Karl Brose wrote:
Is the SIPquest server sending the 401 Unauthorized message verbatim as
you printed it here?
I.e. is the WWW-Authentcate header broken up into several lines like that?
If so, how man spaces are actually at the beginning of each new line?
Continuation lines are allowed in SIP,
We are located in Belgium and just ordered a PRA line,
the telco asked the following questions :
-
120 or 75 ohm ?
-
Support for CRC4 yes/no
?
-
B channels in 2 way ?
We will buy a digium card but which one should we buy
?
could anybody help me with this ?
Thank you
We are located in Belgium and just ordered a PRA line, the telco asked the
following questions :
- 120 or 75 ohm ?
- Support for CRC4 yes/no ?
- B channels in 2 way ?
We will buy a digium card but which one should we buy ?
could anybody help me with this ?
Thank you
Michael
Sorry for the
You can either do a make samples (which will overwrite your existing
configuration) or look in the configs subdirectory in the asterisk
source tree for queues.conf.sample.
It would be nicer if make samples would install the configs as
.sample, for example: /etc/asterisk/iax.conf.sample.
Hi,
I have a HFC based ISDN BRI card in a Fedora Core 2 box (2.6.5 kernel). I
was just wondering, is zaphfc the best way to interface this type of card
with Asterisk? I've managed to get all other types of interfacing on
Asterisk going except for BRI ISDN. I'd would really like to get BRI ISDN
Hi,
I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error
command: make
after compiling for sometime then this error appeared
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o
config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o
ulaw.o
On Fri, 2004-11-19 at 16:40 +0800, Wesley Jay Deypalan wrote:
Hi,
I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error
/usr/bin/ld: cannot find -lssl
I'm not really knowledgeable in compiling. What does this mean? Did I
missed something?
Missing lib SSL, or more
I believe the list is so big that many of us are loosing some
interesting threads. May be the admins can split the users list in some
more specific sub-lists, and the people who wants to receive all the
messages can subscribe to the sublists, or have a digest for someones,
etc.
Regards
Hi folks,
just wondering if there might be any users of these devices on the
lists.
particularly if you are using version 12.5 software.
Gary
.
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Read Asterisk install,
You need to install libssl package
Wesley Jay Deypalan wrote:
Hi,
I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error
command: make
after compiling for sometime then this error appeared
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o
Hi everybody:
Please send me your recommenation of the best fit
linux version for Asterisk application. Is there a one
stop web-site where I can download everything.
__
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Meet the all-new My Yahoo! - Try it today!
On Fri, Nov 19, 2004 at 07:34:19PM +1100, Edwin Groothuis wrote:
You can either do a make samples (which will overwrite your existing
configuration) or look in the configs subdirectory in the asterisk
source tree for queues.conf.sample.
It would be nicer if make samples would install
How can I tell the dialled number from CDR records?
We need to be able to bill our provider based on the dialled number. Is this
possible?
Ben Merrills
Griffin
Internet
T: 0870 8040862
F: 0870 8040805
W: www.griffin.com
___
Martin List-Petersen wrote:
You can't, the T100P is a unchannelized T1 card.
This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1)
If you want to use it with HylaFax you need either SpanDSP OR an analog
port on Asterisk in addition to the T100P.
A search of the mailing lists
Michael Devenijn wrote:
We are located in Belgium and just ordered a PRA line, the telco asked the following questions :
- 120 or 75 ohm ?
- Support for CRC4 yes/no ?
- B channels in 2 way ?
1) Neither. Digium cards require an RJ-45 connection. Search the
mailing list for info on this. I
Does anyone know if the 3com 3C17025 (which supports NBX phones and IEEE
802.3af ) would work with Cisco 79xx phones for PoE ?
Many thanks.
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To
Eric Wieling wrote:
Michael Devenijn wrote:
We are located in Belgium and just ordered a PRA line, the telco asked
the following questions :
- 120 or 75 ohm ?
- Support for CRC4 yes/no ?
120 ohm is an RJ45 connection.
