2006/4/17, Nicholas Kathmann [EMAIL PROTECTED]:
I agree with Lee.I have about 30 machines in production using iaxmodemand hylafax which work perfectly.Most are running off of T1s, but someare on TDM400 and TDM2400s.I only use IBM servers (which are about
twice the cost for the low end Dells), and
Is there any h323 channel driver that supports DTMF inband signalization?
Thank you for your answer!
--
Tomislav Parcina
tparcina#lama.hr
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Simone wrote:
I want to thank you for the suggestions. The office is in the UK, so
probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for
the line so that bandwidth should not be a problem, the internal LAN
will be Gbit as said so the QoS as suggested will be only on the
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network
can register with specific user?
The thing is that I can't use password and I can't use host=ip.of.my.phone. And
I have to be sure that no one, from Internet will register on my * like that
user.
So, please tell me
On 18 Apr 2006, at 03:20, stevanus wrote:
Hmm...my output for getconf GNU_LIBPTHREAD_VERSION is NPTL 2.3.4..
I don't know what it's mean anyway :P
And for Lee, I'm configuring my asterisk through amp (now freepbx),
but I do some custom configuration manually too ;)
I guess Paul is right,
Tomislav Parčina wrote:
Is there any h323 channel driver that supports DTMF inband signalization?
Thank you for your answer!
The native H.323 driver, chan_h323, does support inband DTMF.
Good Luck,
Jeremy McNamara
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On Tue, 2006-04-18 at 08:29 +0100, Tim Panton wrote:
I'd guess you have a startup script for asterisk that is setting the
LD_ASSUME_KERNEL environment variable.
To check, find the 'main' asterisk process id (almost always the
lowest numbered asterisk process)
then look (as root) in the
Any thoughts as to why only 1 of my boxes has this problem? I'm on a
2.6 kernel so any more ideas?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 18 April 2006 09:00
To: Asterisk Users Mailing List - Non-Commercial
On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote:
Any thoughts as to why only 1 of my boxes has this problem?
Is it really a problem?
I'm on a
2.6 kernel so any more ideas?
Can someone answer what was the original purpose of the
export LD_ASSUME_KERNEL=2.4.1 in the asterisk script?
if you need a slim MTA to replace sendmail, in a server that's only acting as an * server and need anything else, you could try nullmailer, a small MTA only capable of smtp via a smarthost. It's so little it sould be considered, at least on embedded-systems.
We use it with success...
2006/4/18,
Yes it is a problem cos after a while of just leaving it the system is
unable to make calls out via the PSTN, which is why I have spent time
with the telco, more like wasted time, and played with zaptel's make
options. After trying a few things I came to the temporary conclusion
that it was
On Tue, 2006-04-18 at 09:33 +0100, Lee Archer wrote:
Yes it is a problem cos after a while of just leaving it the system is
unable to make calls out via the PSTN, which is why I have spent time
with the telco, more like wasted time, and played with zaptel's make
options. After trying a
I've tried cat /proc/*asterisk proc number*/environ | strings | grep
LD_ASSUME_KERNEL and it returns nothing..:(
And just for confirmation : I had the same problem as Lee had (unable
to make calls out) :(
Regards,
Stevanus
Lee Archer wrote:
Yes it is a problem cos after a while of just
Hi all,I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work!
I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1
I forgot to write: When i hangup the call, it hangs correctly!On 4/18/06, Marco Mouta [EMAIL PROTECTED]
wrote:Hi all,I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the
All I can figure is that something I haven't yet figured is causing
these processes to be created, and after a while there is so many that
outgoing calls over zap can't be made. It only applies to 1 system out
of 7, running Suse 10 and a 2.6 kernel.
Lee
-Original Message-
From: [EMAIL
Hi
Check this setting:
bindaddr = 0.0.0.0 :IP Address to bind to (listen on)
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Par?ina
Sent: Tuesday, April 18, 2006 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hallo!
Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102.
Looks like none of them works with Mac mini G4...
