It doesn't have to be IAX. Do you know how to
configure it with another protocol?
have a look at
http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html
=Stefan
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On 6/8/05, VoIP Newbie [EMAIL PROTECTED] wrote:
My 4-port FXO is only $300.
Which product/model are you using then?
/wai-sun
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
-Original Message-
1- Anybody implement mgcp useragent in *.
Nope. Hasn't been done yet.
2- Where can i get that.
Not available in your nearest drugstore.
3- if no then anybody can help me to write it down.
Digium ?
Florian
___
Hi,
-Original Message-
I have set up [EMAIL PROTECTED] with Digium TDM400P 2FXO/2FXS.
I am unable to seize my trunks from either soft or analog phones.
Inbound calls result in answer/disconnection.
I see the following error code on my asterisk server
INIT: Id s0 respawning
On Wed, 8 Jun 2005, David Phelan wrote:
If you download the configuration tool which I couldn't get working on my
systemthere is a cfg template in there for 1.0.1.8
Oh, then they have added it, or we missed it the first time around. We
have it running. We had to tweak the paths in the
Hi
I'm new in this field, have been reading a lot, and have a little question.
could it be possible to connect an Avaya IP office pbx to asterisk using a
E1/T1/Pri?
Original instalation:
Telefone company|Pri---Pri|IP Pffice
My Question:
Telefone company|Pri ---TDM|Asterisk|TDM ---Pri|IP
Hi all,
iam a student trying to build an asterisk pbx as a
simple configuration only two extention (using
Xlite)without outsite telephone line. i already follow
the instruction and seem the asterisk work fine
because there is no error message. when i configure
SIP.conf and extention.conf i hope
Guys (and Gals),
FYI I also have the *same* message here. Wonder is it is related to my
Compaq D500 Space Saver PIV 1.7 or the fact that I don't yet have a
modem card in the * box.
(Please don't shoot me, did try Google first)
Many thanks,
Wagner Gimenes
-Original Message-
From:
Hi,
Is it possible to bypass incoming ring on asterisk so that incoming
calls come to asterisk box will be directed straight into did?
Is anyone able to give me any clues or pinpoint me where I can get more
information about it?
Thanks for your attention..
Best regards,
Stevanus
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
when i try to dial a number it just dies.
Meaning what? Silence? Hangup?
Does dialing voicemail on that same setup work? That would tell
whether it hears the DTMF.
Other wise, check the codec and dtmf mode, some combinations don't
work on some phones.
Enabled: yes
Display Name:
Username:
Authorization User:
Password:
Domain/Realm: mysip.server.com
Is this your username:
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
You can first answer to call, and then provide playtones(ring) to caller.2005/6/8, stevanus [EMAIL PROTECTED]:
Hi,Is it possible to bypass incoming ring on asterisk so that incomingcalls come to asterisk box will be directed straight into did?
Is anyone able to give me any clues or pinpoint me
You can connect it through H.323
Thanks Regards
Ritesh Jalan
Senior Engineer - Test Audit
Net4India Ltd.
703 Bikaji Cama Bhawan
11 Bikaji Cama Place
New Delhi - 110029
Ph: +91-11-26160129 ext. 131
Cell : +91-9818616329
Web site: http://www.net4india.com
HI..!!
Is you windows PC the Asterisk in the same LAN.
-Original Message-
From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 08, 2005 2:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Xlite not communicating with Asterisk
Dear All,
I
A prefix will be passed for authentication from Asterisk to cisco AS5300
Thanks Regards
Ritesh Jalan
Senior Engineer - Test Audit
Net4India Ltd.
703 Bikaji Cama Bhawan
11 Bikaji Cama Place
New Delhi - 110029
Ph: +91-11-26160129 ext. 131
Cell : +91-9818616329
Web site: http://www.net4india.com
You can connect it only via SIP2005/6/8, chawki hammoud [EMAIL PROTECTED]:
Hi:I have been Googling around for documents of how toconfigure aCisco AS5300 to connect to the PSTNthrough Asterisk, IAX channel.Please help me configuring Cisco and IAX or send mesome documentation referral.