YES for the CRC that should be standard for EuroISDN
Do not have a clue
Hi,
Did anyone make sense out of the snom dialplan
strings? I am struggling with it trying to get the
phones to dial 4 digit extensions and 10 digit
numbers without the need for the OK button.
upgrade your phone to 3.56 firmware and
use |([0-9]{4})|sip:[EMAIL PROTECTED]|d to auto dial 4 digit
I'm still stuck this/my problem.
Even if I create a friend entry and register my softphone directly to
Asterisk, the Dail(${EXTEN},entity) seems to replace the From: part with the
From: 349525 sip:[EMAIL PROTECTED]:5065 part instead of the
From: 349525 sip:[EMAIL PROTECTED]
So if I
Asterisk wrote:
Does anyone know if the 3com 3C17025 (which supports NBX phones and
IEEE 802.3af ) would work with Cisco 79xx phones for PoE ?
Many thanks.
I very much doubt it. I bought a 4400 PWR to test with our Siemens
Optipoint handsets, which also support 802.3af. The two do not work
Title: R: [Asterisk-Users] problem with zyxel prestige 2002
We also had this problem with the zyxel 2002. Upgrade to the latest firmware, then it will work. Older firmwares had trouble with incoming calls behind NAT.
-Manuel
-Messaggio originale-
Da: Stig Thune [mailto:[EMAIL
On Wed, 2004-11-17 at 18:32, Alvaro Gonzalez wrote:
I need to know if it is possible to use the IVR and Voicemail using G729, I
have two SIP phones that uses G729 and I can not heard the IVR and the voice
mail.
Yes, you just need to purchase a G.729 licence from Digium. 2 phones,
$20
Hello all,
i'm experiencing a list of unpredictables hangup on SIP phones using a PRI
E100P Card.
All i can see in logs is
WARNING[81931]: File chan_zap.c, Line 5402 (zt_pri_error): PRI: Read on 37
failed: Unknown error 500
I receive a lot of these errors in asterisk/messages.
It doesn't seem
kido noagbodji wrote:
Hello,
I just downloaded and installed the latest version of asterisk under
Fedora. (had it under FreeBSD but was having TOOO many problems)
After my installation i noticed that the channel H323 was not included (
I remember that i did not have to install it under
Hi
I have read on the list about various problems with
the the X100P and have tried some of the suggestions but still have
problems.
I am using the X100P to connect to a Siemens
Gigaset 4175 which is an ISDN PBX with DECT extensions. The Gigaset also has two
POTS ports and I am trying to
Before I raise this as a bug, it appears that * incorrect sets and reads
the caller-id field from incoming sip packets when a Cisco gateway doesnt
send one.
Actually, dug into this further, and its an issue with reading
Remote-Party-ID headers from the Cisco in get_rpid_num, so I've raised a
I'm going to get 2 T100P cards. One for our Asterisk server and one for
the HylaFax Server. Will this work?
My next question is can I have Asterisk detect fax tone and route the
call to an extension. You call 555-1212 and it's a voice call it goes to
his SIP phone. If it's a fax route call to
Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown
error 500.
Specifically:
http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html
http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html
Both CRC4 off and CRC4 on will work fine with those cards.
Since its a belgian carrier, i probably already set it up in the
past. so if needed i could do it again :)
zoa.
Theodoros Georgiou wrote:
Eric Wieling wrote:
Michael Devenijn wrote:
We are located in Belgium and just ordered a PRA
Hi,
I like to know why iLBC and GSM generate a 40-50kbps bandwidth
Is very high, if compared with your calculations for other codecs(G723.1
/ 17kbps and G729 / 24 Kbps).
Alexandre
Kanuri, Seshu (Company IT) wrote:
/SNIP/
Some corrections are needed: 6.3kbps of G723.1 will become around
Im new to the asterisk world and have been playing
with an asterisk server with 1 FXO card for a couple of weeks.
Now Im looking to start adding IP Desk Phones but Im
unable to come to a decision on what phones to use.
I like to look of the Polycoms, but because we are not a phone
You can either do a make samples (which will overwrite your existing
configuration) or look in the configs subdirectory in the asterisk
source tree for queues.conf.sample.
