Darryl Dunkin wrote:
It's not playing a wav file at all, it
is mixing the live audio from all of the callers in that conference
room and sending it back out to them.
I understand. What I tried to say is that if a wav file can be played
at the correct speed, why would a
Kevin P. Fleming wrote:
The code is done and initially tested; it is being reviewed internally
and should be available on Friday or Monday.
Will this code be available in 1.2 and 1.4 versions alike ?
I can testify it's needed in 1.2.
Best regards
Tzafrir Cohen wrote:
On Thu, Apr 19, 2007 at 09:23:48PM +0200, Theo Band wrote:
Eric "ManxPower" Wieling wrote:
In the zaptel source "make config" will install the zaptel init script
in /etc/rc.d/init.d for many distros.
Thanks. This was the
Hi,
Thank you for your response. As you said, I set it for -5. But, its displaying
wrong time. I don't enter any SNTP Server. Is it must? How can I solve this
problem? Can you tell me?
Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote:v\:*
Hi Noah,
Thank you for your response. As you said, I tried to enter -18000 in GMT
offset field. But, its not taking input from the phone dial pad or key board.
Its giving chance to select the value from -12 to 12. I dont enter any SNTP
Server. Is it must? How can I solve this problem? Can
On Wed, 18 Apr 2007, Joseph wrote:
Are there any cell phone (gadgets) that can be connected to standard
switch phone network? (ability to check email would be a plus).
Digium adapter S101i can be connected to any network and it allow a
standard phone to act as your local extension over the
On Thu, 19 Apr 2007, Zoilo Gomez wrote:
Am I the only one using the GXP2000 expansion module?
I hope not ... I'm looking into using one myself for a client soon - it
would be nice to know that someone has had some success with one...
Or can anyone suggest something similar - I need a
On Wed, 18 Apr 2007, Joseph wrote:
Are there any cell phone (gadgets) that can be connected to standard
switch phone network? (ability to check email would be a plus).
Digium adapter S101i can be connected to any network and it allow a
standard phone to act as your local extension over
Hans Witvliet wrote:
On Wed, 2007-04-18 at 17:11 -0400, Dean Collins wrote:
Hi guys,
I know it’s a little off topic but……Wondering if you can help.
My wife has been asked to find a writer to produce a story on “The
dramatic ramifications of IPV6 on commercial businesses and how it
will
Remco Post wrote:
Hans Witvliet wrote:
The only obstacles currently, are the ISP's.
Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well
as an ipv4 address.
Not around here (Zurich, Switzerland) they won't. I think there is one
single provider with IPV6 as an option.
Hi everyone,
I have the project to personalize the voicemail's IVR. During the
intro message (when you call the voicemail of someone), Asterisk pronounce
the number of the personal extension number by number (like that :
0.1..2.3.) and I would like it pronounce it by couple of
Crazy Boy wrote on 4/19/07 11:41 PM:
Thank you for your response. As you said, I set it for -5. But, its
displaying wrong time. I don't enter any SNTP Server. Is it must? How
can I solve this problem? Can you tell me?
Yeah, there's no way to set the clock except by using an NTP server, so
you
Hi, this code is for italian time is inside the sip.cfg file.
SNTP
tcpIpApp.sntp.resyncPeriod=86400
tcpIpApp.sntp.address=192.168.0.8
tcpIpApp.sntp.address.overrideDHCP=0
tcpIpApp.sntp.gmtOffset=3600
tcpIpApp.sntp.gmtOffset.overrideDHCP=0
Or can anyone suggest something similar - I need a console with about
25-30 buttons/lamps, sourced in the UK ...
I've just had an Aastra 57i w/ LCD sidecar delivered for testing... initial
results look very promising. Might be worth looking into those if you want an
alternative to the
Hello again,
Mark Spencer will be joining us for questions on at least one of these
conferences (we've discussed this and he is definitely onboard with
the concept), but his schedule is such that we can't *promise* he'll
be there this week. I haven't heard back from him about today, but
I'll try
I use realtime. Both information and extensions are stored in DB. It
is just a simple setting of the user with dial plan Dial([EMAIL PROTECTED]).
exten = 9003,1,Dial([EMAIL PROTECTED])
What I found is the following.
