are usually caused lost or corrupted data from the T-1.
If you configured the span as PRI in Asterisk and it's not a PRI, I can
imagine you might get this message.
Also, if there is a problem with the line itself, I can imagine you
might get this message.
However, in my
Andreas Sikkema wrote:
I believe this is called camp on. Found some examples on voip-info.org
but they assume that you do not hangup the originating phone. Anyone
have an idea how to implement this feature as described above?
When I worked at Philips there were two variants:
- camp on busy
-
Sounds like a classic case of the card being on the same IRQ as some
other device in the system. cat /proc/interrupts will give you
additional information. You'll have to move the card to a different
slot if you find that it is sharing an IRQ.
Thomas Artner wrote:
hmm.. does really nobody
Fred Noris wrote:
Hi:
I upgraded SuSE to 10 and Asterisk to trunk and now
after deleting all modules and previously compiled
stuff and recompiling asterisk, zaptel, and libpri, I
get this failure of asterisk to start:
[pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]:
loader.c:726
Bryan Mahin wrote:
I'm experiencing touch tone recognition issues when calling some outside
phone systems. For instance, if I call my Nextel phone, and try to press
* to enter my voicemail, Nextel's system does not hear the DTMF tone.
I've also experienced other outside phone systems for which I
Paste your zapata.conf.
Giorgio Incantalupo wrote:
Hi Hadley,
I tried usecallerid=no but unfortunately nothing changed. I used another
pc with only one TDM400P because I thought I had too many TDM400P cards
but I got the same behaviour.
Giorgio Incantalupo
Hadley Rich wrote:
On Wednesday
1) Your exten = _1XX,n,Dial(Zap/1/${EXTEN}) does not start with
priority 1 so it will never match
2) The 10 digit number you dialed does not start with a 1 so it will
never match, even if the priority issue is fixed.
Asterisk knows that once you've dialed 7 digits no OTHER pattern
This is just the way Asterisk works.
hugolivude wrote:
Thanks, but the problem's with the first extension:
exten = _NXX,1,NoOp(Number dialed ${EXTEN})
exten = _NXX,n,Dial(Zap/1/${EXTEN})
The problem is I _do_ get a match as you can see by the CLI output,
but it shouldn't match IMO -
understandable, but the pattern matching
(at least in this case) appears broken to me.
H
On 4/26/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
This is just the way Asterisk works.
hugolivude wrote:
Thanks, but the problem's with the first extension:
exten = _NXX,1,NoOp(Number dialed ${EXTEN
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