Adam,
Thanks for your help.
Does anyone know or is anyone an * guru in the New Hampshire/Vermont area?
how about this example.
User1 sits at his desk, a call comes in.(doesn’t matter how the call
gets to his phone, DID or exten) he needs to go into the warehouse to
look at something. He
Dear All,
I installed an Asterisk on a linux PC, and X-Lite on two Windows
PCs, all in a LAN.
But, when I make phone call from one X-Lite to another, I always get
Call Failed: 404 not found.
Here is my sip.conf:
[Phone1]
Hi all,
I've just updated to current CVS, and have 2 polycom IP phones, one is a
IP600 and the other is a IP300. The IP600 shows the status of the IP300
and a ZAP line quite nicely, but the IP300 won't show the status of the
IP600
Is there any additional debug apart from show hints to see
Hello everyone,
I'm writing a macro to use the telephone keyboard as a means for users
to type in text. For some weird reason, I'm having problems with
Asterisk command Read ... If I dial 0, the asterisk debugger prints
User entered nothing. If I dial OO, asterisk recognizes both
digits...
Hi,
I am Newton from Liguetel in Brazil.
I have now a billing system based on SQLPostgress which is able to collect
real time CDRs and present in a web site all the accounts and CDRs related
to their calls.
This billing is also able to set accounts balance and for each call
balance goes down as
On Sun, September 4, 2005 9:53 pm, Stijn Jonker said:
I have a BRI 2-channel Eicon Diva on KPN and yes, look at the sig, I'm
dutch.
Which part of your conf should I use and not use?
That depends on the driver, do you use chan_capi or junghanns bristuff
for the eicon?
zapfhc driver seems
On Sun, September 4, 2005 10:07 pm, Armin Schindler said:
That depends on the driver, do you use chan_capi or junghanns bristuff
for the eicon?
bristuff for Eicon Diva card? That's not possible.
Which card do you use exactly? Is it a DIVA PCI or DIVA Server card?
In case of DIVA PCI you can
This billing is also able to set accounts balance and for each call. Now I
need to disable accounts which balance gets a determined value. I was
thinking on changing account pass for that specif account which we need to
disable. And then in the sip.com reload info.
Can you help me with new
Von: Pascal Speck
[mailto:[EMAIL PROTECTED]]
Gesendet: Montag, 5. September
2005 10:37
An:
'[EMAIL PROTECTED]'
Betreff: Timeout when Dialing -
HELP
When i try to do a call I get this message after a few
seconds:
I IND :TIMEOUT pid:1 mode:NT
addr:51400102 port:2
--
Hi
Although canreinvite option is yes,
the asterix doesn't send reinvite and the media is going through the asterix
instead of between the two sip phones.
Both sip phones (handytone 486)
don't use NAT and are configure with canreinvite option yes and use the same
codec G.729. And
In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
Anybody having issues with ztdummy under the current 2.6 RC7? I get the
following errors when trying to modprobe ztdummy:
Hi All,
I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with
RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt
and asks caller to dial four-digit extension. Caller has to dial slowly,
otherwise, Asterisk cannot recognize the extension number. I
I can't seem to get any ZAP trunks on my TE110P to match any extensions
for incoming DID.
I've even used the exten = _X.,1And it still will not match that.
All I get is:
-- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at zap-custom,s,1 failed so falling back to exten
's'
==
hya,
try using relaxdtmf=yes in zapata.conf and see if that solves it.
checkout these recent postings as well:
http://lists.digium.com/pipermail/asterisk-users/2005-August/122737.html
http://lists.digium.com/pipermail/asterisk-users/2005-August/122656.html
cheersOn 9/5/05, larry lin [EMAIL
Hello,
I have two polycom ip300.
I patched Asterisk However it don't show status of
phones when I press busy, Away, ...
So I use Sip Express Router (proxy sip) for IM and
Presence SIMPLE.