It would be nicer if make samples would install the configs as
.sample, for example:
For real life bandwidth tests : check the ppt on www.astertest.com
Zoa.
alexandre::aldeia digital wrote:
Hi,
I like to know why iLBC and GSM generate a 40-50kbps bandwidth
Is very high, if compared with your calculations for other
codecs(G723.1 / 17kbps and G729 / 24 Kbps).
Alexandre
Kanuri,
Just got the message below from the Pliva people. Does someone have
admin access to the list to remove him?
Begin forwarded message:
From: Modric, Kristijan [EMAIL PROTECTED]
Date: November 19, 2004 6:57:47 AM EST
To: [EMAIL PROTECTED]
Subject: RE: MARIO SPOLJAR is not longer working for PLIVA
RE: the S100Us - I think you can get them from www.tjnet.com (TigerJet).
You are probably after their USB to RJ11 adapter. I think that the
Personal Phone Gateway-PCI cards are generic X100Ps too (they look
identical except no heat sink glued to the chip). I'm guessing that
TigerJet supply
I just avoid people who think it's ok to create proprietary extensions
to free software. People like that should be ashamed of themselves,
as it's just an insult to the people who have freely contributed to
the project.
I fully agree.
How hard would it be to integrate OpenSS7.org with Asterisk
I have an asterisk box with a public IP for people on the Internet to
connect to. I also have a Lucent TNT on the same physical network but on a
10.0.0.0 subnet. It isn't safe to put the TNT on a public IP address and I
never want it to talk to the net directly anyhow so this seemed like a
good
Matthew Crocker wrote:
I just avoid people who think it's ok to create proprietary extensions
to free software. People like that should be ashamed of themselves,
as it's just an insult to the people who have freely contributed to
the project.
I fully agree.
How hard would it be to integrate
Hello,
i try to send out a sms, but with no success.
The trunk is a E100P, and the sms should go out to the
Telekom SM-SC. What i want to to at the first run is,
sending out a sms when a certain number is dialed.
I tried:
In extensions.conf:
exten =
canreinvite=no ?
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tracy R
Reed
Sent: Friday, November 19, 2004 6:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Routing between different
Hello all,
I just register my
asterisk with Digium g729 codec. But now when I place a call with my SIP phone
through my Cisco ATA 186 box, I have this error: rtp.c:319 process_rfc3389: Don't know how to handle RFC3389
for receive codec 256. Can some body tell me why??
Part of my
Cisco 79xx phones are NOT 802.3af compliant or even compatible. If you have
a mid-span 802.3af injector, this can work with the phone, provide you
follow the instructions at -
http://www.voip-info.org/tiki-index.php?page=Cisco%20POE
If you have an end-span injector, such as 3-com switch forget
On Fri, Nov 19, 2004 at 07:44:34AM -0600, Tim Jackson spake thusly:
canreinvite=no ?
I already thought of that and canreinvite is already set to no. I also
know about bindaddr and localnet but neither of those do what I want
either. Thanks.
--
Tracy Reedhttp://copilotcom.com
This message
From: Eric Wieling [EMAIL PROTECTED]
Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown
error 500.
Specifically:
http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html
http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html
Derek Conniffe schrieb:
Re: the S100Us - I think you can get them from www.tjnet.com (TigerJet).
You are probably after their USB to RJ11 adapter. I think that the
Personal Phone Gateway-PCI cards are generic X100Ps too
Do you know if the USB phone and the USB IP Phone adaptor is Linux
Stefano Finetti wrote:
From: Eric Wieling [EMAIL PROTECTED]
Google: Results 1 - 10 of about 149 from lists.digium.com for
Unknown error 500.
Specifically:
http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html
Helo test brazil
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Eric,
What state are you in?
Ron
Original Message
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unpredictables Hangups
Date: Fri, 19 Nov 2004 08:10:32 -0600
Stefano Finetti wrote:
From: Eric Wieling [EMAIL PROTECTED]
Google: Results 1 - 10 of about
Hello all,
I'm provisioning a T1-PRI for a Digium T410P with
my local TELCO.The TELCO has asked me to picka line protocol
and has theoption of several RBS protocols, like 5ESS (Lucent), IN2 and
others. The switch is a 5ESS, but the "normal" (according to the sales
rep) protocol is IN2. I
Citat Eric Wieling [EMAIL PROTECTED]:
Martin List-Petersen wrote:
You can't, the T100P is a unchannelized T1 card.