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In sip.conf, how can I define that only IP phones from 192.168.0.0/24
network can register with specific user?
The thing is that I can't use password and I can't use
host=ip.of.my.phone. And I have to be sure that no one, from Internet
will register on my * like that user.
So, please
2006/4/18, Tomislav Parčina [EMAIL PROTECTED]:
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network
can register with specific user?
The thing is that I can't use password and I can't use host=ip.of.my.phone.
And I have to be sure that no one, from Internet will
Hi Ivan!
Thank you. I have already find it on
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask
Problem is that they didn't define this in sip.conf.saple where I first have
take a look. This should be fixed.
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL
Hi Alejandro!
I have solved my problem. Look at mail above.
One more thing, what you have suggested it's not an option. I have allowed that
people from Internet can call me. So, I can't define any rule that will protect
me the way you have mention.
Thank you for trying.
--
Tomislav Parcina
Hi Alberto!
Have you try it, do you use it? I ask because I was in contact with developers
of ooh323 channel driver and they have told me that they can't grantee that it
will work...
I'm using oh323, version 0.67, and INBAND signalization doesn't work for me. I
have an issue with version 0.73
Hi Jeremy!
I have noticed than almost nobody uses native H.323 driver. All that I have
read about them is complaining that they don't support various stuff. Do you
use h323 in production?
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Tomislav
- How to restart the phone? (On 7960 it is *+6+Settings)
- How to setup working dtmf?
- How to setup hinting?
For line is
line button=4
featureID9/featureID
...
For speeddial is
line button=5
featureID2/featureID
featureLabel341/featureLabel
speedDialNumber341/speedDialNumber
/line
How to
On 18 Apr 2006, at 09:27, Dave Cotton wrote:
On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote:
Any thoughts as to why only 1 of my boxes has this problem?
Is it really a problem?
I'm on a
2.6 kernel so any more ideas?
Can someone answer what was the original purpose of the
export
Hi,
I am using Asterisk with misdn connected to an ISDN Line, so I have
several numbers I can use...
I know that I can use misdn like this in my extensions.conf:
exten = _0.,1,Dial(mISDN/1/${EXTEN:1})
But how can I use another number/MSN of my ISDN connection... it always uses
the default
I'm running 1.2.3 and that seems to be the most stable version, had problems with other versions too.
-- Original message -- From: "Brian Roy" [EMAIL PROTECTED]
On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote:
I'm running 1.2.4 and just about every call is cut
I recently posted a question RE the Sipura 941 and using
different ring tones, Thanks to hads I managed to use
SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's
didnt seem to do the trick?? Has anyone one had any luck on this topic?
Also havent been able to
Hello Fellow
Users,
This is my first
post so please be kind :-)
I am using
asterisk@home with freePBX talking to an
Alcatel 2400 analogue portvia a Digium TDM01B - 1 FXO
card
I can make calls
into theasterisk by dialling theasterisk extn number from the
4200.
Ican dial
extns on the
Try 1.2.3, it works fine.
-- Original message -- From: "James Sturges" [EMAIL PROTECTED] I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once a day for 1 second, calls getting
See
http://bugs.digium.com/view.php?id=6457
On Tue, 2006-04-18 at 13:11, [EMAIL PROTECTED] wrote:
I'm running 1.2.3 and that seems to be the most stable version, had
problems with other versions too.
-- Original message --
From: Brian Roy [EMAIL
Greetings;
I've got a situation where I really need to be able to select from a
number of various (body) textx for email/pager messages when voivemails
are left, but can't seem to figure out how to change them on the fly.
Now, I know there are pre-defined defaults, which can be overridden by
Just for shits and giggles, have you tried using a cross over cable? I'm
not saying it's gonna work because everything I read says you're doing
the right thing but it's worth a try.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Dmitry Ivanov wrote:
Hallo!
Anyone tried connect PC
There is no way to do it currently on these phones. Although the web
interface supports a distinctive ring based on callerID it does not
accept wildcards.
When I contacted Grandstream about this I got the following reply :-
Unfortuantely, not with present firmware.