Harry
--- Adam Goryachev
[EMAIL PROTECTED] a écrit :
Hi all,
I've just updated to current CVS, and
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
hi everyone. can anyone provide me concrete examples on how to use the GotoIf application? can't figure out how to use it in my dialplan coz im having errorsthanks! : )__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection
Hello list,
while using the new Asterisk 1.2 beta, I keep noticing this when an agent
transfers a call from a queue to another extension:
[Except from queue_log]
1125912636|1125912630.134|queue-dps|NONE|ENTERQUEUE||21
1125912638|1125912630.134|queue-dps|Agent/101|CONNECT|2
Here you go, eagerly awaiting comments:
-- Executing SetCallerID(SIP/xlite1-e0a7, 0) in new stack
-- Executing Dial(SIP/xlite1-e0a7, CAPI/hfcpci/17/b) in new stack
data = hfcpci/17/b
parsed dialstring: 'hfcpci' '17' 'b'
capi request for interface 'hfcpci'
What about faxing yourself if you don't have a scanner?
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Johan van Tongeren
Sendt: 5. september 2005 09:11
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [Asterisk-Users] sending fax
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
There was nothing wrong with the original kernel config, as both rtc and
genrtc were set to be compiled as modules.
What you need to do is find where the system is deciding to load genrtc
and make it
-Original Message-
Thanks for the input Tony, but the instructions that Rob Thomas wrote
took care of my issue.
Thanks again to both of you!
You're welcome, Happy to help.
--Rob
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On Sun, Sep 04, 2005 at 07:10:29PM -0600, Michael Welter wrote:
I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates).
The system has a TE110P card, and zaptel.conf is configured for an E1.
When I do a 'zaptel stop' I get a kernel panic.
Did you stop asterisk first?
--
AbdelRahman Tarzi ha scritto:
If you wish to connect it to an FXS you will need a special cable which
Nokia sells..
you don't really need a special cable for FXS, the cable is a standard
phone cable with a j11 4/6 pin plug. Just read the tech manual from the
nokia website for the pinout.
Hello,
I solved the problem - i was setting wrong caller-ID and thus got rejected.
Thanks for help.
On 05/09/05, Konrads Smelkovs [EMAIL PROTECTED] wrote:
Here you go, eagerly awaiting comments:
-- Executing SetCallerID(SIP/xlite1-e0a7, 0) in new stack
-- Executing
Hello,
I have [EMAIL PROTECTED] with zaphfc patch applied
(http://dondisperato.blogspot.com/), but I can not make call to
legacy PBX (Alcatel 4400). I can only accept incoming calls.
I am dialing with this: exten = 202,1,Dial(Zap/g1/242)
---
asterisk1*CLI bri debug span 1
Enabled
hi
i've heard it should be possible, but i can't find out how...
I want to configure a bunch of asterisk boxes to do SIP/PSTN
connectivity, and I need SER or something to do some balancing in
front of them. The requirements are listed below.
* SER MUST accept and load balance incoming
Sure yourself that your card haven't IRQ shared.
In this case (you have IRQ conflict) change your card of PCI slot, or
modify IRQ assignment on BIOS and try again unload wcte11xp/zaptel
drivers.
Tzafrir Cohen wrote:
On Sun, Sep 04, 2005 at 07:10:29PM -0600, Michael Welter wrote:
Hello configuration as follows, dial-out works:
capi.conf:
[hfcpci]
;;PointToPoint (55512-0)
isdnmode=MSN
incomingmsn=*
;msn=61
controller=1
devices=2
context=incoming
extensions.conf:
[incoming]
exten = _XX,1,Playback(demo-abouttotry)
exten = _XX,n,Dial,SIP/xlite1
exten = _XX,n,HangUp
When
I have a TDM card with one FXO and one FXS. I am trying to make sure I
understand correctly the TX and RX Gain in the Zapata.conf correctly. If
I have a phone cord plugged into an FXO port tied into a POTS line and
boost the TXGain, am I correct in thinking that the audio going back to
the phone
In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
Thanks for the input Tony, but the instructions that Rob Thomas wrote
took care of my issue.
That's good news. I saw what Rob wrote, and hadn't been aware of HPET
before, so I was glad to find out.
Cheers
Tony
--
Tony
I have a TDM card with one FXO and one FXS. I am trying to make sure I
understand correctly the TX and RX Gain in the Zapata.conf correctly. If
I have a phone cord plugged into an FXO port tied into a POTS line and
boost the TXGain, am I correct in thinking that the audio going back to
the
On Mon, 5 Sep 2005, Konrads Smelkovs wrote:
Hello configuration as follows, dial-out works:
capi.conf:
[hfcpci]
;;PointToPoint (55512-0)
isdnmode=MSN
incomingmsn=*
;msn=61
controller=1
devices=2
context=incoming
extensions.conf:
[incoming]
exten = _XX,1,Playback(demo-abouttotry)
Wel I tried by hand first that failed.. so I emptied the table..
and tried the perl script. That gave errors of category can not be
NULL. and didn't insert anything into the table.