You must answer the call anyway. And then playback some message2005/6/8, Sahil Gupta [EMAIL PROTECTED]:
Hi,I'd like to know how I can playback a pre-recorded message to a user usingour system without answering the call.I want to do the above in the scenario where the user dials a number andthe
Please visit www.broad-tel.com for details.
On 6/8/05, Wai-Sun Chia [EMAIL PROTECTED] wrote:
On 6/8/05, VoIP Newbie [EMAIL PROTECTED] wrote:
My 4-port FXO is only $300.
Which product/model are you using then?
/wai-sun
___
Asterisk-Users
http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html
http://www.asteriskguru.com/tutorials/xlite_softphone.html
Read these two tutorials and you should be fine.
Zoa
Wilson Pickett wrote:
Enabled: yes
Display Name:
Username:
Authorization User:
Password:
Domain/Realm:
Hi
I have a PHP agi-bin scripted called callhander.php and its setup to
answer anything that comes into the PBX,
In the script I am trying to the get the system to play a file called home
which I know works, as I can get the Play function to work from the
extensions.conf file. However within
Hi Shahan,
yes both are in the same LAN
Thx
MAG
Shahan Kalutanthri wrote:
HI..!!
Is you windows PC the Asterisk in the same LAN.
-Original Message-
From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, June 08, 2005 2:29 PM
To: asterisk-users@lists.digium.com
Subject:
Hi Wilson,
yes I am leaving it blank although I did try to use a username in the
sip.conf but with the same result also I have tried to put the extension
881 but the same result.
Wilson Pickett wrote:
> Enabled: yes
> Display Name:
> Username:
> Authorization User:
> Password:
>
Hi all,We have a little problem.One of our customers has a problem with DTMF pass-thru.They use GrandStream 286 devices to connect their pstn phones to asterisk.everything works like a charm, except DTMF pass-thru. when they call an IVR system, they cannot select options because the DTMF tones
Title: Message
on the
asteriskconsole puta "sip debug" and see if you get any debug
information.
coz
even though you extension.conf or sip.conf is not properly configured still you
should get the debug info..!!
shahan
-Original Message-From: Mohamed A.
Gombolaty
I have already looked into this page. I thought this
was for AS 5350, I am not familiar with Cisco products
and I don't know if there is a difference. And there
is no Asterisk set-up in this example.
Regards;
--- Stefan Reuter [EMAIL PROTECTED] wrote:
It doesn't have to be IAX. Do you know
keep reading
- Original Message -
From:
infra
struct
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 07, 2005 10:02
PM
Subject: [Asterisk-Users] so what are the
additional hardware componentsneeded?
I have 20 personal computers in LAN with full
Hi,
Is here someone who could provide meany information from
practical using of * ?
I need to know more about performance. The
main question is:
"How many extensions should i have configuredin and
provided with my * box in several cases":
1. * is usedonly for SIP signalling, no rtp
You will need 1 tdm card in the server, with 1 or more fxo ports on it.
Thats all you will need. All pc will dial out through this 1 server.
Zoa.
Steve Totaro wrote:
keep reading
- Original Message -
*From:* infra struct mailto:[EMAIL PROTECTED]
*To:*
Hi Roywish,
The best way is to publish here your .conf files to correct.
Good luck...
Best Regards,
Francois BERGERET,
Happy * french user :-)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de craz sead
Envoyé : mercredi 8 juin 2005 09:45
À :
Yes, I hooked one up yesterday. Although we have an Asterisk server in
house, I wanted to connected directly to a host in the US for Faxing. There
was no issue with NAT, and I did not do anything special beyond the usual.
[111]
callerid=test 111
type=friend
username=111
password=mine
host=dynamic
Joshua Colp wrote:
You're actually confusing me when you say this due to the fact you're not
giving much information, probably why nobody has responded yet. If the SIP
server on the Nortel does an INVITE for the phone number, then asterisk will
act accordingly and go to the phone number in the
Hi,
I have connected 4 analog public telephone lines to an Asterisk server using a
Digium TDM400P card and that working fine. But my 4 lines are connected to
each other in a group by the telecom operator. So if someone calls me all 4
lines are ringing. I wrote a AGI script which will handle
Nick Barnes ha scritto:
I've only ever seen when the signalling is wrong. For example if the line is
in PTMP mode when it should be in PTP or vice-versa.
this is the zapata.conf:
group = 1
context=default
signalling = bri_net_ptmp
channel = 1-2
So, you're using NT mode PTMP signalling.