This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1)
If you want to use it with HylaFax you need either SpanDSP OR an analog
port on Asterisk in
New user *
Test Brasil
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Near New Orleans Louisiana, but I am interested in long term, part time
consulting work in the Toronto, ON area.
[EMAIL PROTECTED] wrote:
Eric,
What state are you in?
Ron
Original Message
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unpredictables Hangups
Ok I've just joined and attempted to search the archives but have not found
anything...
Is Fedora Core 3 Supported?
Directions for Fedora core 3 install if available would be nice. If not I'll
be attempting it anyway and can start a crude set. Assuming that they do not
does anybody have a
Hi,
For the last 40 days i've been using Unicall on an Asterisk connected to
an Ericsson MD-110 PBX.
It was working fine for two weeks when there were just some random calls
but for the last two weeks when the load increased to between 5 and 10
simultaneous calls the system became unreliable
Martin List-Petersen wrote:
Citat Eric Wieling [EMAIL PROTECTED]:
Martin List-Petersen wrote:
You can't, the T100P is a unchannelized T1 card.
This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1)
If you want to use it with HylaFax you need either SpanDSP OR an analog
port on
On 19/11/2004 21:13 alexandre::aldeia digital said the following:
I like to know why iLBC and GSM generate a 40-50kbps bandwidth
Is very high, if compared with your calculations for other codecs(G723.1
/ 17kbps and G729 / 24 Kbps).
the other codecs have better compression, but there's a
On 19/11/2004 21:30 Steve Underwood said the following:
I can't imagine anyone successfully integrating openss7 into anything. I
believe it works OK on its own, and is in use as a gateway. It wasn't
as a gateway between what ? if it's SS7 on one side, what's on the other ?
SIGTRAN (SS7 over IP)
Hi again Stefano,
I noticed your E100P card generates 10 times as many interupts as your timer -
don't know if that could be the issue.
On my own system the E110P and two TDM400P cards generates aprox. the same
number of interupts as the timer.
[EMAIL PROTECTED] root# cat /proc/interrupts
Anybody else having broadvoice
problems?
-- Executing SetAccount(SIP/101-d03b, LD) in new
stack
-- Executing Dial(SIP/101-d03b,
SIP/[EMAIL PROTECTED]) in new stack
-- Called
[EMAIL PROTECTED]
-- Got SIP
response 408 Request Timeout back from 147.135.0.128
== No one is
Dinesh Nair wrote:
On 19/11/2004 21:30 Steve Underwood said the following:
I can't imagine anyone successfully integrating openss7 into
anything. I believe it works OK on its own, and is in use as a
gateway. It wasn't
as a gateway between what ? if it's SS7 on one side, what's on the
other ?
Hello all, somebody
can tell me how h.323 status is? it is working OK?... it has implemented
faststart and tunneling per peer based?...
thanks a
lot!!
Sebastian from
Argentina.
---
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.791 / Virus Database: 535 -
Hi all,
I have Asterisk sat between the PSTN and a PBX. Input and output is E1 PRI
When people from the PSTN call a line on the PBX which is engaged, the
line just sits there silently until they hang up.
It is there in the Wiki, but not where I was looking.
A working way to handle
Ditto. There's another very clear advantage to OpenVPN over IPsec,
and that's the fact that many firewalls are hard to run IPsec through,
but OpenVPN, using a single ephemeral UDP link, will work just fine.
I believe that the original poster is not concerned with getting it
through a Linksys
I have my SPA-3000 taking a PSTN line inbound and forwarding it to my
Asterisk server after a few rings. I don't hear any dial tone when I
do that kind of forwarding. I do it via the dial plan and I also tried
it via CFwd SelX Caller/Dest. How are you attempting to do it?
I am just starting in
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having
an issue with the zaptel init script..
If I run..
#modprobe zaptel
#modprobe wcfxo
#modprobe wcfxs
.. from a command line it load and appears to be working fine..
If I try and use the init script I get errors about
Michael,
I just check'd my kernel configuration...
I have APIC support and no Enhanced Real Time Clock, exactly as you have on
your hardware.