If I allowed NULLS for category it put things in pretty much exactly
how I put them in...
MySQL RealTime
Armin Schindler ha scritto:
There are no more messages?
SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not
signal the call to Asterisk.
The sending complete field is pretty new in the libcapi, maybe he just
need to update the capi20 lib.
Sergio
Hi
I have the following error in the logs when trying to login to
the Manager API:
... tried to authenticate with non-existant user 'admin'
Login is as follows:
Action: Login
Username: admin
Secret: secret
Action: Originate
Channel: SIP/snom
Context: default
Exten: 2412
Priority:
Hello,
I am working in non profit organisation Brailcom which develop Free
Software for blind and visually impaired people. Now we think about a
new switchboard for our current work and for better communication with
our blind clients. If I good understand can be useful asterisk with
some hw card
In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
Hi
I have the following error in the logs when trying to login to
the Manager API:
... tried to authenticate with non-existant user 'admin'
= managet.conf is ===
I assume you mean manager.conf :-)
;
Thanks for the help.
Ok, it is now authenticating.
But the command:
Channel: SIP/snom
Context: default
Exten: 2412
causes no action and nothing in the logs.
Any idea?
Thanks a lot
Eric
Tony Mountifield said:
In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
Hi
I have
On Mon, 5 Sep 2005, Sergio Chersovani wrote:
Armin Schindler ha scritto:
There are no more messages?
SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not
signal the call to Asterisk.
The sending complete field is pretty new in the libcapi, maybe he just
Hi Kurth,
I'm in NJ. I'd be happy to help you out either on the phone or in person.
Gimme a call 973 828 1625
Mark
Kurth Bemis wrote:
Adam,
Thanks for your help.
Does anyone know or is anyone an * guru in the New Hampshire/Vermont area?
how about this example.
User1 sits at his desk, a
MySQL RealTime Static seems to see the settings as it goes through and
does the select.. but the it just kinda ignores them
Strange. Have you verified this behavior with ODBC RealTime? The code
that parses the results is virtually identical so I don't see this as a
mysql-rt specific issue.
It is connected to the PBX, alcatel omnipcx.
My libcapi20is dated Oct 21, 2004.
Where can I get the libcapi? There seems to be 100 sources and none
smells official.
On 05/09/05, Sergio Chersovani [EMAIL PROTECTED] wrote:
Armin Schindler ha scritto:
There are no more messages?
SETUP or
Hi All
I have problem with LIBMFCR2 for once Exchange
I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2.
my system:
Asterisk CVS 1.1.X
I've not yet tried ODBC.
And if I do the select statement that it's doing I get back the
results it wants.
Does anyone have documentation on the fields?
`id` - Assume just a key ID.
`cat_metric` -- ?
`var_metric` -- ?
`commented` -- assume if the value is commented ; or not.
`filename` --
I am getting an error compiling latest stable version from CVS, but compiling
CVS-HEAD on the same machine compile ok.
I have installed TE110P
the error is
chan_zap.c: In function `zt_handle_event':
chan_zap.c:2772: error: `ZT_EVENT_DTMFDIGIT' undeclared (first use in this
function)
Thanks Guillermo.
Can you share your experience with the software ?
Network Architecture, Linux Kernel, etc. ANI, DNIS, etc
And very important, version of the unicall library, and if you had any
problems receiving and making calls.
How many calls do you have to outside ?
Can you shara with us
I've seen some programs that install as a printer and create an image.
However this would be to cumbersome for your average user.
It would need to be able to print to as local printer and then send out
Asterisk.