When I am receiving faxes, which will go through a Sipura 2002, the server
says
rtp.c:504 ast_rtp_read: Unknown RTP codec 100 received
I still get the fax, any idea what this is?
Chris Mason
___
Asterisk-Users mailing list
I also have someone in New Zealand who has done some for our own
Asterisk server.
Mark Phillips wrote:
I've found a woman whom is happy to help make English voice files!
Ironic that she should be in New Zealand.
More when I have the files.
___
Joshua Colp wrote:
Okay lemme give you something that should work some magic!
Stuff for sip.conf:
[nortel]
type=peer
host=IP ADDRESS OF NORTEL
disallow=all
allow=ulaw
context=inbound_nortel
insecure=very
Stuff for extensions.conf:
[inbound_nortel]
exten =
On Tuesday 07 June 2005 09:44, Giordano Grandis wrote:
Ok, just a thing...cuold is see a sample peer in tuou extensions.conf
I'm newly testing the atxfer and i always the same question: if i
transfer a call to a peer that don't answer me, ho can i re-take the
call. Actually i got
I'm new in this field, have been reading a lot, and have a little question.
could it be possible to connect an Avaya IP office pbx to asterisk using a
E1/T1/Pri?
Original instalation:
Telefone company|Pri---Pri|IP Pffice
My Question:
Telefone company|Pri ---TDM|Asterisk|TDM
I am not sure if this is really possible but I figured I would ask
anyway. I have a customer who wants an asterisk system. Currently they
have a BizFon system.
The feature that he really wants is to be able to pick up any line and
have all the stations show up on his phone. Is this possible in
Followup to myself:
I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying
to make it work with Asterisk. I tried versions 1.0.7 and yesterday's
CVS and the behavior is the same.
The phone registers with no problem, and can accept calls.
But when I try to make outgoing
Greetings,
I have my first asterisk installation up and running, thanks to a lot
of reading. Could anyone point me in the direction of things to read
on automated outbound dialing? NOT predictive dialing - I will not
have agents handling the calls. These calls are reminders for
appointments,
Isn't it easier to talk to your Telco, and tell them to just ring the
first free line, instead of all 4?
Julian J. M.
On 6/8/05, Erwin Lubbers [EMAIL PROTECTED] wrote:
Hi,
I have connected 4 analog public telephone lines to an Asterisk server using a
Digium TDM400P card and that working
Stefan,
Is it possible to have the Cisco forward calls between T1 or E1 interefaces,
without VOIP DSPs, but only Modem DSPs ?
I need to have an AS5350 that is currently configured as a dial-in RAS to
forward incoming calls to Asterisk, but I can't do it with SIP, as I don't
have VOIP DSPs on
On Wed, Jun 08, 2005 at 08:38:27AM -0400, Sean Cook said:
The feature that he really wants is to be able to pick up any line and
have all the stations show up on his phone. Is this possible in
asterisk? If so can someone point me in the right direction?
That describes a key system. Asterisk
Rich Adamson wrote:
Any chance that we could get someone to implement the milliwatt
generator and echo test number. Would be kind of handy for testing
various items (eg, jitterbuffer).
It's running CVS HEAD (which means it has the new jb since we didn't
disable it, but then again it's all
We have it:
http://www.thevoipconnection.com/store/catalog/product_16198_VoIP_Telephony_
with_Asterisk_by_Paul_Mahler.html
Michael Crown
Managing Partner
The VoIP Connection
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: John H [mailto:[EMAIL PROTECTED]
Sent:
On Wednesday 08 June 2005 08:38, Sean Cook wrote:
I am not sure if this is really possible but I figured I would ask
anyway. I have a customer who wants an asterisk system. Currently they
have a BizFon system.