We will implement this
Do your (wonderful) RPMs install also on CentOS?
I suppose so because it is a Red Hat clone...
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Axel Thimm
rpms for Fedora Core 1-5, RHEL 3-4 and RHL 7.3-9 have been updated:
Olivier Krief wrote:
2006/4/17, Nicholas Kathmann [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
I agree with Lee. I have about 30 machines in production using
iaxmodem
and hylafax which work perfectly. Most are running off of T1s,
but some
are on TDM400 and TDM2400s. I
On Tue, Apr 18, 2006 at 03:12:27PM +0200, Mimmus wrote:
Do your (wonderful) RPMs install also on CentOS?
I suppose so because it is a Red Hat clone...
Yes, the RHEL builds will work on CentOS, WhiteBox, SL etc. by
definition :)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
The PC port on a BT-102 should work with any computer that has an
Ethernet card. Have you tried these phones with other computers than
the Mac Minis you mention? It shouldn't make any difference whether
the computer is a Mac, PC or anything else. Perhaps something is wrong
with the BT-102s you
Nicholas Kathmann wrote:
Both hylafax and * are on the same machine and using the same PSTN
interfaces (whether T1 or TDM). It uses iaxmodem to communicate
between the two systems (imagine a softmodem). I'll create
When someone hangs up before getting to the leave voicemail prompt,
asterisk still attempts to record a voicemail message, so I end up
getting a bunch of empty voicemails.. Is there any way to change this
behaviour, so asterisk realizes that the channel has been
disconnected, and does not attempt
Are you seeing link on either end? Not sure if the GS shows or not.
On the Mac, open a terminal window and type ifconfig to see if the
port is active - ie has link
it should have a line similar to this if so
media: autoselect (100baseTX full-duplex) status: active
If this is
Hi everyone:
My escenario is:
Meridian PBX (Connected to the PSTN) is connected to Asterisk-1 via PRI T1,
the Asterisk 1 is connected to Asterisk 2 via IAX trunk or SIP trunk, the
Asterisk 2 is used for predictive dialer with Answer Machine Detector, so
for some reason, the AMD is starting
increase your silence setting
On Apr 18, 2006, at 8:31 AM, Mike Garey wrote:
When someone hangs up before getting to the leave voicemail prompt,
asterisk still attempts to record a voicemail message, so I end up
getting a bunch of empty voicemails.. Is there any way to change this
behaviour,
It doesn't seem as much broken as just annoying. I am holding off on
upgrading until this resolves, but it doesn't seem to affect performance,
anyways. BTW, some folks say that the server address only gets appended to
the CID when a redirect or something comes about. Our experience here shows
that
So how do you get a Polycom phone to work with * over NAT? I can't seem to get
it to work. If I forward ports, I can get one-way audio, but that’s it.
Looking at a packet capture, it appears that my phone is trying to send data to
the internal address of the * server, which is of course, not
Andrew Kohlsmith wrote:
On Monday 17 April 2006 07:44, Rich Adamson wrote:
I don't believe you will ever get POTS - FXO-TDM400P-to-anything to
work properly due to TDM card limitations. So, move all of those to the
bottom of your list.
I *had* this working.
POTS - TDM400
TDM400 -
I don't have any experience with that specific phone, but i have a little
experience with Grandstream ATAs.
Check web configuration to see is phone in gateway or bridge mode. It's just
a guess since you haven't provide detailed info ;)
What do you mean when you say that none of them works with
Title: IVR: playing multiple streams simultaneously?
Hi all,
I'm setting up an IVR using Asterisk.
Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for
On Tuesday 18 April 2006 09:57, Sean Garland wrote:
So how do you get a Polycom phone to work with * over NAT? I can't seem to
get it to work. If I forward ports, I can get one-way audio, but that’s
it. Looking at a packet capture, it appears that my phone is trying to
send data to the
Hello,
I read the polycom microbrowser post here
http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser
Can we access a webmail application like horde/imp or
others (which ones) to read and listen voicemails ,
send e-mails, ... ?