Chris
- Original Message -
From: Arne Morten Johansen [EMAIL
Hello,
I have the following setup:
(*)---IP---Micronet 5012 H.323 box --- POTS --- PBX (Alcatel OmniPCX)
Grand idea is to use the micronet's POTS interfaces to connect SIP
phones to the PBX and to the PSTN. I think i even managed my way in
the arcane and cryptic management interface of that
How is your line provisioned?? (EW, PRI, Trunks, etc.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wright
Sent: Monday, September 05, 2005 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
exten = ,1,GotoIf($[${CALLERIDNUM} = 2000]?3)
exten = ,2,GotoIf($[${CALLERIDNUM} = 2001]?4:5)
exten = ,3,WaitExten(10)
exten = ,4,WaitMusicOnHold(60)
exten = ,5,Hangup()
If the caller ID of the caller is 2000 then run WaitExten(10)
if it's not 2000 and it's 2001 put the
Hi,
I was running asterisk using the init.d script. Turns out if I don’t use the
init.d script and run asterisk either directly 'asterisk -vg ' or
putting safe_asterisk in rc.local then the cpu utilization problem does not
happen anymore.
Hi,
My variant is standard ITU, I tried
Hi guys,
one question:
I've got 2 IAX accounts, and I would like to let use them in the same time, so that if one is busy I can call using the other?
Thanks-- .:FaberK:.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing
I want to authenticate a user before he is able to use the phone. I also
want to set his privilege as to where he is allowed to call to...
Preferably, the password should be their VoiceMail password, (every
extension (or is that user?) can have voicemail defined - even if its
not in use?)
Hi All.
I have notice a problem with FM feature (screen macros) on Asterisk CVS
version.
When call goes via IAX and calling part accept the call on Dial
command with option M, in macros context it's setting
MACRO_RESULT=CONTINUE, but anyway it hangups both channels.
If anyone faced with such
In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
Thanks for the help.
Ok, it is now authenticating.
But the command:
Channel: SIP/snom
Context: default
Exten: 2412
causes no action and nothing in the logs.
Any idea?
Well in your original post you had:
Action:
Hi,
Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ?
For me so far no success.
Bartosz
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
I'm seeing rather odd behaviour on a new box with TE110P card.
I'm running the TE110P span with ccs,hdb3,crc4 in pri_net, connected to a
second machine with a TE410P in pri_cpe.
The span is idle.
I'm using pri intense debug span 1 and can see the RRs going back and
forth.
So - things
I found this great link, it has both putty and winscp
available for download (with asterisk these are invaluable tools).
http://www.cs.sunyit.edu/network/downloads/
It also has one of the last downloadable copies of PGP 8.1
that I know is available (still the best pgp program and
Oh, and I am using chan_cap via mISDN on HFCPCI.
On 05/09/05, Konrads Smelkovs [EMAIL PROTECTED] wrote:
It is connected to the PBX, alcatel omnipcx.
My libcapi20is dated Oct 21, 2004.
Where can I get the libcapi? There seems to be 100 sources and none
smells official.
On 05/09/05,
[EMAIL PROTECTED] wrote:
But, pri show span 1 shows the span now as Provisioned, Down, Active.
But all along we are exchanging RRs (receive ready) with the remote
system.
The telco has turned your circuit 'administratively down' so their
operator console would stop getting spammed with
On Mon, 5 Sep 2005, Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
But, pri show span 1 shows the span now as Provisioned, Down, Active.
But all along we are exchanging RRs (receive ready) with the remote
system.
The telco has turned your circuit 'administratively down' so
[EMAIL PROTECTED] wrote:
Thanks for taking the trouble to actually read my post.
Doh! Blame it on my weekend laziness :-)
The other end of the circuit is another Asterisk box.
Hmm... I have never seen that happen before, Asterisk is pretty
aggressive about bringing the D-channel up as
Carlos:
I have no problem. I was answering a question. :)
In fact I'm managing around 1 call/day. Argentina uses no ISDN standard
by default, but the old R2 standard. My telco is Telefonica
Guillermo
From: Carlos Alperin [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List -
On Mon, 5 Sep 2005, Konrads Smelkovs wrote:
It is connected to the PBX, alcatel omnipcx.
My libcapi20is dated Oct 21, 2004.
Where can I get the libcapi? There seems to be 100 sources and none
smells official.
You can get the sources from isdn4linux.de via CVS:
cvs -d :pserver:[EMAIL
Hi.
I'm using kernel 2.4.27 over sarge, with 0.0.2c. I'm using Argentina variant
of R2, and have no problems receiving or sending ANI with the telco. 99% of
the calls are incoming ones, but I have a small percentaje of outgoing ones
too. Using 0.0.2c I resolved all the problems I had with
Yes I did with no problems...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Monday, September 05, 2005 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11
Hi,
Did
Hi, asterisk Users, sorry for my bad English
im really newbie with this excellent pbx. But I ve a
problem with callerid num when I recive a call from PSTN.