9002 --- S1 --- S2
9002 can make request to S1 and S1 forward the request to
Ok, I've made all those changes, called my operator from an outside line
and tried alternatively whispering / shouting into the mic, banging the
microphone with a metal object and pressing DTMF digits.
So far - so good, it seems to work.
I've now got an other problem. Clamping DTMF disabled my
On Fri, 20 Apr 2007, Cosmin Prund wrote:
Ok, I've made all those changes, called my operator from an outside line
and tried alternatively whispering / shouting into the mic, banging the
microphone with a metal object and pressing DTMF digits.
So far - so good, it seems to work.
I've now
I've implemented my IVR using an FastAGI thing, using the READ
application. core show application read shows no information on how
the read function gets it's digits, I assume it does it the right way.
With DTMF clamping off it works, with DTMF clamping on it no longer
works. I've also toggled the
Hi,
I'm looking for somebody who has managed to get their IPmux8 or
IPmux11 talking to an Asterisk machine. I have it setup properly I
think (the two IPmux's are talking to each other, and the zttool
says that the PRI is acting okay, but I'm flooded with HDLC aborts
and FCS problems.
Edwin
--
On 08:32, Fri 20 Apr 07, Gordon Henderson wrote:
On Thu, 19 Apr 2007, Zoilo Gomez wrote:
Am I the only one using the GXP2000 expansion module?
I hope not ... I'm looking into using one myself for a client soon - it
would be nice to know that someone has had some success with one...
Or
Hi Dovid,
Thanks for ur reply. Any problem with using Asterisk 1.2.13?
Thanks,
rani
On 4/20/07, Dovid B [EMAIL PROTECTED] wrote:
1) Why arent you using 1.2.17 ?
2) So you have to use an AGI ? You can use the mix monitor command.
- Original Message -
*From:* selva rani [EMAIL
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.
Also, could the calls go into the cdr for retrieval/browsing later?
What
On Fri, 20 Apr 2007, Cosmin Prund wrote:
This message includes two snips of CLI output, with DTFM CLAMPING ON
(first) and with DTFM CLAMPING OFF (second). You can search for **
to skip to the second CLI Output. In the first case I've enterd 6 DTFM
digits (123456), you can see them in a CLI
On 4/20/07, Olivier [EMAIL PROTECTED] wrote:
2007/4/20, asterisk [EMAIL PROTECTED]:
Hi,
Does anyone know if it is possible to plug a tdm400p pci digium card
into an pci-e 16x slot ?
np
Olivier meant no here as well.
___
--Bandwidth and
About once a week or so my Asterisk box stops responding to all phones.
I can pull up the console, do whatever I want at the CLI but the only
way to get things working again is to restart Asterisk altogether.
I finally cranked verbose debugging way up (and watched my log files
go from 1mb/day
Hi,
in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.
thanks
Yes,
As I have mentioned below I tried the link
http://bugs.digium.com/view.php?id=6683nbn=24
but was not able to make it work.
Regards,
Sanjay Rajdev
- Original Message -
From: Steve Murphy [EMAIL PROTECTED]
To: sanjay rajdev [EMAIL PROTECTED]
Sent: Thursday, April 19, 2007 4:37:39 AM
Remember the big lie with Verizon is not The tech will be there at
noon. It is Your FOC (Firm Order Commit) date is xxx The dates they
give are neither Firm or Committed. Just ask them.
As long as you remember, they are the Phone Company and you are just the
customer. no body will be
Hi everybody,
I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm
using Trixbox..).
I must be as fast as a flash to press *2 and do an attended transfer. If I
wait only a tenth of a second nothing happens.
I think it is an issue. I have seen the source code and found
On 20/04/07, John Treble [EMAIL PROTECTED] wrote:
Gavin,
Call Endance and ask them about their Lawful Call Intercept solution(s)
using their DAG TDM E1 cards on Linux (Endance.com).