Regards
Harry
On Tue, 2006-04-18 at 15:12 +0200, Mimmus wrote:
Do your (wonderful) RPMs install also on CentOS?
I suppose so because it is a Red Hat clone...
There are asterisk 1.2.7.1 RPMs and SRPMs for CentOS 4.3 at
http://www.laimbock/asterisk/ And as Axel already mentioned,
there are asterisk RPMs and
Hi Avi
This is great - the problem was how I configured my trunk so this part
of your v. good wiki page was my solution:
-
Maximum channels: num of ports * 2
I have 2 ISDN lines active, so I have 4 maximum channels. If you have
all 4 ports running, you have 8 maximum channels. Each ISDN
Hi all,
i have been having problems with voice quality. We run asterisk Asterisk CVS-HEAD version 2.4 as a production servre as a call centre/customer support engine. Pressenty, we have about 25 soft phones and 10 hard phones (Perfect Tone SIP phones). when handling internal calls, i usually
Doug Lytle wrote:
Nicholas Kathmann wrote:
Both hylafax and * are on the same machine and using the same PSTN
interfaces (whether T1 or TDM). It uses iaxmodem to communicate
between the two systems (imagine a
Hi,
I have this setup in my extensions.conf:
[inbound-analog]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Background(tag-welcome)
exten = 1,1,Voicemail(u100)
exten = 1,2,Hangup 'Zap/1-1'
this means - press 1 and
santosh y wrote:
I'm very new to Asterisk, I'm tracing the Asterisk code,
i'm feeling difficulty in understanding the code, so please tell me
where i can get the documentation of the code and,
design and architecture of the code.
www.asteriskdocs.org is your best bet. Also sign up to
Hello,
snom 360s can handle xml messages via SIP-Notify.
Descriptions how to implement this on:
http://snom.com/minibrowser/doc/xmlapplsnom360.pdf
http://snom.com/minibrowser/notify.txt
Common infos you can find out on: http://snom.com/wiki/index.php/Xmlobjects
Hope this will help...
cheers,
There is a free version of G.729 available? I would be very interested in that!Alex On Mar Abr 18 15:50 , 'Dumpolid Exeplish' sent:Hi all,
i have been having problems with voice quality. We run asterisk Asterisk CVS-HEAD version 2.4 as a production servre as a call centre/customer support
Steve Underwood wrote:
Doug Lytle wrote:
Nicholas Kathmann wrote:
If you need to tweak gains something is seriously wrong.
The 2 fax machines that I was having problem with were failing to train
at 9600bps, they would then try at 7200 and finally train at 4800.
Around 15 pages into the
The switch in the Budgetone is 10Base-T. If the PC NIC cannot auto-detect
or otherwise handle that, it will be a problem.
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: Jerry Jones [mailto:[EMAIL PROTECTED]
Hi,
i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2.
I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib
and oh323) they got to work with Asterisk 1.2.4+.
--
thanks,
yusuf
___
Doug Lytle wrote:
Steve Underwood wrote:
Doug Lytle wrote:
Nicholas Kathmann wrote:
If you need to tweak gains something is seriously wrong.
The 2 fax machines that I was having problem with were failing to
train at 9600bps, they would then try at 7200 and finally train at
4800.
Hi All,
This is a performance update. I have built appliance type servers
with the following specs:
Motherboard Asus P5MT-M
Memory 1Gig DDR2
No hard drive, running in Ramdrive but using Sandisk Compact Flash to
hold compressed image and /var directory
Processor 3.2 Gig Pentium 4, HT Turned Off
Hello,
I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch
versions of OpenH323 and Pwlib (available on
http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works
OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see
2006/4/18, Doug Lytle [EMAIL PROTECTED]:
Nicholas Kathmann wrote: Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM).It uses iaxmodem to communicate
between the two systems
Hi,
when I call the voicemail app, it starts and die suddenly. Has anyone
already had this problem?
Log:
app.c:644 ast_play_and_record: No audio available on SIP/-6fca??