It *could* be a timer issue, except that i can't manage how to accelerate
mi timer or to slow down my t1xxp driver...
--
Stefano
--
Outgoing mail is
Ok, it seems that by executing a Playback prior to GET DATA, you won't hear
the audio from get data a majority of the time. When I changed the playback
to stream_file, it worked. However, I don't hear the first please enter
your, I only hear card number, then press pound. Also, after I have
Assuming that they do not does anybody have a set for Fedora core 2?
Unfortunately I don't have the Hardware to go with it just playing and
testing the server and yes I'm using it on FC2. It compiled fine and
was able to connect to the testing server useing CLI.
--
Mike Ramirez [EMAIL
On 19/11/2004 22:44 Steve Underwood said the following:
as a gateway between what ? if it's SS7 on one side, what's on the
other ? SIGTRAN (SS7 over IP) on top of SCTP ?
Yep, that kind of gateway. He has his own SCTP, and doesn't use the
native Linux 2.6 one.
in which case, if * got itself a
Fred Skrotzki wrote:
Is Fedora Core 3 Supported?
Fred,
I've just installed FC3 on a new box and will be installing Asterisk
today. I've done it a couple times and had no problems with the compile
and install. Just starting to learn *. I haven't gone beyond the
compile/install and play
Dinesh Nair wrote:
On 19/11/2004 22:44 Steve Underwood said the following:
as a gateway between what ? if it's SS7 on one side, what's on the
other ? SIGTRAN (SS7 over IP) on top of SCTP ?
Yep, that kind of gateway. He has his own SCTP, and doesn't use the
native Linux 2.6 one.
in which case,
We are running a call queue - with, say, 5 agents, and have a requirement to
record all agents calls.
Incoming calls to a queue (555-1234) are being monitored correctly
outgoing calls from an agents extension (where they have logged on) using
AgentMonitorOutgoing are being recorded correctly
Eric,
What about some consulting in Metairie. We are working with asterisk
in our Metairie office and could use some consulting. Can you help
us?
Ron
Original Message
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED], [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unpredictables Hangups
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100,
Here is what I was trying to do
Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using
a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax
server.
My question is will a Wildcard T100P work in a Hylafax server?
-Original Message-
From: [EMAIL
Sir,
I am using FC3 with no problem. I have the T1 card.
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Eric Hall wrote:
Here is what I was trying to do
Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using
a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax
server.
My question is will a Wildcard T100P work in a Hylafax server?
No it will not.
Your only option
Your actual question then is can the zaptel driver be connected with to
a faxgetty? faxgetty expects a serial port, if I am not mistaken. So,
can zaptel give me a pseudo-serial port I can use with faxgetty?
Not having tried it myself, my expectation would be that it can not.
Greg
Eric Hall
Eric Hall wrote:
Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using
a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax
server.
My question is will a Wildcard T100P work in a Hylafax server?
That depends on your definition of work. If you mean will it
A T1 PRI should be B8ZS, ESF. The protocol
can be either 5ESS or NI2(not IN2). Either will work, primarily both ends
need to be setup for the same protocol, but I would go with NI2 as that is a
more 'universal' procotol(not switch specific like 5ESS).
Lyle
- Original Message -
My BroadVoice account has been down for over a week with neither an
explanation nor a service credit. Our problems may be a little
different though because I don't remember what happened when I tried to
dial out. I know that I do get a Request Timeout error while trying
to register though.
On
I had problems with the init script not working ing FC2 also. I fixed it by
editing the init script and changing 'insmod' to 'modprobe'. Don't know if
that will fix your problem or not, but it's worth a try.
--
Jim Dossey Computer Services
-- Original message
Hi,
Have you configured features.conf file? the line which
enabled call pickup is commented and you have to un comment the line for call
pickup to work. Also you can define the numbering for call pickup
there
Thanks.
Yusuf
Alakavuk
Teknik Danman - Technical
Consultant
Grid Biliim
On 2004.11.19 07:47 Eric Hall wrote:
My question is will a Wildcard T100P work in a Hylafax server?
This question would be best fielded on the [EMAIL PROTECTED]
mailing list, but the simple answer to your question is, no.
