Olle E. Johansson wrote:
Sean Kennedy wrote:
Hi all,
I'm trying to see if I can get the hint priority working with my polycom
500.
So far I have 2 /reg entries with the same sip registration, one is
labeled as private, the other as shared. I have set the hint priority
before anything else
hello all,
i have already configure sip.conf and dialplan.
i done the follow me script.
first problem:
i want to call(with kphone) someone at my extension, i
must dial the extension number.
i can't dial their username.
[EMAIL PROTECTED] (work)
[EMAIL PROTECTED] (call fail)
is it
Hi,
If you use digium card, then maybe you set wrong signaling on fxs...
Best regards,
Stevanus
Tim P wrote:
I have multiple sipura 2100 boxes connected to my * box and for some
reason that i cannot figure out when making a call to one and
answering it and then hanging up results in the
Hi,
The problem is that the Sipura boxes don't do call progress monitoring. I
saw this on a thread about a week ago.
If the call is dropped at the * side, then the sip channel is droppeds and
the sipura will drop the PSTN connection. However if the Sipura has trouble
with the PSTN's start/stop
On Wednesday 01 June 2005 06:45, Jennifer Hales wrote:
Hello Matthew,
You need to put exten = o,1,Hangup underneath your voicemail macro, then
if your dial zero the call will come back to you, however it does read back
an error in your ear. It still works.
... or alternatively, if you add
Hi:
I read many documents and I posted my question several
times here without luck. I hope someone can help now
please. Here is an example to demonstarte my problem:
Suppose you manage the FWD server, how do you define
an IAX client behind nat so he can receive calls from
FWD.
NAT client would
hi,
i m new to asterisk word, pl. help me for the below scenario
i have installed TDM22B card. Module is - wcfxs
i m in India so first of all wat zone is to specified is not defined?
Zaptel.conf is -
fxoks=1-2
fxsks=3-4
# ztcfg parses it cleanly.
Zapata.conf contains-
signalling=fxo_ks
While everything seems to be working for the most part
correctly in my mix-network of Zap and Sip phones, it occurred to me that every
call, regardless of whether or not it was answered, is reporting ANSWERED
in the cdr records on mysql.
I was having problems with strange hang-ups the
Hi,
Thinking about an IVR application and trying to get a handle on the best
way to structure it so that the maximum number of concurrent calls can
be achieved..
If the voice prompts were stored in a GSM format and were being played
out through an IAX trunk that uses GSM compression would
Hello!
I would like to know which hardware I need, to use asterisk with up to 20
analog lines.
Also I woul like to know if there is any card that suport both analog and isdn
lines, and if there is any way to make the analog phones now I'm using work
with asterisk.
Thanks
I think we should be thankful that the authors are relasing the software,
rather then crying out loud when you cannot get it to work. More people
will be willing to help you that way. Be ashamed of yourself!
Best Regards,
==
David Choo
Sales Engineer
Business
Daryl G. Jurbala wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Tuesday, May 31, 2005 5:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Hi,
I'm getting unusable DTMF detection with DISA on incoming ZAP channel
(bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in
normal ISDN incoming line.
How can I check what's going on ? What settings to check ?
Anyone with more experience on such scenarios ?
Thanks in
Hello,
We found out that after upgrading the firmware in our GrandStream
BudgeTone phones, that we were not able to transfer calls anymore. We
use the BT's own tranfering mechanisme. We can dial the phone where the
call should be tranfered to. But after that, the original caller stays
in
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Wednesday, 1 June 2005 6:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Daryl G. Jurbala
I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up.The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running [EMAIL PROTECTED] and have a digium 4port line card. This
Hi,
What version of Asterisk @ Home are you
using?
I had problems like that until I upgraded to version
1.0
The problem has not recurred since.
Regards,T
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Wednesday, 1 June 2005 7:24
I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle
batteries (10 year warranty), and 1000VA inverters.
Those deep cycles batteries look quite appropriate... in which kind of
store do you get them?