The feature that he really wants is to be able to pick up any line and
have all
Hi
In the end we found it easy to record our own using this section in
extensions.conf. This also meant that we could add our own company specific
ones in the same voice (not shown here). Basically you get someone to dial the
8NNN1 to record or 8NNN2 to playback. The prompts are shown below
Matt wrote:
Isn't the SS7 code for Asterisk available under the commercial
Asterisk license and that's the only way to get it?
No, that's a poor description of the availability... one of these days
I'll have to ask them to stop wording it in quite that way.
If you want to use the
I think you miss the point Andrew. She's not from NZ but from England.
She speaks English. Says six and not sex etc.
Mark
Andrew Thrift wrote:
I also have someone in New Zealand who has done some for our own
Asterisk server.
Mark Phillips wrote:
I've found a woman whom is happy to help
I had this same issue - it's because AAH tries to run a getty on ttyS0,
and if you have COM1 disabled in the bios (or it doesn't exist), this
won't work. If you're getting this issue, edit /etc/inittab, and
comment out the line that says:
s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an error
on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
Fixed, you can find the patch
Like to share who can record NZ / Australian voices?
Regards,
Sahil Gupta
VoiceValley
On Wed, 8 Jun 2005, Mark Phillips wrote:
I think you miss the point Andrew. She's not from NZ but from England. She
speaks English. Says six and not sex etc.
Mark
Andrew Thrift wrote:
I also have
Hello
I'm newbie in asterisk and i have a AVM Audiovisuelles MKTG Computer System
GmbH Fritz!PCI v2.0 ISDN (rev 02) with CAPI Driver.
I would like install fax detection, but i don't know if i should use
NVBackground detect; or CapiAnswerFAx; or other.
I don't understantd operation of fax.
Tx
I suggest you wait a little for the new o'reilly book about asterisk.
Amazon already accepts pre-orders for it
The VoIP Connection wrote:
We have it:
http://www.thevoipconnection.com/store/catalog/product_16198_VoIP_Telephony_
with_Asterisk_by_Paul_Mahler.html
Michael Crown
Managing Partner
If you activate (via sip.cfg) the feature Group Call Pickup, its no
surprise that asterisk doesn't know what to do with this feature
request. But it is being sent as a SIP SUBSCRIBE request, and I'm
wondering if, as asterisk stands, there is a way to take advantage of
this feature to emulate the
When calling from sip phone to sip phone ( cisco 7940 ) we have very little or
no echo. When conferencing through meetme through a sip only server, we
experience lots of echo.
Would this have anything to do with the timing
source?
The server is using ztdummy on 2.4 with uhci usb. Would
All,
I have an [EMAIL PROTECTED] installation with a TDM40B card. I can make
internal
IP calls with no problems, but when I try to dial out I get a message that All
Circuits are Busy. I looked into the Zapata.conf files and such but see no
modifications. Is there a step that I am
Julian,
Thanks, but it isn't an option because the Telco is actually connected to
a PBX which is connected to Asterisk which should act as a intelligent
answering device during non-office hours. The PBX isn't capable of doing
this. Any other option?
Regards,
Erwin
Isn't it easier to talk to
unplug the other three lines
This is an after hours ring group or is this enabled after hours only?
On 6/8/05, Erwin Lubbers [EMAIL PROTECTED] wrote:
Julian,
Thanks, but it isn't an option because the Telco is actually connected to
a PBX which is connected to Asterisk which should act
I've just had polarity reversal provisioned by our telco to test hangup
detect with a TDM400P
I've set hanguponpolarityswitch=yes in zapata.conf
When I start Asterisk I get ignoring hanguponpolarityswitch
in /var/log/asterisk/messages
I assume that the option is either not valid or conflicts
Hi,
-Original Message-
Thanks, but it isn't an option because the Telco is actually
connected to
a PBX which is connected to Asterisk which should act as a intelligent
answering device during non-office hours. The PBX isn't
capable of doing
this. Any other option?
Hmm, this is
With the current CVS-HEAD line 88 of app_rxfax.c causes an error.
#if (ASTERISK_VERSION_NUM = 010300)
chan-callerid,
app_rxfax.c:88: error: 'struct ast_channel' has no member named
'callerid'
Commenting out the if else combination of course gives a clean compile.