Thanks, will have a look.
Cheers.
John Treble
Ottawa, Ontario, Canada
-Original Message-
Hi all,
I'm having a problem with a polycom 301 not picking up a ZAP call.
Below is the CLI output of the call. I have:
TDM400 with 2 FXO lines
Asterisk 1.2.14
Polycom 301
When I dial the first ZAP line, I choose an extension that rings the
polycom, polycom rings and I can pick it up and
Gavin,
Call Endance and ask them about their Lawful Call Intercept solution(s)
using their DAG TDM E1 cards on Linux (Endance.com).
Cheers.
John Treble
Ottawa, Ontario, Canada
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Steve Totaro wrote:
They are all terrible in their own way. Don't you have someone you can
delegate the Verizon babysitting responsibility to? I would consider
sales calls a little more important than being a babysitter.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
On Fri, 20 Apr 2007, Mauro Zanin wrote:
Hi everybody,
I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm
using Trixbox..).
I must be as fast as a flash to press *2 and do an attended transfer. If I
wait only a tenth of a second nothing happens.
I think it is an
Hi Folks.
I have installed two sip phones and two PCs in a network. The later with
iaxComm. Calls are made between the sip phones and between a sip phone and a
PC.
When calling from one PC to the other the iaxComm shows ??? in the status
column and the call can't be answered.
The same goes
Hi, folks:
Yesterday I added a second TDM400P card to a working, echo-free server
running HPEC.
Today, I'm getting these messages:
Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to enable
echo cancellation on channel 3
along with complaints of severe echo. The channels
Any ideas on this?
Closest thing that comes to mind is FOP : http://www.asternic.org/
hth
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Noah Miller wrote:
Had an appointment for these schmoes to come out and install another
line. Was supposed to be 8-12. Its now 6PM and not even call.
Missed
3 sales calls waiting on these jerks.
No wonder customers were jumping ship to Vonage.
I once had to oversee Verizon install a PRI
I make a habit of just buying hamburgers and stopping by the CO or hut
where I see the vans. I tell them that I am buying favors with food
and they like it.. Its a lot of work but it helps
On 4/20/07, Steve Totaro [EMAIL PROTECTED] wrote:
Noah Miller wrote:
Had an appointment for
Thanks a lot for the fix Humberto.
On 4/18/07, Humberto Figuera [EMAIL PROTECTED] wrote:
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as
2007/4/20, asterisk [EMAIL PROTECTED]:
Hi,
Does anyone know if it is possible to plug a tdm400p pci digium card
into an pci-e 16x slot ?
np
Is there a possibility to work?
no
I have a sun fire x2100 which doesn't have pci slots.
Does Digium make pci-e cards?
not yet
Hi,
Does anyone know if it is possible to plug a tdm400p pci digium card
into an pci-e 16x slot ?
Is there a possibility to work?
I have a sun fire x2100 which doesn't have pci slots.
Does Digium make pci-e cards?
___
--Bandwidth and Colocation
Hi,
I`m looking for boards to use with Asterisk and as I already was used Pika
boards few years ago (in a Windows IVR application), I found that they have
new options to Asterisk. It will be nice if I can see some opinions from
here before go ahead on it.
Thanks in advance!
Peter
(*) I`d put
1) Why arent you using 1.2.17 ?
2) So you have to use an AGI ? You can use the mix monitor command.
- Original Message -
From: selva rani
To: asterisk-users@lists.digium.com
Sent: Friday, April 13, 2007 8:03 AM
Subject: [asterisk-users] Recording and Conferencing
Hi,
I
List,
I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication
between the phones is G.729, and my sip.conf looks like this:
disallow=all; First disallow all codecs
allow=g729 ;
allow=gsm
allow=ulaw
allow=alaw
However, I cannot
I did this with a Nortel MICS a few years ago. No problem.