The combination makes for perfect power and about 2.5 days run
On Wed, 2005-06-01 at 21:23, [EMAIL PROTECTED] wrote:
I am having problems with * not hanging up an incoming PSTN line, if that
line is not answered before the person calling in hangs up.
The line hangs in various states, it has hung with a busy tone, with no tone
at all.
I am running
Hello,
did anyone already experience MMS? SMS works fine, but I can't find infos on
how to send and receive MMS on a similar way with Asterisk.
Thanks
Daryan
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Im running 0.9 I will try upgrading
thanks
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan
Sent: Wednesday, June 01, 2005 9:33 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Hi,
I cannot seem to establish what is causing my analogue line
to be generating incoming calls, so I would like to do some debugging on my Zap
channel.
Can anyone confirm the syntax?
I have tried;
Debug channel Zap/2
Debug channel Zap/2-1
Debug channel zap/2
Debug channel
Hi Gavin,
I'm testing atxfer and it looks work fine, but i have a small problem:
A call B
B answer, dial atxfer extension and then the new peer (C)
If C does not answer the phone, A and B got hangup and cannot speak again
I set canreinvite to no.
Can u help me ?
Thanks
Giordano
Hi all
i been trying to manually hangup a sip channel which is inactive.
Peer User/ANR Call ID Seq (Tx/Rx) Format
x2.xx.xx.x5 6574260125 6f06bf400e9 00102/2 UNKN (d)
i tried soft hangup callerID and User but asterisk said is not a channel.
and i tried sip show channels User and CallerID as
On Wednesday 01 June 2005 11:01, Giordano Grandis wrote:
Hi Gavin,
I'm testing atxfer and it looks work fine, but i have a small problem:
A call B
B answer, dial atxfer extension and then the new peer (C)
If C does not answer the phone, A and B got hangup and cannot speak again
I set
Title: Message
Hi everyone I am
having trouble with codec negotiation. I have Asterisk running at the office and
a SIP phone at home. In my sip.conf, I have allow ordered as follows: alaw
ulaw g729 and gsm On all my office extensions, I have no allow, or
disallow entries. My Cisco gateway
Please forgive the (almost?) OT post. (and the fact that I need a clue-bat)
We've got a situation at one of our sites where a construction crew is
likely to dig up our conduit which houses some data fiber and one pair
of fiber used to tie a Definity 3gsi at a small office building to the
On Tue, May 31, 2005 at 12:35:32PM +0100, Gentian Bajraktari wrote:
Then try to 'modprobe zaptel' and then 'modprobe ztdummy'
'modprobe ztdummy' should load zaptel as well.
If ytou happen to use debian, add the line 'ztdummy' (without quotes) to
the file /etc/modules to modprobe it at system
I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle
batteries (10 year warranty), and 1000VA inverters.
Those deep cycles batteries look quite appropriate... in which kind of
store do you get them?
In the US just about any store that sells batteries including
Hello,
Firstly sorry for covering old ground - I'm new to this. . . .
I've read that you have to be careful when configuring SIP phone extensions
so that an incoming call can't be connected to the trunk.
Anyone have some info on how this can happen and how to stop it?
Next,
Can anyone tell me
Hello,
I need to run an application that sets a few Asterisk
variables, that will be used by AGI scrpits.
Therefore, I believe that application should be run
somehow from within Asterisk, on startup. The
application needs to be always running, since it may
need to update those variables. Is there
[EMAIL PROTECTED] wrote:
On Tue, May 31, 2005 at 12:35:32PM +0100, Gentian Bajraktari wrote:
Then try to 'modprobe zaptel' and then 'modprobe ztdummy'
'modprobe ztdummy' should load zaptel as well.
I've seen this faul, when only modprobe zaptel first would help.
(Debian sarge)
--
No...maybe i don't explain u well.
