On Apr 30, 2006, at 9:03 AM, Eric ManxPower Wieling wrote:
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is
used to transport audio for SIP (and other protocols). This means
that ANY jitter on the SIP Phone - Asterisk link will cause audio
What does your dial command look like?
If had DID working before, all you have to do is send the same information on to your pbx. For example, if the telco was sending the full number dialed on to your pbx, then you need to add to you dial command. If the telco was only sending, say, the last
What are the configuration steps to set up the polycom ip500 to where by
pressing on the messages button will activate voice mail? I suspect the xlm
language will need to be changed but where.
Also, have one main pstn line going in from main telco with cw call
notification. Every time a second
Thanks for your reply.
I put mode = insecure in the sip.conf file but I'm still getting the 407
message from asterisk when I a call is forwarded from the Mediatrix gateway.
I'm pasting my sip.conf file. Is mode = insecure right or should I use
different syntax. Thanks for your help!
[general]
Eric ManxPower Wieling wrote:
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.
This is only an issue if
This is only an issue if your SIP phone has a poor/nonexistent jitter
buffer.
I agree with that. Asterisk should just forward any RTP immediately and
let endpoints handle the jitter buffer - unless asterisk is the endpoint
itself (e.g. with phones plugged in its fxs ports).
On 4/30/06, Remco Barende [EMAIL PROTECTED] wrote:
I e-mailed Dell support and asked them if it is possibel to assign a
unique IRQ to one of the three PCI slots.
Their reply was, not possible, you are ALWAYS sharing IRQ's, I guess this
is the reason for the poor results I'm seeing.
If you're
On 4/29/06, Vidar [EMAIL PROTECTED] wrote:
Has anyone managed to add the bristuff patch to 1.2.7.1 successfully?
My attempts has ended up bad, so if anyone has a working patchfile for
1.2.7.1 I would be grateful to receive it.
Have a look at this URL: http://www.junghanns.net/downloads/
You
Fedora core stable version now is FC4 (which is to say Red hat 10
version 4 or even Red Hat 11 if we count in the old way RH did ).
IAX and the configuration have changed a bit from 1.0.3 so you'll need
to modify the file to match the new configuration but other then that it
should be no
Terence Burnard wrote:
# cat /etc/zaptel.conf
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
I don't know if this will fix your problem, but the span line should read:
span=1,1,0,esf,b8zs
You get your timing from the telco.
Doug
--
Ben Franklin quote:
Those who
Assaf Flatto [EMAIL PROTECTED] wrote:
Fedora core stable version now is FC4
Actually, it's FC5.
Fedora Core (FC) 2 and 4 radically changed the GCC/GLibC versions (as
well as the kernel in the case of 2). FC 3 and 5 were largely
revisions of the former, respectively.
(which is to say Red hat
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So, how do you know which conf files one can hand edit versus those that
might be overwritten?
You may only change the *_custom.conf files. :)
And the *_additional.conf files are the ones overwritten by the config
in the DB. So you can edit the other ones.
hth
I can not seem to configure to work with login. I thought it was pure
sip. It is unlocked. Can anyone help me
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Hi
I am trying out , auto-dail features, I want to use
auto-dail to dial a phone and on attempt , play a file
. I have made a file sample.call with contents
Channel: SIP/326
Callerid: Joseph
Application: Playback(nobody-but-chickens)
MaxRetries: 2
RetryTime: 60
WaitTime: 10
Context:
Time Bandit wrote:
And the *_additional.conf files are the ones overwritten by the config
in the DB. So you can edit the other ones.
You could, but it'll get overwritten by any FreePBX upgrades. The *.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten. The
Basically you can change the ones which don't have _additional.conf at the
end. So with your extensions.conf you can change the extensions.conf and
extensions_custom.conf
Best to keep all your configs in the custom file though.
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 9882
Hi
I had installed Cepstral , and it is working in
Asterisk , it workfine for
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Cepstral( This is Just a test )
exten = s,4,Cepstral(Hope u are getting this voices)
but instead of the text contents for Cepstral , can I
use the file name
Avi Miller wrote:
You could, but it'll get overwritten by any FreePBX upgrades. The *.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten.