-- User hung up
Tks,
D.K.
___
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Hi there,
i have a Problem with dialtone and a TE401P Card. I swear I surfed
the wiki, the mailing list and google for 4 hours and did not find the
solution, can you help me ?
In Germany I have an E1-Line and an Alcatel 4200 PRO PBX.
Without using asterisk I dial the 0 on an Alcatel Phone and
Hi Herci,
I have tried this. pwlib, openh323 and Asterisk-OH323 0.7.3 compiled with no problems. But when
you start asterisk,
Apr 18 17:47:39 ERROR[11385]: chan_oh323.c:5353 load_module: H.323 listener
creation failed.
Apr 18 17:47:39 WARNING[11385]: loader.c:414 __load_resource:
I have used many sangoma cards, and have not had *any* irq issues
Anton Krall wrote:
Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple
cards? Problems with irq and such (same as with digium ones)?
|-Original Message-
|From: [EMAIL PROTECTED]
Olivier Krief wrote:
2006/4/18, Doug Lytle [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
Nicholas Kathmann wrote:
Both hylafax and * are on the same machine and using the same PSTN
interfaces (whether
Olivier Krief wrote:
2006/4/18, Doug Lytle [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
Nicholas Kathmann wrote:
Please, forgive my ignorance but could you elaborate how your system
would be working ?
I've
HiI'm a new user of Asterisk and made the first VoIP call on my own LAN with a good quality. Now I want to configure a CMTS (motorola BSR 1000) and a server to support QoS. Does anyone knows how to configure this in order to work with SIP and with Asterisk? any ideas or tutorials?
Thanks
Hi,
Anyone have a Call In number offered by Gizmo (
http://www.gizmoproject.com/call-in.php ) and have it configured in Asterisk
sending and reciving calls to that number?.
I have set one peer to Gizmo via SIP number provided by them but when I call
to my Call In number assigned, calls arent
Mac Ethernet ports are auto-switching. Don't need a cross-cable :)
On 4/18/06, Mark Phillips [EMAIL PROTECTED] wrote:
Just for shits and giggles, have you tried using a cross over cable? I'mnot saying it's gonna work because everything I read says you're doing
the right thing but it's worth a
Hi,
I'm trying to find how to configure Asterisk 1.2.7.1 to allow two
EyeBeam (3015c) to send Instant Messages between them... But I cannot
find anything that explains how to do it!
Anybody as a clue? is it possible?
Now, when we try to send an Instant Message in the eyeBeam it says:
Is this with Asterisk in the RTP stream? Is it doing any transcoding?
-Original Message-
From: JR Richardson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18, 2006 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent
Title: IVR: playing multiple streams simultaneously?
I
have worked with several persons on this and there is currently an open request
for sponsors for a whisper function. This is one of the features it will
provide.
Mixing streams to one
channel.
From: [EMAIL PROTECTED]
João Paulo Antunes wrote:
Hi,
I'm trying to find how to configure Asterisk 1.2.7.1 to allow two
EyeBeam (3015c) to send Instant Messages between them... But I cannot
find anything that explains how to do it!
Anybody as a clue? is it possible?
Now, when we try to send an Instant Message
What would cause this? It happened out of the blue:
-- Executing VoiceMail(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/6' (language 'en')
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Tuesday, April 18, 2006 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] correct version of asterisk for oh323
Hi Herci,
I have tried this.
Hi.
I have a problem with two asterisk servers with version 1.2.5. In
one server there is a Digium TE411P in the second the Digium
TE100P.
We use E1 and EuroISDN.
'/etc/zaptel.conf':
- begin -
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
- end -
Anyone experience the double ringing when calling out over TelIAX? I am
using a Cisco 79[46]0, and do not use the r option in the Dial() command.
I always thought that the r is what causes double ring, and is never
really needed except to cause problems...
Sincerely,
Brent A. Torrenga
[EMAIL
Looking for someone with a successful experience similar to
this;
I have a need to cross connect a 3COM NBX PBX PRI interface
to asterisk, but over a long distance. We do not need any IP connectivity and
the solution requires G.711u audio so there is no benefit to using IP.