After that B call C andC not answer (go in timeout), B hear first the beeperr
and then, together A the busy tone. Now i can't re-take the call :|
Thanks
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin
On Tue, May 31, 2005 at 12:06:55PM +0200, David Hajek wrote:
Hi,
I'm trying to configure Sipura 2000 (behind NAT) which connects to
Asterisk (public IP, no NAT) and having interesting results. When Sipura
is behind Linux/NAT firewall it works great and no special NAT settings
on Sipura
On Wednesday 01 June 2005 12:43, Giordano Grandis wrote:
No...maybe i don't explain u well.
After that B call C andC not answer (go in timeout), B hear first the
beeperr and then, together A the busy tone. Now i can't re-take the call :|
I'm afraid I don't have any more suggestions to offer -
Ok, thanks for all.
Just a thingh: how do u set DTMF on your phones ?
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill
Inviato: mercoledì 1 giugno 2005 13.51
A: asterisk-users@lists.digium.com
Oggetto: Re: [Asterisk-Users] AT-320 +
Hi All,
I am having trouble with MOH. I have downloaded the latest CVS head and
when I try to call from PSTN side and play MOH on a queue then the voice
breaks. However if I play the same file using Playback() application and
listen to it through PSTN side then there is no problem. CVan
On Wednesday 01 June 2005 13:04, Giordano Grandis wrote:
Ok, thanks for all.
Just a thingh: how do u set DTMF on your phones ?
We have them set to RFC2833.
I think I've noticed some cases where the remote party hears the tones, but
it's not an issue that bothers me :)
Cheers,
Gavin.
I have installed Vovida STUN server and point Sipura to use it. But no
luck, I still can't hear the other party. I've ended up with having
Linksys to forward all ports to my Sipura (DMZ host) which works.
What is interesting is that when I'm using Vonage service (Cisco ATA) it
works just
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten =>
Fedora core 3 supports SATA on that model.
listas iPfone wrote:
Hi All,
I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t
make it work because linux cant recognize the Hd (HP 160 mb).
No drivers for Centos ...Red Hat... i´t´s drivig me crazy..
Someone have a tip? if i
HI,
I would like to know how can I check if gateway is registered with gnugk?
Thank you,
Mitja
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Terry H. Gilsenan
Sent: Wednesday, June 01, 2005 5:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] UPS rating for SOHO asterisk box
I have
telnet your-gnugk-ip 7000
use AllRgistrations command or limply ? or ?? for
PrintAllRegistrationsVerbose
Of course you have to configure your gnugk to allow you to use telnet on port
7000 ... but i think
you can use it by default
--- Micko [EMAIL PROTECTED] wrote:
HI,
I would like to know
I have installed Vovida STUN server and point Sipura to use it. But no
luck, I still can't hear the other party. I've ended up with having
Linksys to forward all ports to my Sipura (DMZ host) which works.
What is interesting is that when I'm using Vonage service (Cisco ATA) it
works just
This is what happen when i call a peer that not answer:
-- Executing Dial(SIP/401-4de6, SIP/402|60|Thtr) in new stack
-- Called 402
-- SIP/402-fa23 is ringing
-- SIP/402-fa23 answered SIP/401-4de6
-- Attempting native bridge of SIP/401-4de6 and SIP/402-fa23
-- Started music
You just need to read up on IAX a little. IAX has no trouble with
firewalling.
As long as the client registers to the IAX server, the path will be
defined and connectivity will occur.
It may look like an odd port if you don't have a static port forward in
place but it will work.
If you really
Are you using custom music files? If so, how did you transfer them to
the box?
If you transferred via FTP, you need to be sure you set the tranfer type
to Binary before sending.
Tranferring using ASCII has always hosed mp3 files for me on the * box.
The net result being similar to your
On Wednesday 01 June 2005 14:15, Giordano Grandis wrote:
This is what happen when i call a peer that not answer:
Jun 1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable
to create channel Local/[EMAIL PROTECTED]/n do you have chan_local?
