The VoIP Connection wrote:
It's here:
http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/gxp2000_config.txt
Very interesting, it wasn't available at Grandstream's site :-)
Thanks! I will adjust some things on the page now I have the new
template..
Damon Estep wrote:
I assume the bandwidth is being donated or something, but surely someone
would be willing to donate reliable bandwidth as the knowledge hosted on
the site (which is also donated!) is worth way more than the bandwidth.
Sure it's the bandwidth? If the wiki is loaded, I see
I have the same problem.
seems that tdm400b is not working on CVS HEAD
On 6/18/05, Steve Totaro [EMAIL PROTECTED] wrote:
did you udate first?
- Original Message -
From: David Romero
To: Asterisk-Users@lists.digium.com
Sent: Friday, June 17, 2005 9:36 AM
Subject:
On Tue, June 21, 2005 23:07, Kristof Hardy said:
Francesco Peeters wrote:
I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
pricing for these in Europe, so I'd like to hear from people here
whether
that is a reasonable price for them?
Prices I know are around 99 EUR,
I have a box running [EMAIL PROTECTED] 1.0 and I noticed today when I
went to change some settings to do with the DIDs
that it is no longer detecting the different lines.
I have a Digium 4 port line card
and Im pretty sure that the DIDs
used to work when I used fxs_ks signaling on them.
Hello,
I wish to use asterisk as voicemail system with ser
My phones are registered on SER and asterisk provide
conference ivr and voicemail system.
I want to use Asterisk Realtime Architecture to store
voicemessages and configuration .
Michael Shuler has written a patch for mwi.
Hi,
install oh323... see link for installation
http://lists.digium.com/pipermail/asterisk-users/2005-April/100061.html
--- craz sead [EMAIL PROTECTED] wrote:
Hi all
could somebody help me how to install and setup H323
i
would like to connect asterisk box with
huawei/cisco,
but i still
My configuration is one TDM31B for internal subscribers and one AVM B1 v.4 for incoming/outgoing calls.
How do I:
a) transfer a call to an internal subscriberb) commence a 3-way conference (two internal - one external subscribers)
I've searched voip-info.org and google god but regarding to zap
hi,
recently I installed zeroconf for asterisk...
I've already followed the asterisk+zeroconf how to (which is too short),
but it came with an error message...
asterisk: relocation error: /usr/lib/asterisk/modules/res_zeroconf.so:
undefined symbol: DNSServiceRegister
Ouch ... error while
Hello,
My asterisk crash from time to time, at least twice a day.
Reomendation from a site is to do a gdb on core dump
Below is what i get, but i have no idea what is going on
Does anybody have any idea?
(gdb)bt
#0 0x00181aed in _int_malloc () from /lib/tls/libc.so.6
#1 0x00180dfd in malloc
On Wed, Jun 22, 2005 at 08:35:58AM +0200, Francesco Peeters wrote:
The shop I saw these also sells - pretty cheap - little devices (forgot
the name, they look like a translucent blue ice-hockey puck) that do SIP
conversion for analog telephones or PBX extensions. (I am thinking
migration
Hello,
I make a sawdwish course in network and software engeneering at CPE lyon and
in my company I'm working on Asterisk from 1 year.
So, I'm looking for a internship (3 month) in a english country on a Asterisk
project.
Thanks,
Thomas DEILLON
___
On Wed, June 22, 2005 10:02, Ming-Wei Shih said:
On Wed, Jun 22, 2005 at 08:35:58AM +0200, Francesco Peeters wrote:
The shop I saw these also sells - pretty cheap - little devices (forgot
the name, they look like a translucent blue ice-hockey puck) that do SIP
conversion for analog telephones
Anyone know how you can use DBGet inside extensions.ael?
Thanks
Chris
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To UNSUBSCRIBE or update options visit:
Hello!
If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
How can I make calls from Asterisk client to NEC NEAX 2400
traditional phone ?
Is it possible to have a connection between Asterisk and NEC
NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP.
Please help me to find a solution ;;;
Thanks Regards
Ram Kumar
After a big help from Peter Svensson, I got ISDN Data-calls up and
running.