I thought I should clarify this statement: I meant that FreePBX could
overwrite both the *.conf and the *_additional.conf
On Monday 01 May 2006 01:42, Terence Burnard wrote:
Module Size Used by
wcusb 21760 0
wctdm 36512 0
wcfxo 13408 0
wcte11xp 24896 0
wct1xxp16544 0
wct4xxp97664 24
tor2
Andrew Kohlsmith wrote:
First of all, don't load every Asterisk module under the sun. Load the
modules for the hardware you have, and if you're using something like [EMAIL
PROTECTED]
which loads everything, edit your /etc/modules.conf to alias the ones you do
NOT have to 'off' to
Hi,
Red Hat 9.0
Asterisk 1.2.7.1
Whenever I start Asterisk, I am unable to call out on my SIP channel:
-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]||t|) in new stack
Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such
host: 6477235412
Apr 30 11:02:01 NOTICE[12814]:
You could, but it'll get overwritten by any FreePBX upgrades. The *.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten.
I thought I should clarify this statement: I meant that FreePBX could
overwrite both the *.conf and the *_additional.conf files. You are
On Monday 01 May 2006 07:27, Kevin P. Fleming wrote:
I believe you mean _Zaptel_ module, not _Asterisk_ module, in this case :-)
Ahh yes, you are correct. Place blame where due. :-)
-A.
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Wouldn't use it in production for a customer personally. Too many
limitations in terms of having a flexible diaplan. What would be nice
though is if they were to produce a 'lite' version that gave a gui interface
to add/change/move things - sip.conf, voicemail.conf, meetme.conf but
staying
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.
Here's what I have been using to produce a callback feature. It's not
pretty but
Just saw this come across the debian bug list. Can anyone comment?
How does this affect those of us not running Debian installs?I see
it seems it even affects 1.2.7 versions (According to Debian)
Several problems have been discovered in Asterisk, an Open Source
Private Branch Exchange
OK, so this list claims to have1000's of users on it. Let's see where they are I was putting myself on frappr map for something else and found an unused asterisk map. I saw Olle on that (imagine that) but no mention of it on the lists. So.
Frappr yourselves. Let's see how many
On 04/29/06 20:15 Josué Conti said the following:
Dinesh the agents they receive a call and this call will have to be
transferred, them uses only functions hold and trnsf in device
i'm not sure how the polycom's hold and trnsf buttons are mapped, but using
blindxfer and atxfer dtmf
Hi
I was able to solve it
the modified sample.call is
Channel: SIP/326
Callerid: Joseph
Application: Playback
Data: goodbye
MaxRetries: 2
RetryTime: 60
WaitTime: 10
Context: ext-local
Extension: 326
Priority: 1
thanks
Joseph JOhn
--- John Joseph [EMAIL PROTECTED] wrote:
Hi. I have become interested in the console driver because it would
be nice to have a nice usb conference phone to use with asterisk -- my
problem is that if you are on a call you can only dial one dtmf digit
at a time -- all the subsequent digits are ignored. Any one have a
solution to this
Is there a way to monitor a call for DTMF tones an trigger some actions based on
those DTMF tones?
I am interested in any arbitrary DTMF tones, not those related to the usual PBX
functions like call transfer, music on hold, call diversion etc
/Obelix
On Mon, May 01, 2006 at 12:16:18PM +0400, Jean-Michel Hiver spake thusly:
This is only an issue if your SIP phone has a poor/nonexistent jitter
buffer.
I agree with that. Asterisk should just forward any RTP immediately and
let endpoints handle the jitter buffer - unless asterisk is the
I've enabled voice mail for extension 200 in the extensions menu, and I've set the password to 1234. When I dial *97 which is listed as Your messages in the applications menu, it says Password I enter 1234 and it says, login incorrect, Password so what am I missing?
Thanks,Jim.
Hi,
Red Hat 9.0
Asterisk 1.2.7.1
** Apologies if you notice this posted multiple times, I'm just not
seeing it on the boards **
Whenever I start Asterisk, I am unable to call out on my SIP channel:
-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]||t|) in new stack
Apr 30 11:02:00
run asterisk in verbose mode (-vv) and see if the digits are being properly picked up. A common problem is a DTMF type mismatch, so the keypresses may not be getting to the server.
-yair
On 5/1/06, Jim Lynch [EMAIL PROTECTED] wrote:
I've enabled voice mail for extension 200 in the extensions
Jim Lynch wrote:
I've enabled voice mail for extension 200 in the extensions menu,
Asterisk has no extensions menu.
I'm guessing that you are using some type of graphical interface. I
would suggest you post your question to either their forum or mailing list.
Doug
Done
There are at least 15 other in NY who have been to the NY Users
group.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Roy
Sent: Monday, 1 May 2006 8:31 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
Time Bandit wrote:
You could, but it'll get overwritten by any FreePBX upgrades. The
*.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten.