I don't like the look of this part at
As some of you know Ive been trying
to facilitate an involvement with www.tellme.com
speech recognition tools and Asterisk. See www.studio.tellme.com
There have been a number of people who are
already integrating the two and utilizing Tellme as an ASP to deliver speech
recognition to
In conjunction with my last post on Tellme I want to write another
suggestion for an application I had.
I dont know if you guys have come across Google Gas http://www.ahding.com/cheapgas
But basically it is an application that this guy has developed
using the Google API to search an
--- Ronald Wiplinger [EMAIL PROTECTED] wrote:
Nardis Dome wrote:
in your sip.conf:
[general]
videosupport=yes ;
That helped a lot
in your eyeBeam settings- try to enable all the
h.263
codec.
hope it helps..
However, I am still not there.
I have installed
Thanks for your reply.
I wouldn't expect more than half a dozen concurrent calls. Also, we can do the
bridge with proxims if needed (not the model with a telco t1 broken out).
The reason I ask about the media converters is to save the trouble having to
interface an * box to each Definity.
Rich
Thanks for the reply.
I'll get up there today and get more details on the Definity.
Alexander Lopez wrote:
We use Wireless b/w two office in Miami We are using the Proxim stuff
and it is solid. Two Asterisk servers doing Iax b/w them should (will)
work fine. What is the interface into the
Greetings to all!
I have been writing a great new voice messaging application on Asterisk, and
am getting to the point of moving it to my own hosting environment. I have
been in discussions with service providers who can provide me with a TDM
voice T1 line (analog?), but cannot provide a
Hi all,
i am looking for informations about large installation
with Asterisk (~3000 users). Has anybody experience
with such a setup. Any comments, suggestions or
problems would be appreciated.
thx in advance...
__
Do You Yahoo!?
Tired of spam?
In an order to save money, I would like to use a PRI that we
have going to one of our dial-up modem banks (We are an ISP.) During
business hours these channels are idle and during our peak internet times, we
are closed. Sounds too good to be true, but I thought I would throw it
out there.
Hi.
Where I can get chan_misdn that compiles with latest asterisk and
mISDNuser cvs ? Or may be chan_misdn is already present in some asterisk
cvs branch ?
TIA
Rus
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Asterisk-Users mailing list
This may or may not work due to timings slips that you
may experiance with the Digium Cards. Your are correct in assuming this
scenaro.
I did the same (pre-asterisk) with an Adtran
Atlas. It is rock solid and works great. What modem access bank are
you using, there has been some talk about
Adam Vocks wrote:
In an order to save money, I would like to use a PRI that we have
going to one of our dial-up modem banks (We are an ISP.) During
business hours these channels are idle and during our peak internet
times, we are closed. Sounds too good to be true, but I thought I
would
Were still using Lucent PM3s
Adam
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Alexander Lopez
Sent: Wednesday, June 01, 2005
10:24 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Pass-through
This may or may
On Wed, 1 Jun 2005, Dustin Wildes wrote:
Adam Vocks wrote:
In an order to save money, I would like to use a PRI that we have
going to one of our dial-up modem banks (We are an ISP.) During
business hours these channels are idle and during our peak internet
times, we are closed. Sounds
I did it...but with no good results.
Could i see a example of peer in extensions.conf ?
I'm trying everythinghs but i always have differenta results :|
Thanks
giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill
Hello,
Has anyone experienced a segmentation fault in asterisk for using G729
against an AS5300 in SIP ?
I'm having this problem. Also, any other codec I use gives me incompatible
media (can be read in SIP DEBUG messages).
AS5300 configured for multiple codecs, so is Asterisk.
Tried G711u/A
Would something as simple as this work?
[InFromZap1] ;Context
for incoming telco calls
exten = 1234567890, 1, Dial(Zap/g2) ;g2
would be the second digium card connected to our Lucent PM3 with a crossover
cable.
Thanks
Adam
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi all,
After finally making the web interface for AreskiCC work I am now running into new issues.