The real answer to your question, though is this:
PRI - T100P -
Kevin P. Fleming wrote:
Eric Hall wrote:
Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using
a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax
server.
My question is will a Wildcard T100P work in a Hylafax server?
That depends on your definition of work.
On Thu, 18 Nov 2004 17:23:28 -0800, Adam Fineberg [EMAIL PROTECTED] wrote:
Having some trouble with segfaults and sound quality all of a sudden (since
I recompiled from the latest source) when 2 iaxComm clients connect. First
off immediately after the server reports:
-- Attempting native
-- Original message --
From: WipeOut [EMAIL PROTECTED]
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having
an issue with the zaptel init script..
If I run..
#modprobe zaptel
#modprobe wcfxo
#modprobe wcfxs
.. from a command line it
- Original Message -
From:
Yusuf Alakavuk
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' ; 'Walt Reed'
Sent: Friday, November 19, 2004 5:02
PM
Subject: RE: [Asterisk-Users] Call
pickup
Hi,
Have you configured features.conf file? the
FYI
SBC Makes VoIP Moves
SBC has indicated in an FCC filing that it plans to file a federal tariff
that will establish fees to be paid by ISPs that deliver VoIP calls to SBC's
circuit switched end users. This service would not be mandatory. The rates
for this service would be higher than the
Dear Users,
i have the following scnario.
1. Alcatel PBX with e1 module
2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1
connected to alcatel pbx.
i m having problem in outgoing from alcatel.
incoming from pstn - asterisk - alcatel working fine, but outgoing from
Hi,
I'm having terrible trouble getting a Tecom IP2005 Sip phone working
with Asterisk 1.0
I installed Asterisk couple weeks ago, then installed a X100P card and
tested with X-Link
softphone, all seemed well.
So I thought I would buy a Sip phone from a UK company.
However I cannot seem to get it
[EMAIL PROTECTED] wrote:
I had problems with the init script not working ing FC2 also. I fixed it by
editing the init script and changing 'insmod' to 'modprobe'. Don't know if
that will fix your problem or not, but it's worth a try.
--
Jim Dossey Computer Services
Hi Jim,
Thanks for that,
/SNIP/
My BroadVoice account has been down for over a week with neither an
explanation nor a service credit. Our problems may be a little
different though because I don't remember what happened when I tried to
dial out. I know that I do get a Request Timeout error while trying
to register
FuturaHost.Com Lists [EMAIL PROTECTED] wrote:
I believe the list is so big that many of us are loosing some
interesting threads. May be the admins can split the users list in some
more specific sub-lists, and the people who wants to receive all the
messages can subscribe to the sublists, or
John Millican wrote:
-- Original message --
From: WipeOut [EMAIL PROTECTED]
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having
an issue with the zaptel init script..
If I run..
#modprobe zaptel
#modprobe wcfxo
#modprobe wcfxs
.. from a
On November 19, 2004 11:43 am, Kevin Walsh wrote:
You'll find that many people will want to be subscribed to all of the
mail lists - just in case something interesting is said or asked.
Personally I subscribe to -users, -dev and -cvs.
You'll also find that some Muppets will post their
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 19, 2004 10:52 AM
Subject: [Asterisk-Users] Fedora Core 3 supported?
| Sir,
|
| I am using FC3 with no problem. I have the T1 card.
|
Has Core 3 been made to behave like Core 1 with
On Nov 18, 2004, at 8:25 PM, Steven Critchfield wrote:
Could someone get their hands on the driver to give it a good look and
inform of licensing. IT mentions linux, and it mentions that it is
channelized down to 672 DS0s. Sounds like the perfect card.
Also, since you can get PCI-PMC carrier
Hi all,
I have a tdm04b card with 4 fxo's connected to 4 POTS of a media
gateway. Supposing that I want to place the following calls:
Zap/1 dials Zap/2 (by placing in /spool/outgoing a call file which
dials a number corresponding to Zap/2)
Zap/3 dials Zap/2 (also placing another call file)
Zap/4
Has anyone had any success setting up a 7970 to work with asterisk. I have
searched all over and not found very much. Any advise would be greatly
appreciated.
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