I thought I should clarify this statement: I meant that FreePBX could
overwrite both the *.conf and the *_additional.conf
We ran some tests on
our line here where a scope and voltmeter, we were able to see the waveform
changes on incoming call and the voltage at 6-7v during the conversation, but no
drop in voltage (or reverse tip/ring) when the remote side hanged-up. The
voltage went back to around 48v after
Does anyone know the secret to get the
GXP-2000 Message waiting lamp to illuminate?
Or can point me toward some docs that
might explain it?
Thanks!
--Jeffrey
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Since FreePBX is module based it seems that with all the good people out
on the internet there is someone will write an add-on to extend the
capabilities for those that need it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Monday,
FreePBX is catching up reasonably quickly.
There are still some basic things missing (for example if you don't use
voicemail it is not possible to set a destination for the call if not
answered, you have to create a ring group for each extension to work
around it, this is a major issue) and
I tried that, usecallerid=no, didn't help. I discovered that IRQ 5 is shared with an nVidia card, and that may be the problem. -brTime Bandit [EMAIL PROTECTED] wrote: The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of
It looks like it could be Asterisk generating ringing tones waiting for a fax signal.
On 5/1/06, Ryder Brook [EMAIL PROTECTED] wrote:
I tried that, usecallerid=no, didn't help.I discovered that IRQ 5 is shared with an nVidia card, and that may be the problem.-br
Time Bandit [EMAIL PROTECTED]
Make sure your using RFC2833 for DTMF mode. You can go to the server and
check by going to extensions - view extension and change by modify
extension.
This was for Snom phones anyway, but I believe that is a widely accepted
RFC.
-Original Message-
From: [EMAIL PROTECTED]
Make sure *97 is indeed your voice
mail number first. If you have remote voice-mail setup for users, you can just
try it from outside(calling in).
If that is indeed correct, then
move another phone to that extension. At least you can narrow down thether it is
the phone or the server
On 01/05/06, Jeffrey Macko [EMAIL PROTECTED] wrote:
Does anyone know the secret to get the GXP-2000 Message waiting lamp to
illuminate?
No secret - just set a 'mailbox' line in the appropriate peer entry in
sip.conf. Later GXP-2000 firmware shows the number of messages waiting
on the LCD
Yes, Sir/Madam, that was it. I, now, have immediate=yes,faxdetect=no in zapata.conf Picks up immediately. Thanks, -SBLacy Moore - Aspendora [EMAIL PROTECTED] wrote: It looks like it could be Asterisk generating ringing tones waiting for a fax signal. On 5/1/06, Ryder Brook [EMAIL PROTECTED] wrote:
I have no NAT: phones and Asterisk are on different subnets (VLANs)
connected to a layer3 switch and correctly routed
Peraphs there is some issue with VLAN tagging...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Waldo Rubinstein
Sent:
You can already do that. You ca specify different access to different users
with the Administrators module.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, May 01, 2006 6:33 AM
To: Asterisk Users Mailing List -
OK, a combination of all the suggestions has pointed me in the right direction. The phone I'm using is the Budge Tone 100. For some reason it is not sending the dtmf tones. I had set the dtmfmode in the extensions menu to RFC2833, but I chanced to recall there was a dtmf mode in the phone config
On 1/24/06, Lee Archer [EMAIL PROTECTED] wrote:
Thanks, I've got it running on my test box but didn't know if there was
any global objection to using it. I've had a few funnies with it but
that might be down to Supermicro and P4's with the EM64T thing.
Regards
Lee
-Original Message-
and rebooted the phone, it started working. Thanks to all. Does anyone
else think rebooted my phone sounds a little funny?
You get used to it after a while :P
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
How can I install a softphone on my USB flash drive like Xlite and have
it ready to go when I plug it in at any Windows XP computer?
(Same for a Linux softphone, both on one USB flash drive).
bye
Ronald Wiplinger
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How can I install a softphone on my USB flash drive like Xlite and have
it ready to go when I plug it in at any Windows XP computer?
(Same for a Linux softphone, both on one USB flash drive).
I believe Dan's softphone is suitable for this. See
http://www.laser.com/dante/diax/diax.html
Actually Harry, there is no setup needed. You can send early audio with the
Playback command by adding the noanswer parameter (see example). But the
other end of must support/offer it.
But back to my problem, I thought maybe if asterisk generated the ring tone
it might accept the early audio
I do this with the windows version of idefisk from Asteriskguru.com. The configuration is stored in the dir with the program and dll. I have actually configured it and emailed it to users. There is no installer and a simple shortcut or autoplay menu should take care of the rest.