PSTN- SipGateWay(Welltech3504)- Asterisk- BT100
How can I configure my asterisk to receive the callerid from
callers and not the
I am missing some files my grandstream phone wants to download:
bootloader.bin. I cannot find that file in release 1.0.7.11.
Any ideas ?
Bartosz
- Original Message -
From: Santiago Vega [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
It suppose to be bootload.bin . not bootloader.bin like in my previous
mail.
I am missing some files my grandstream phone wants to download:
bootloader.bin. I cannot find that file in release 1.0.7.11.
Any ideas ?
Bartosz
- Original Message -
From: Santiago Vega [EMAIL
I seem to be having a conceptual problem with the While and
EndWhile applications. It seems that on the first cycle, even if
the result of the While is false that the enclosed applications
will get run. Is this expected? It seems to be counter-intuitive,
but I don't know what the intent
Hello,
Hope somebody can help me Asterisk is behaving very
oddly and Im totally stumped! I have SER and Asterisk running on the
same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on
port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and
Sorry ,
I only did the upgrade firmware version without erros!
Software Version:
Program-- 1.0.7.11
Bootloader-- 1.0.7.1 HTML-- 1.0.7.11
VOC-- 1.0.1.0
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
I will as you suggested. And I will also post the configuration
somewhere so maybe some other newbie like me can benefit from it.
Millions of thanks for your help. If you ever travel to Panama in
central america let me know about it!!!
Thanks,
On 9/3/05, astgroups [EMAIL PROTECTED] wrote:
hello everybody,
i have updated my rpm asterisk to current cvs 1.0.9. I had been usingrpm asterisk which comeswith suse 9.2. the main reason i updated my asterisk to get the attended transfer feature. i haveinstalled anothercvs
1.0.9 asterisk in Redhat 9 and it works perfect.
here what i found:
Thanks Guillermo.
Seems like nothing special is your configuration.
I have problems with outbound calls in a R2 Line here in mexico. I
dont now what is wrong yet.
Is goot to know that is working fine for you.
Regards.
2005/9/5, Guillermo Freige [EMAIL PROTECTED]:
Hi.
I'm using kernel 2.4.27
Hi Chris, Hi Arne,
Am 5.9.2005 schrieb Chris Shipman [EMAIL PROTECTED]:
I've seen some programs that install as a printer and create an image.
However this would be to cumbersome for your average user.
It would need to be able to print to as local printer and then send out
Asterisk.
What
Hi,
I found on a forum a script that emulate a hylafax this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423
You can use the WHFC in order to send a fax to asterisk.
On 9/5/05, Harald Klein [EMAIL PROTECTED] wrote:
Hi Chris, Hi Arne,Am 5.9.2005 schrieb Chris Shipman [EMAIL
I am new with asterisk and hope somebody can help me.
Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not
Hi,
I got a problem of having to upgrade 35 Cisco 7960 phones
from default firmware of 3.1 to 7.5.
The problem I get is that when trying to upgrade I see on the tftplog that it
cant seem to find the file (8 character issue).
So I renamed the files to suit what is supposed to be in
>From my point of view I do not see any issue with that scenario.
On 9/5/05, housi mueller [EMAIL PROTECTED] wrote:
I am new with asterisk and hope somebody can help me.
Is a configuration like shown on the picture with asterisk
correct? Some phone calls arriving in Branch 1 should be
redirected
On 04:48, Tue 06 Sep 05, Asterisk Sales wrote:
hello everybody,
i have updated my rpm asterisk to current cvs 1.0.9. I had been using rpm
asterisk which comes with suse 9.2. the main reason i updated my asterisk to
get the attended transfer feature. i have installed another cvs
I am trying to trouble shoot one of my ISP's network and compare to my
other ISPs offering. Although network 1 is reasonably fast and has low
enough latency, voice quality is not good and the reason for this is not
readily apparent using standard network tools.
What tools can be used to assess
I'm having a bad time getting 3 TE410P's and one TDM400 working in the same
box.
At some point during the install of second card, wcusb starts loading. I
believe
this is one of the TE410 Cards causing this as there is no usb enabled.