--
[EMAIL PROTECTED] wrote:
All,
I have an [EMAIL PROTECTED] installation with a TDM40B card. I can make internal
IP calls with no problems, but when I try to dial out I get a message that All
Circuits are Busy. I looked into the Zapata.conf files and such but see no
modifications. Is there
--- Alexander Ilyushin [EMAIL PROTECTED] wrote:
You can connect it only via SIP
If you know how to configure the cisco AS5300 and SIP,
I appreciate it if you write the configuration down.
Thanks;
__
Yahoo! Mail Mobile
Take Yahoo! Mail
read in voip-info.org about Asterisk Call Manager API, and may be an
easier soultion are the .call files that you can pleace in
/var/spool/asterisk/outgoing/ these files have a description of the
type of call you wanna make, in the very moment that you place the
file there, a call will be
Hi, I have a problem I will describe. I have PAP2 connected to the internet
to an asterisk box with 2 TDM cards, one TE100P E1 with PRI and one TDM400P
with 2 FXS an one FXO.
When I call to the TDM400 cards from the PAP2 eveything is OK, sound quality
is perfect.
When I call to terminate the
This feature is called attendant - night answer position. Is it not
possible to switch the incoming call to an alternate extension based on
time of day ?
Henry
Florian Overkamp wrote:
Hi,
-Original Message-
Thanks, but it isn't an option because the Telco is actually
make sure that the DTMF mode configuration in Asterisk match the
configuration inside the Grandstream devices. I mean, in asterisk
config you may need something like
[20]
type=friend
.blah
dtmfmode=info
and of inside the configuration of the Grandstream device you may have
to use the same
On Wednesday 08 June 2005 10:57, Neil and Fiona wrote:
I've set hanguponpolarityswitch=yes in zapata.conf
Do you also have the signaling on the channel set to kewlstart? I don't
believe polarity detection does anything without this signaling type.
When I start Asterisk I get ignoring
rxfax doesnt work with voip, you need something like NVFaxDetect from
Newman Telecom to detect the incoming fax.
Essentially you sent him an email and he'll send you the code. Once you
compile them into asterisk you can add it.
http://www.voip-info.org/tiki-index.php?page=NVFaxDetect
JD
Dean,
Actually, I have run genzaptelconf -s -d but it still didnt seem to modify any
of the config files that I look at in the AMP console. Should I try modifying
the config files manually?
Thanks,
Marc
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
On Wednesday 08 June 2005 11:19, Alejandro G wrote:
When I call to the TDM400 cards from the PAP2 eveything is OK, sound
quality is perfect.
When I call to terminate the call in PSTN through E100P I hear clicks which
aparently are RTP packet looses. This clicks are only heard in the PSTN
On Wednesday 08 June 2005 11:24, Henry Coleman wrote:
This feature is called attendant - night answer position. Is it not
possible to switch the incoming call to an alternate extension based on
time of day ?
You need to read up. This exact situation is given in the Asterisk Handbook.
Yeh, this is called line hunting all telco's offer this... you get
one published number but say 12 lines each line actually has a
number but just calling the main number will automatically roll-over to
the first available line in that hunting group. By the way, outgoing
calls that use
On Wed, 2005-06-08 at 11:34 -0400, Andrew Kohlsmith wrote:
On Wednesday 08 June 2005 10:57, Neil and Fiona wrote:
I've set hanguponpolarityswitch=yes in zapata.conf
Do you also have the signaling on the channel set to kewlstart? I don't
believe polarity detection does anything without
Everyone using CVS head, and owning flex-2.5.31 (or higher)--
Please note that a new version of the expression ( $[ ] constructs used in
extensions.conf ) parser
is automatically built by the makefile if your flex is at 2.5.31 or higher. You
can see what your
flex version is by saying flex
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have
I'm trying to setup
remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc
Anyone have example
of what I need to change to make an asterisk server log on a remote mysql
server?
___
Asterisk-Users
Will do ..Thanks Henry
Andrew Kohlsmith wrote:
On Wednesday 08 June 2005 11:24, Henry Coleman wrote:
This feature is called attendant - night answer position. Is it not
possible to switch the incoming call to an alternate extension based on
time of day ?