1 In Asterisk the manager doesnt seem to
connect
2 When I try to create the file
additional_areskicc_sip.conf it says Could not open buddy file
Ok guys, due to someone recently backing out I have a couple more
servers left. These are tested, all freshly installed freebsd, double
boxed and ready to ship. I need to get these shipped out by tommorow
before I got out of town, so I need to k now today if anyone wants
them. Make an offer
Well, my faxes passes through asterisk successfully, however I still have
some problems about fax reception by rxfax.
The softfax answers, and negotiates transmission, however then as some stage
of communiation something is wrong.
But I have nothing more but this log:
Jun 2 00:10:21
On May 31, 2005, at 4:30 PM, Karl J. Vesterling wrote:
Garrett, evidently there is some verbage to that effect on the site.
But just to let you know, no other business that we've done business
with requires anything like that. Not a one.
Also worthy of note is that the purchase was
Im using outlook 2003 on windows xp. [EMAIL PROTECTED] v
0.8
Is anyone else having issues with Astapi?
About 50% of the time after I make a call and then terminate
it I have a memory 0X093 error.
Does anyone know what this is?
Cheers,
Dean
Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes.
I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I
configure dtmfmode=rfc2833 (I've tryied inband and info).
Asterisk seems not to see the tones. Could somebody help me? Thanks
What I need to do? Rename/Copy P003-07-4-00.bin to P0S3-07-4-00.sbn
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
El mar, 31-05-2005 a las 23:15, Andrew Latham escribió:
sbn is a signed bin file
P0S-xx-x-xx.sbn would be the format for the SIP image after version 5
List doesnt seem to be posting out still active
here http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html
but not being received by email (time warner is the isp but other emails coming
in every few minutes as per normal).
Cheers,
Dean
I have installed xorcom and [EMAIL PROTECTED] on 2 different pc's,
with astcc.
It only registers the once of connection billing, and never again.
I have tried everything. Am I doing something wrong?
I will appreciate any help!
--
No virus found in this outgoing message.
Checked by AVG
OS79XX.TXT should contain:
P003-07-4-00
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday,
Check this out:
http://www.engagecom.com/Products/iptube_T1.htm
On 6/1/05, Rich Adamson [EMAIL PROTECTED] wrote:
Please forgive the (almost?) OT post. (and the fact that I need a clue-bat)
We've got a situation at one of our sites where a construction crew is
likely to dig up our
On May 31, 2005, at 8:05 PM, Andy Hamilton wrote:
On 5/31/05, Robert Goodyear [EMAIL PROTECTED] wrote:
Does anyone know how to suppress the Missed Calls indication --
perhaps on a per-line basis -- on the 7960 running SIP?
Reason: I've configured a group of extensions to ring for inbound
That should work but you need to have the asterisk box
setup to do pri-net on the connection to the PM3. I would add the did dialed so
that the PM3 knows about it for radius accounting..
exten = 1234567890,
1, Dial(Zap/g2/${EXTEN})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hello,
I'm getting unusable DTMF detection with DISA on incoming ZAP channel
(bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in
normal ISDN incoming line.
I'm having similar problems with a gsm gateway connected to x100p.
The DTMF for 1, 4 and 7 are detected fine, but 2, 5
No problems here. 27 min behind according to your post time.
Dean Collins wrote:
List doesnt seem to be posting out still active here
http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html but
not being received by email (time warner is the isp but other emails
coming in every
On Jun 1, 2005, at 9:38 AM, Ing CIP Alejandro Celi Mariátegui wrote:
What I need to do? Rename/Copy P003-07-4-00.bin to P0S3-07-4-00.sbn
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
No, renaming won't work, as it's a signed binary. Plus S versus O
designates the
Hi,
I have an asterisk running with a passtrought conf with G729,
when I try to send a fax from SIP to SIP the ATAs make a good codec
negociation and the fax transmicion is OK,
But when I try to send the fax to PSTN fax machine
(SIP -- AS5400 -- PSTN)
The ATA Device try to send the RTP with
Try manually creating the file first.