The dialplan was something like:
[incoming]
exten = _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten = _X.,2,Monitor(filename)
exten = _X.,3,Dial(Zap/G2/${EXTEN})
I (have to) would like to move my agents out of agents.conf in
preparation for the deprecation of agentcallback login. Everyone I have
spoken to is upset about this but the functionality can be accomplished
in the dialplan and that is fine by me. I do have an issue with losing
the features
On 4/19/07, Zoilo Gomez [EMAIL PROTECTED] wrote:
Am I the only one using the GXP2000 expansion module?
I'm using one but I'm not terribly happy with it. With firmware 1.1.3.1 the
phone wouldn't boot, and with 1.1.3.2 having the buttons configured for BLF
caused complete lockups on the phone
This could go on forever, I mean take your pick Verizon, Att, Bell South
any of them. Same story We are the phone company, who else can you call?.
We have time and again seen it take weeks to get the order documents
created, not the actual order, just the paperwork to create the order. I
Has anyone found a softphone that supports pulling it's configuration from a
central server via TFTP/FTP/HTTP, much like hard desk phones use?
I'm looking for something for a call center that I can provision from a
central location by generating config files. If the phone has soft keys
(yes, I
On Fri, 2007-04-20 at 15:37 +0300, asterisk wrote:
Hi,
Does anyone know if it is possible to plug a tdm400p pci digium card
into an pci-e 16x slot ?
Is there a possibility to work?
I have a sun fire x2100 which doesn't have pci slots.
Does Digium make pci-e cards?
Can you insert
Stephen Bosch wrote:
Hi, folks:
Yesterday I added a second TDM400P card to a working, echo-free server
running HPEC.
Today, I'm getting these messages:
Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to
enable echo cancellation on channel 3
along with complaints
Hi,
I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at Ringall
I checked the queues.conf file and the settings
Theo,
I'm glad my reply was helpful to you. The responses pointed out that
it's time for me to update my procedures and documentation, so I'm
benefiting as well. My thanks go out to everyone who participated in
this thread.
Thank you,
Matthew Roth
InterMedia Marketing Solutions
Software
Alan Bunch wrote:
Remember the big lie with Verizon is not The tech will be there at
noon. It is Your FOC (Firm Order Commit) date is xxx The dates they
give are neither Firm or Committed. Just ask them.
As long as you remember, they are the Phone Company and you are just the
customer. no
I once had to oversee Verizon install a PRI line in Manhattan. I live
2.5 hours away, but we made the appointment, and I was there, but the
Verizon tech never showed. I made another appointment, and it
happened again, and again, and again. I don't even remember how many
times it finally took,
On 20/04/07, David Gomillion [EMAIL PROTECTED] wrote:
I did this with a Nortel MICS a few years ago. No problem.
The dialplan was something like:
[incoming]
exten = _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten =
2007/4/20, James FitzGibbon [EMAIL PROTECTED]:
Other than that, I'm back at X-Lite/eyeBeam, which stores it's configs in
binary files, preventing me ...
--
j.
James,
Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.
Regards
lsmod | grep ^zaptel
lsmod | grep ^zaptel
zaptel183076 2 zttranscode,wctdm
Cameron
___
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Hi Mauro;
Try to add featuredigittimeout = 1500 at features.conf in the [global] section.
On 4/20/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 20 Apr 2007, Mauro Zanin wrote:
Hi everybody,
I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm
using
On 4/20/07, Olivier [EMAIL PROTECTED] wrote:
Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.
Having looked into it further, you're right. For some inexplicable reason
it's not putting the files where the manual says they should be - instead of
a directory
On 20 Apr 2007, at 17:10, Kenneth Padgett wrote:
I once had to oversee Verizon install a PRI line in Manhattan. I
live
2.5 hours away, but we made the appointment, and I was there, but the
Verizon tech never showed. I made another appointment, and it
happened again, and again, and again.
Can anyone tell me which config file tells the phone what file to load
as bootrom.ld?
Or is this hardcoded in the phone? I just got a IP501 but I have a
bunch of IP500s...
Will the bootrom (2.6.2) work OK with both the IP500 and 501?