You need to read up. This
Tim wrote:
I'm trying to setup remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc
Anyone have example of what I need to change to make an asterisk server
log on a remote mysql server?
If you are going to store CDRs on MySQL, why not skip
On Wednesday 08 June 2005 12:00, Neil and Fiona wrote:
/var/log/messages seems to be indicating that the wctdm driver thinks
that the polarity of the line is reversed on start. (ie incorrect
polarity)
Polarity reversed (0 - 1)
Reverse the tip and ring on the line then. :-)
I'll check it
Thanks Johann. - that helps out .
Johann wrote:
Hugo,
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
(716)250-3405
1st column is unixtime stamp for the current date
2nd column is not really sure...maybe the duration?
3rd column is the queue name
4th column is their agent
I've used that feature in asterisk HEAD, and it has worked for me (i
needed to apply a little patch for it to work for incoming calls
also), but i also used answeronpolarityswitch=yes. Maybe it's a logic
bug in the code. Try with that option and tell us the results ;)
BTW, it doesn't matter is
On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm
CG-410 4 FXOs device. Now I just ordered a few Sipura 3000.
With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the
following :
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes (I tried 800 with TDM04B
The configuration in the blog does not depend on the product, it
depend on the IOS used. Should work for your 5300, the only problem you
could have, AFAIR is with the SIP-ua config. Authentication, starts after
12.2.something.
If you have problem come back and I give u a
Yes you can. There are some examples @ cisco look for TDM switching.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcelo
Pacheco
Sent: Wednesday, June 08, 2005 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
is one of the things also affected. Now I'm using TE110 card in my
system. I hope
Is there any metric on the number of AGI's that can run
at the
same time. Shouldnt be a limit in my mind but I am thinking in
terms of system performance.
My AGI is a C program with 3 meg executable size.
Thanks,
Jerry
___
Asterisk-Users
Hello list.
I'm going te explain my trouble.
I have my asterisk with a TDM400P with 4 FXS channels. Two ports are
connected to a Panasonic PBX (it's working fine), and others two ports
are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected
to a CO port (where i had a pstn line).
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0
Unless you are making measurements and actually analyzing the results you're
only stabbing in the dark playing with these things.
by the way I live in Canada and the
I have asterisk 1.0.7 and I made the required patch and got everything
installed. I have libtiff 3.7.0, and I'm using the zaptel stuff.
When I send a fax to it, it autodetects the fax and starts rxfax, however, the
fax machine just sits at 1% and then disconnects. I don't have any error
On 6/7/05, Johann [EMAIL PROTECTED] wrote:
Hugo,
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
(716)250-3405
2nd column is not really sure...maybe the duration?
Asterisk UniqueID of the call.
-Brian
___
Asterisk-Users mailing
did genzaptelconf -s -d say it found any cards?
--- [EMAIL PROTECTED] wrote:
Dean,
Actually, I have run genzaptelconf -s -d but it
still didnt seem to modify any
of the config files that I look at in the AMP
console. Should I try modifying
the config files manually?
Thanks,
Marc
I use Digium TDM400 cards as well. Asterisk's software echo cancellation
sucks. From what I've heard on the IRC channel, you'll never completely
eliminate echo with it. And unfortunately, hardware echo cancellation starts
out at a full T1. They don't seem to have any solution for someone with 4
I have seen the same problem. The zaptel hardware looks fine in zttool and
appears to be ok when genzaptel -s -d is run, but when you look at the zap
channels in CLI, you only see the pseudo channel.
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Jorge,
As far as I've read, you won't be able to handle 8 E1 in one box.
By the way, have you had success with interconnecting E1 R2 argentina? I´m
having trouble with a Meridian... I can only make calls from asterisk, but
the other way arround...
Tks
Franco
- Original Message -
From:
Dean,
Here are the results of the genzaptelconf -s -d. As you can see, it is
throwing some errors, but I am a bit of a newbie so any help you could provide
would be greatly appreciated!
[EMAIL PROTECTED] /]# genzaptelconf -s -d
STOPPING ASTERISK
Asterisk ended with exit status 0
Asterisk
1 - 100 of 158 matches
Mail list logo