--- [EMAIL PROTECTED] wrote:
Hi all,
After finally making the web interface for AreskiCC work I am now running
into new issues.
1 - In Asterisk the manager doesn't seem to connect
2 - When I try to create the file
Hi
Need help on bridging SIP with TDM400P(4 FXO Modules )
My setup is as follows
US OFFICE -TDM400P(FXO) --SIP--- TDM400P(FXOs)INDIA OFFICE
(DSL Line) Asterisk
Asterisk PBX(Siemens) /DSL Line
Give it a break you freakin
Cry Baby
Race the Tyrant Vanderdecken
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling
Sent: Tuesday, May 31, 2005 8:05
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE:
Hi List
I bought 1 Dell SC1425 server and 1 Digium TE110P T1/E1 card.
I installed Asterisk from aah 1.0
In the CLI I type 'genzaptelconf -svd' as I have done with other servers and
FXO cards to detect and configure the cards; this time it is not recognizing
the T1 card.
Any ideas why this might
Btw just in case someone is looking, maybe we can get
someone on the inside to help out J
http://www.tellme.com/job_voice-xml.html
Service Production
Engineering: Senior Engineer, Applications and Tools
Tellme leads the industry in large-scale
Hmmm,
You are going to price yourself out of the market if you go with
hot swap.
If I understand you correctly that is.
Your residential gateway sits in a home and connects to the
internet to do VoIP calls for the owner.
What is your cost for this gateway? Doing
It is likely possible. It's going to depend on getting * and your
modem bank to play nice together. If your modem bank is collecting ANI
or any kind of other carrier signaling info for normal operation, you
might have an easier time doing EM wink between * and the modem bank
if your modem bank
Hi,
I made a full strace of the running Asterisk process during a high load 99% of
cpu usage, aprox. ~800 MBytes of data was gathered and found
lots of errors in this log.
The errors started when * tried to open a /dev/zap/channel file (before this,
there were other errors but I think there
I have 3 Asterisk systems that connected through IAX2 trunks. System 1 has a
TE110P installed with a PRI and routes calls based on calling number to systems
2 and 3 through the IAX2 trunk.
Systems 2 and 3 have TDM400P cards installed for failover and emergency/911.
I am having problems
Thank you very much for all answers.
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I'm experimenting with asterisk. This is my environment:
- Debian sarge (vanilla kernel 2.4.29)
- Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g
- Two sip phones (One cisco 7905 and one soft-phones X-Lite)
- Digi International Datafire Micro V (Europe) (rev 02) (zaphfc)
After two days of work now I can
Hi,
we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that
are restarting after 5 - 15 minutes, before and after that we can make calls in
and out w/o problems. The cards have been tested in two computers (Atholon XP
2200+ and Celeron 2.6Ghz), are on there own IRQ, not
I was pondering of the best way to implement voice-coloring within
Asterisk, e.g. pass a channel thru a multiband equalizer and modify it
enough where it could be distinguished from other voices in a
conference call. This could make conference calls much less confusing.
Perhaps the easiest way
I've gotten my CDR working the way I like, but I am looking to customize it a
bit. I have set up an IVR menu, which works great. I would like to be able to
capture the prompted DTMF digits pressed by callers, to my CDR database but I
don't see any AGI or Asterisk commands that allow one to
El mié, 01-06-2005 a las 12:34, Robert Goodyear escribió:
No, renaming won't work, as it's a signed binary. Plus S versus O
designates the application type.
Yes, that's correct, S isn't the same to O
My firmware version is 6.3. I check info on these files:
cat OS79XX.TXT
POS3-07-4-00
and
Dear all,
Sorry to ask, but...
Do you know where I can find a full list of configuration parameters and
values for each of the .conf files?
Do default .conf files include all options?
Thanks Again
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Asterisk-Users mailing list
Does anyone know the pinout to make a cable so that My Asterisk can talk to
my Mitel 200SX?
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 24, 2005 1:47 PM
Subject:
I don't see the SugarCRM being part of the install.
How do you activate this?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 31, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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