Thanks!
Wiley E. Siler
Director of Information
On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote:
On 4/20/07, Olivier [EMAIL PROTECTED] wrote:
Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.
Having looked into it further, you're right. For some inexplicable reason
it's not putting the files where
On Fri, Apr 20, 2007 at 02:59:28PM +0200, Michael Landin Hostbaek wrote:
List,
I have some cisco phones (7940) and asterisk 1.4 running nicely..
Communication
between the phones is G.729, and my sip.conf looks like this:
disallow=all; First disallow all codecs
transcode out of .gsm?) I am not sure what parts of the system are
enabled/disabled without the licence.
This mentions voicemail g729 in pass-thru mode. I'm not sure if it works as
I've never tried it, but it may be worth a try...
Hi All,
I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth). We've no E1/T1 links, everything is IP based.
My boss complains that many of the calls he
Hi
Also - are there any useful stats/logs that I can examine to see the
quality of calls?
You didn't mention that you have any QOS on your router, so we can
basically guarantee that your problem is the internet connection.
Remember that all the research on networking has been how to
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth).
Remember in computer terms this means that you used 100% of the
connection, 50% of the time Your voice will loose out against the
big data packets and spoil the voice quality big time
Andrew Latham wrote:
I make a habit of just buying hamburgers and stopping by the CO or hut
where I see the vans. I tell them that I am buying favors with food
and they like it.. Its a lot of work but it helps...
This is by far the most effective way of getting something done with a
telco.
Try turning the jitterbuffer off, I found that often the endpoints can
do better on their own.
On 20 Apr 2007, at 19:01, Adrian Marsh wrote:
Hi All,
I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre
Hi, everybody:
Stephen Bosch wrote:
Kevin P. Fleming wrote:
Eric ManxPower Wieling wrote:
Any updates on this?
The code is done and initially tested; it is being reviewed internally
and should be available on Friday or Monday.
Under what circumstances would this clipping be present? Is
Per Jessen wrote:
Remco Post wrote:
Hans Witvliet wrote:
The only obstacles currently, are the ISP's.
Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well
as an ipv4 address.
Not around here (Zurich, Switzerland) they won't. I think there is one
single provider with
Tim Panton wrote:
Ah, back in the old days our government privatized the state monopoly
(BT) intact (attitudes and all).
As one of the conditions they had to deliver within 6 weeks of order.
So I ordered a data line to my house (ok a bit obscure in those days, but
I needed it). 6 weeks
Arturo Ochoa wrote:
Hi List...
I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
and it also has the echo canceller...
I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
2.6.9-34.0.2.EL
I'm using Polycom's 501 with the SIP 1.6.2.0041
The problem is when
I've got a system with a TE412P installed under Fedora Core 6 and I continue to
see this message in the logs. The card most certainly does have an EC module
installed. The system is suffering from echo problems, and I suspect this is no
coincidence... I've double checked to ensure the module
Hi guys,
I'm trying to implement STUN support in *, is there anyone here which
have any experience in something like that?
I've got the STUND and I'll try to buld a patch or something for sip.
Any ideas or existing implementation would be nice. I know openpbx have it.
Hi Wiley -
Can anyone tell me which config file tells the phone what file to load as
bootrom.ld?
Or is this hardcoded in the phone?
Yup, it's hardcoded. I believe this is the way it works: If there's a
bootrom.ld on your configuration server, and it is newer than the one
on the phone, the
I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
and it also has the echo canceller...
I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
2.6.9-34.0.2.EL
I'm using Polycom's 501 with the SIP 1.6.2.0041
The problem is when someone dials to or from the PSTN
I was just watching the informational video on cisco's web site about
the 7921G and they guy mentions that the phone is running Linux. Anyone
know if they've released the source code?
This page confirms that the phone is running Linux
Hi Shawn -
We have several Polycom 500/501/601's on both a LAN and at employee homes.
The problem we are having is if our internet connection goes down the Local
LAN phones loose their connection to the Asterisk Server.