Has
Hi all,
general query here --- I'm about to set up an asterisk box for use in Japan
but can't figureout if it's all ISDN there or what?
I have gathered so far that the two major providers, NTT and KVH both offer
ISDN lines with ...INS1500 and maybe INS64 protocols?
Not sure...
But I'm
Because of the small screen real estate you might want to use something
like squirrelmail from squirrelmail.org -- this would require some
chopping up to make it fit anyway, but might be easier to implement then imp
moj
[EMAIL PROTECTED] wrote:
Hello,
I read the polycom microbrowser post
On Tue, 18 Apr 2006, Chris Earle (CBL) wrote:
Hi all,
general query here --- I'm about to set up an asterisk box for use in Japan
but can't figureout if it's all ISDN there or what?
I have gathered so far that the two major providers, NTT and KVH both offer
ISDN lines with ...INS1500
I don't think Asterisk supports SIP MESSAGE, does it?
-Original Message-
From: João Paulo Antunes [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18, 2006 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message
Hi,
Hi,
I have several aastra 9133i phones, which are connected to an asterisk
1.2.6 system. I have setup MWI on the phones to point to the IP of
the asterisk server, but although there is a message waiting new in
the mailbox, the phone's light does not light. Any thoughts?
Asterisk was in the RTP and no transcoding, straight Ulaw g.711.
--
JR Richardson
Engineering for the Masses
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Ok
here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using
RedFones FoneBridges. We also have a Nortel Option 11C that we
have hooked up to the Asterisk.
We
have 3 T1s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami
signaling. Then we have a local
In theory it should work, I have just written a proposal for someone
that involves such a setup. in worst case I can always utilize the
circuit as data, and use 2 asterisk boxes one on each end to convert
it back to NI2.
On 4/18/06, Damon Estep [EMAIL PROTECTED] wrote:
Looking for someone
Yep,
I agree,
Just watch out for regulatory issues if you are in the USA, handing a
CUSTOMER a TDM interface vs. a SIP/VoIP interface falls under a much
different regulatory and jurisdictional set of rules...
Have you talked to anyone that has confirmed an implementation like
described works
How's tou're service with Sellvoip, I was not able to intergrate them into my system and they had no phone support. I'm using Gafachi now but prefer the rates Sellvoip provide.
-- Original message -- From: "AR Tarzi" [EMAIL PROTECTED]
SellVoIP appears to follow a US
I believe it is
important to determine if the issue arrives at TDMoE level (RedPhone uses it to
avoid a direct T1 link with the box )
Or at PRI level. I
am not clear what kink of link bridges your redPhone to Asterisk. Is it an
ethernet link or a T1 crossover?
Ing. Oscar Andrés
Hi,
Sample database
++---+---+--+-+-
-+
| id | context | exten | priority | app | appdata
|
++---+---+--+-+-
-+
| 1 | incoming|
there 2 types of inbound metered and unmetered. unmetered is unlimited inbound and metered charges per the minutes.
-- Original message -- From: "Steve Totaro" [EMAIL PROTECTED] Inbound should be free as far as I am concerned unless you have a toll free number.
I will tell you straight up that NFS mounted volumes will cause asterisk to
croak if it needs access to something that's not mounted. The first time
the NFS share disappears for a moment, you're going to be restarting
services and losing time on the asterisk machines that need the mounts. It
I have our Avaya connected to Asterisk using NI D channel
protocol over a standard ESF/B8ZS span. It works
great.
Pretty easy. On Asterisk's side I just had to tell
it:
in zapata.conf:
[channels]switchtype=nationalsignalling=pri_cpegroup=1channel
= 1-23
in zaptel.conf:
loadzone=
We have a crossover from telco to the CSU and a crossover
from the CSU to the RedFone and then a regular Ethernet cable from the RedFone
to the Asterisk.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles
Sent: Tuesday, April 18, 2006 2:01
PM
To:
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