On 5/1/06, Time
Hi All
when I try to use auto-dial to connect to
outside phone , my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use
Channel: ZAP/1/050745 in my sample.call file , if
I use Channel:SIP/326 , it works fine
my
Hi
How do i disable dialling out from voicemail?
--
Jon Farmer
Telford, Shropshire, UK
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John Joseph wrote:
Hi All
when I try to use auto-dial to connect to
outside phone , my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use
Channel: ZAP/1/050745 in my sample.call file , if
My fix for this is
Jon Farmer wrote:
Hi
How do i disable dialling out from voicemail?
It's enabled/disabled via the voicemail.conf
Doug
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Doug Lytle wrote:
It's enabled/disabled via the voicemail.conf
I have commented out
dialout=from-vm
but the option is still given even though any number dialled results in
unobtainable. So I dont want the option given.
--
Jon Farmer
Telford, Shropshire, UK
I'm having trouble getting callerid name to show up on my phones (Cisco
7960 and a few softphones)
When I look in the CDR database I see the name but not on any phone when
being called.
I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
Any help would be great !
The current versions of IDEFISK use a Windows installer,
wether it is required or not now I dont know.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk
Users Mailing List - Non-Commercial
Jon Farmer wrote:
Doug Lytle wrote:
It's enabled/disabled via the voicemail.conf
I have commented out
dialout=from-vm
but the option is still given even though any number dialled results in
unobtainable. So I dont want the option given.
You'll also need to do a stop/start of
Has anyone successfully gotten the 7941G working on Asterisk? We're
looking at getting some of those instead of the 7940's, but there's really
not much info out there about them.
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
Doug Lytle wrote:
You'll also need to do a stop/start of Asterisk.
Done that also, no difference
--
Jon Farmer
Telford, Shropshire, UK
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Does idefisk support that iogear usb phone that I think you were
talking about?
on Monday 05/01/2006 Kerry Garrison([EMAIL PROTECTED]) wrote
The current versions of IDEFISK use a Windows installer, wether it is
required or not now I dont know.
-Kerry
_
From: [EMAIL
Check out the zip version at under a meg.On 5/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
The current versions of IDEFISK use a Windows installer,
wether it is required or not now I dont know.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Bruce
Jon Farmer wrote:
Doug Lytle wrote:
You'll also need to do a stop/start of Asterisk.
Done that also, no difference
Let's see that section of your voicemail.conf
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Can i use same group in zapata.conf with 2 or more hfc-cards ?
Is this possible :
instaed of Dial(Zap/g1/10Zap/g2/10)
Dial(Zap/g1/10) - with g1 fuer both cards.
cu
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Doug Lytle wrote:
Let's see that section of your voicemail.conf
; tz=central; Timezone from zonemessages above. Irrelevant
if envelope=no.
; attach=yes; Attach the voicemail to the notification email
*NOT* the pager email
; saycid=yes; Say the caller id
Bruce Reeves wrote:
I do this with the windows version of idefisk from Asteriskguru.com
http://Asteriskguru.com. The configuration is stored in the dir with
the program and dll. I have actually configured it and emailed it to
users. There is no installer and a simple shortcut or autoplay menu
The problem is that Answer for Zap channels, means that the FXO
card has accepted the call and is in process of making the call, but
that does not means that the other end (ie. the phone in the PSTN) has
already answered. Personally I have the same problem, but I have not
looked further on the
Jon Farmer wrote:
Doug Lytle wrote:
Let's see that section of your voicemail.conf
What do you see at the console when someone presses 4 from voice mail?
Doug
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I'm still trying to accomplish the same thing and a solution might come from
Newman Telecom but I'm still waiting to see. Note: I have no affiliation
with Newman Telecom but they're application code appears to be the closest
to what I'm looking for.
The problem is because asterisk needs a way of
Hello,
I'm just starting
out with asterisk and I'm playing around with the system. Currently I have
a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP
soft phone on my laptop for testing that is working fine. When I try to
place a call from my soft phone I get this from
Doug Lytle wrote:
What do you see at the console when someone presses 4 from voice mail?
-- Executing VoiceMailMain(SIP/502-ac3f, s502) in new stack
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
--
Hello
Im new to Asterisk and generally to IPPBX.
Quick specification of my SOHO system:
1. 2 Interfaces: WAN and LAN
4. 5 VoIP phones configured with SIP attached to LAN
5. 1 external SIP line configured as gateway for VoIP phones to allow
outbound calls.
What I want is to set secure Asterisk
Hi,
You can call an agi script to convert the text file to wave format.