I've checked everything I can think of but can't figure out why its
Hi all,
This is slightly off-topic, but I was hoping to be able to receive some
insight as I'm sure plenty of experts with c7960's exist on this mailing
list.
I'm attempting to upgrade from SIP 8.3 - 8.6 on a C7960G that I
inherited. I have my TFTP setup and unfiltered. The phone is doing TFTP
From: Steve Davies [EMAIL PROTECTED]
Date: Fri, 20 Apr 2007 18:26:57 +0100
On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote:
On 4/20/07, Olivier [EMAIL PROTECTED] wrote:
Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.
Having looked into it further,
Hi All,
Im just getting started in the asterisk world and im wondering if anyone
can point me in the right direction towards getting asterisk working
from my house to my asterisk server in my colocation facility.
Thanks
--Don
___
I forgot to add the hardware. Im using Gentoo Linux a Pix 515
Thanks
--Don
From: Don E. Wisdom
Sent: Friday, April 20, 2007 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Asterisk PiX devices
Hi All,
Im just
Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels
like this
In zaptel.conf
fxsks=1-4
fxsks=5-8
fxsks=9-12
loadzone=us
defaultzone=us
in zapata.conf
in channels section
context=incoming
signalling=fxs_ks
channel = 1-4
channel = 5-8
channel = 9-12
when i run ztcfg
From: Arun Kumar [EMAIL PROTECTED]
Date: Fri, 20 Apr 2007 17:58:10 +0400
Hi,
in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
Don E. Wisdom wrote:
Hi All,
Im just getting started in the asterisk world and im wondering if
anyone can point me in the right direction towards getting asterisk
working from my house to my asterisk server in my
Hi,
I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at Ringall
I checked the queues.conf file and the settings
Do you have fxs modules or fxo modules? PSTN connects to fxo, but the
signaling is fxs like you have.
On 4/20/07, Ricardo Melendez [EMAIL PROTECTED] wrote:
Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap
channels like this
In zaptel.conf
fxsks=1-4
fxsks=5-8
fxsks=9-12
Do you have fxs modules or fxo modules? PSTN connects to fxo, but the
signaling is fxs like you have.
On 4/20/07, Ricardo Melendez [EMAIL PROTECTED] wrote:
Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap
channels like this
In zaptel.conf
fxsks=1-4
fxsks=5-8
fxsks=9-12
Hi,
I am working on developing a professional Marketing Materials for my
systems.
I plan on using a very good(expensive) company to do that so splitting the
costs with several people would be nice.
Let me know if you are interested on taking part in it.
robert
Sorry I should clarify. I need to pass sip traffic thru the pix to the
asterisk server. (from sip phones at my house and wherever else I might
be) The pix has 7.2.2 os
--Don
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Mendoza
Sent: Friday,
Noah Miller wrote:
I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's,
and it also has the echo canceller...
I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel
2.6.9-34.0.2.EL
I'm using Polycom's 501 with the SIP 1.6.2.0041
The problem is when someone dials
set_format: Unable to find a codec translation path from ulaw to g729
Both endpoints are PAP2 set to G711 only
I have 1.2.17 on Suse 10.1
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--- Barton Fisher [EMAIL PROTECTED] wrote:
Looks like:
amaflags=billing
switchtype=national
is being carry-over from prior PRI.. (All PRI stuff) Try
moving below
before the first PRI?
Thanks all, I tried the:
T1 as port 1 and then the PRI as ports 2 and 3 but zap
dumped again. I
On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:
set_format: Unable to find a codec translation path from ulaw to g729
Both endpoints are PAP2 set to G711 only
Where precisely are they so set?
--
Alex Balashov [EMAIL PROTECTED]
Are your agents logged into the queue?
-brandon
Tim Verscheure wrote:
Hi,
I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also
Sip Debug,
But I can tell you now that one of them is requesting g729, or, asterisk
has g729 set for one of its codecs in sip.conf and needs to translate it.
grep -r g729 /etc/asterisk/*
Alex Balashov wrote:
On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:
set_format:
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