Module Size Used byNot tainted
audit
AbdelRahman Tarzi ha scritto:
If you wish to connect it to an FXS you will need a special cable which
Nokia sells..
you don't really need a special cable for FXS, the cable is a standard
phone cable with a j11 4/6 pin plug. Just read the tech manual from the
nokia website for the pinout.
You cannot go from 5.3 - 7.5. You must go from 5.3 - 7.0 then to 7.5.
-Matthew
From: Sascha Ferley [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 5 Sep 2005 13:19:40 -0600
To: 'Asterisk Users Mailing List -
Am 5.9.2005 schrieb Sascha Ferley [EMAIL PROTECTED]:
Hi,
Hi there,
The problem I get is that when trying to upgrade I see on the tftplog that
it can't seem to find the file (8 character issue).
dunno about your 8 char issue.. use a sane os ;)
It seems to stall going no-where .. Any one
On Mon, 5 Sep 2005, Konrads Smelkovs wrote:
Oh, and I am using chan_cap via mISDN on HFCPCI.
Hmm, mISDN... I don't know the status of mISDN, but maybe the CAPI
interface of mISDN is not fully implemented yet!?
Does someone else on the lists know if mISDN-CAPI does provide INFO_INDs
for IE
Hi,
although I have spent a lot of time on searching the wiki and Google, I didn't
find an answer to the question whether it is possible to use Asterisk as a
GSM-Gateway.
The wiki mentions the Ateus VoiceBlue Box, but I don't want another box but
integrate the GSM gateway directly into the
On Monday 05 September 2005 15:38, Asterisk wrote:
I'm having a bad time getting 3 TE410P's and one TDM400 working in the same
box.
Good luck. The interrupt issues alone are enough to make me run for my happy
place.
How can I stop asterisk from loading this ?
Don't put everything in one
On Mon, 5 Sep 2005, Asterisk wrote:
How can I stop asterisk from loading this ?
Asterisk isn't doing this.
Asterisk doesn't load kernel modules.
You need to look at and understand the boot scripts that are loading the
modules and remove the load of wcusb.
You aren't using [EMAIL
Preparing to order a T1 (not PRI) for our asterisk box. The telco has
offered me several options that I am not sure of. Which would be best
for use with asterisk? The box has the Digium card in it, BTW.
1. Dial Tone - No, Yes - Precise, Yes - SCC
2. Framing - SF, ESF
3. Line Coding - AMI, B8ZS
Darren Wright wrote:
I can't seem to get any ZAP trunks on my TE110P to match any extensions
for incoming DID.
I've even used the exten = _X.,1And it still will not match that.
All I get is:
-- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at zap-custom,s,1 failed so
Why not? How would you solve then the Brench1/Branch2 issue??a [EMAIL PROTECTED] wrote:
From my point of view I do not see any issue with that scenario.
On 9/5/05, housi mueller [EMAIL PROTECTED] wrote:
I am new with asterisk and hope somebody can help me.
Is a configuration like shown on the
The call generate from branch2 can be send to the asterisk in Branch1
with a trunk the same think the call received from branch1 the only
thing that is not cleat how you want transfer automatically the
call received from the pstn. What rule you want use?On 9/5/05, housi mueller [EMAIL PROTECTED]
Chris Mason (Lists) wrote:
I am trying to trouble shoot one of my ISP's network and compare to my
other ISPs offering. Although network 1 is reasonably fast and has low
enough latency, voice quality is not good and the reason for this is not
readily apparent using standard network tools.
What
Hello
I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series
motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink.
Currently system is running off standard IDE hard drive - because I couldn't
get astlinux to run with my Digium TDM04B card (only PCI
Chris Mason (Lists) wrote:
I am trying to trouble shoot one of my ISP's network and compare to my
other ISPs offering. Although network 1 is reasonably fast and has low
enough latency, voice quality is not good and the reason for this is
not readily apparent using standard network tools.
What
Go T1 with PRI signaling. Farming and line coding is for all T1's. We
use ESF (extended super frame) B8ZS ( I forget B8ZS stands for but it's
a newer line coding) . If you have it avaible to you, Signaling type
should be PRI. The rest of your numbers 4-7 are in the PRI signaling.
No sound
Aisling wrote:
Hello,
[general]
context=default
port=5062
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
autocreatepeer=yes
Parameter changed. Its now called bindport.
Andres.
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