Example:
http://www.voip-info.org/wiki/view/swift.agi
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Joseph
Sent: Monday, May 01, 2006 7:08 PM
To: Asterisk Users
Hello
Im new to Asterisk and generally to IPPBX.
Quick specification of my SOHO system:
1. 2 Interfaces: WAN and LAN
4. 5 VoIP phones configured with SIP attached to LAN
5. 1 external SIP line configured as gateway for VoIP phones to allow
outbound calls.
What I want is to set secure
Hi,
What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Tuesday, May 02, 2006 12:37 AM
Using SIP and SCCP. The softphone uses SIP.
Doing a debug I see no name being sent.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: Monday, May 01, 2006 2:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May 01, 2006 12:37 PM
To: Asterisk Users Mailing
Rich Adamson wrote: Let's see if I can summarize various recent postings relative to the broader topic of whether FreePBX/AAH is production-ready.It's not proper to put FreePBX/AAH in the same breath. AAH puts FreePBX ontop of their build, along with a bunch of other software. Although
Jon Farmer wrote:
Doug Lytle wrote:
What do you see at the console when someone presses 4 from voice mail?
-- Executing VoiceMailMain(SIP/502-ac3f, s502) in new stack
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages'
Some more info to my problem:
ipt-dev01*CLI zap show status
Description
Alarms IRQ
bpviol CRC4
T2XXP (PCI) Card 0 Span
1
OK
0
0 0
ipt-dev01*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down,
Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
There are still some basic things missing (for example if you
don't use
voicemail it is not possible to set a destination for the call if
not
answered, you have to create a ring group for each extension to
work
around it, this is a major issue)
Remco
Looks like your D-channel is down.
Ztcfg reports all is ok, b/c as far as iut
is concerned, it is talking to your card just fine. LibPri handles the PRI
implemetaton.
Since you are able to see the pri commands
from the CLI, Isdn supprt is linked into your asterisk core.
Call your
Just a
stab:
exten
= _9NXX,1,Dial(ZAP/g1/${EXTEN})
Note
uppercase ZAP and explicitly specifying the dialled number.
hth
-Original Message-From: Dan Brummer
[mailto:[EMAIL PROTECTED]Sent: Monday, May 01, 2006 12:48
PMTo: Asterisk Users Mailing List - Non-Commercial
Thank you for the reply Alexandar.
After I restarted my dev machine I recieved these messages
from asterisk:
chan_zap.c:2290 pri_find_dchan: No D-channels
available! Using Primary channel 24 as D-channel
anyway!
chan_zap.c:8202 pri_dchannel: PRI got event: No more alarm (5) on Primary
Title: Re: [Asterisk-Users] Problems with zaptel and TE210P
Shouldnt your zapata.conf be
span=1,1,0,esf,b8zs
As it stands, you are not taking timing from the PRI. Changing the second digit of the span entry to 1 will tell Asterisk to use that line as the clock master.
HTH,
James
On
On 5/1/06, Obelix [EMAIL PROTECTED] wrote:
Is there a way to monitor a call for DTMF tones an trigger some actions based on
those DTMF tones?
I am interested in any arbitrary DTMF tones, not those related to the usual PBX
functions like call transfer, music on hold, call diversion etc
take a
His PRI span is showing down, If you
forget to add the ${EXTEN} as you said it would show as connecting and he _should_ get an intercept from the telco.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Monday, May 01, 2006 2:53 PM
To:
Did your Telco tell you what switch they
are using?
Also change your timing settings in the /etc/zaptel.conf
file to have this PRI do primary timing.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 2:59 PM
To: Asterisk
On Mon, 1 May 2006, Kerry Garrison wrote:
There are still some basic things missing (for example if you
don't use
voicemail it is not possible to set a destination for the call if
not
answered, you have to create a ring group for each extension to
work
around
Evening all.
I'm looking at building an Asterisk system for one
of the projects of the Charity where I'm the
SysAdmin.
The project has two analogue phone lines - BT
Featureline Compact, we're in the UK - that I'd
like Asterisk to handle.
My current quandary is which FXO interface to use.
I've
Steve Totaro wrote:
Not sure if cheesy is the right word. Sound solution may be a better
adjective. Adding two NICs, one to each machine and connecting them
directly via crossover cable on a totally separate network may be my
best solution. No FTP traffic would even hit the NIC or the
Bruce Reeves [EMAIL PROTECTED] wrote:
I use teliax.com and exgn.net to do my initial test
of toll free calls into
my system.
How's your experience been with their audio quality,
and with their inbound call completion reliability?
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