But now when everything seems connected, pppd has been authorized by
other peer and even got an IP address, the whole connection seems to
stop working.
Very unregulary, the PPPD's EchoReq's stop being answered, and of
I have a question.
I have two numbers on Asterisk like 902121234567 and 902123645789
and i want to divert first numbers call to Trunk if second number is
unregistered. Is it possible? İf yes, how?
Flow Diagram:
*Two numbers are registered on Asterisk
Steve Underwood wrote:
Robert Rozman wrote:
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.
Since Asterisk is claimed to have good dtmf
[EMAIL PROTECTED] wrote:
Hello!
If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale
___
Asterisk-Users mailing list
Hi,
I also changed as following sequences;
app_voicemail.c
1. Line 3724 tmp[256] to tmp[4096] vm_exec
2. Line 3760 tmp[256] to tmp[4096] append_mailbox
3. Line 3796 tmp[256] to tmp[4096] vm_box_exists
4. Line 3290 tmp[256] to tmp[4096] vm_execmain
5. Line 80 tmp[256] to tmp[4096] #define
I had gxp-2000 for testing some days, but features are (in current
firmware) _very_ limited!
phone does not have missed, dialed numbers, phone book, speakerphone is
useless...
phone have nice backlight display and in-line power :-)
but if you like features, grandstream is not for you...
PJ
Hi,
I am trying to send a variable to emailbody of voicemail.conf. That is
to say, I try to send I number that it is entered by the numeric keypad
of the telephone and arrives in the body of the mail with the recording
of the message.
In the single documentation they mark some variables that
I'm trying to use ZapRAS to enable ppp connection through my E1.
After the ZapRAS command is executed, all sound is crappy on all lines!
The only solution is to reboot the machine (or halt it, and then power
it on since Digium's hardware doesn't like reboots).
Anyone know how this can happen?!
Hello,
I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI
as channels. A call comes in via IAX2 and should be redirected to CAPI.
So I wrote the following dialplan:
[fromiax]
exten = _8XXX,1,Answer
exten = _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r)
[fromcapi]
exten =
Hello
I need to know who's the agent that picks up a call in a queue, preferably in
real-time so I can invoke a script that sends info to the agent's PC. It
doesn't seem easy to do unless I use the info on asterisk's log file (using
swatch or something? is this possible?) ... is there any other
Hello everyone,
So, this isn't exactly what it seems. I am not looking to integrate
Asterisk and Mambo. I am the maintainer/creator of AstLinux, and I have
recently decided that I should really have a better web site for it. I
would like to use Mambo so that I can do updates easily, from
You could try the new chan_capi-cm-0.5.1 in which you dont need to have any
msn defined in capi.conf
On Wed, 22 Jun 2005 12:28:58 +0200 (MEST), Patrik Schindler wrote
Hello,
I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX
and CAPI as channels. A call comes in via IAX2 and
Hi !
Managed to fix app_changrab.c to compile and start working under 1.0.X.
it is working on my installation, but is tested well enough.
Regards,
Nenad
Here is diff -u :
--
--- app_changrab.c.orig 2005-06-20 22:10:50.0 +0200
+++ app_changrab.c 2005-06-22
On Wed, 22 Jun 2005, Pavel Jezek wrote:
I had gxp-2000 for testing some days, but features are (in current
firmware) _very_ limited!
phone does not have missed, dialed numbers, phone book, speakerphone is
useless...
Some of these features are in the 1.0.1.9 version that was released last
On Tue, 21 Jun 2005, Leandro Morgado wrote:
Steve Underwood wrote:
Robert Rozman wrote:
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.
We
Hi,
I tried to cable #21 with a thomson cable modem mta:
-- SIP read from 192.168.153.100:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.153.100;branch=z9hG4bK1aa77a586
Max-Forwards: 70
Content-Length: 258
To: #21 sip:[EMAIL PROTECTED]:5060
From: sip:[EMAIL
I've tried to find some details on the wiki, but was unable to get a
satisfactory result, so I am asking here:
I have a Linux (FC3) box with these specs:
vendor_id : AuthenticAMD
cpu family : 6
model : 3
model name : AMD Duron(tm) Processor
stepping: 1
cpu MHz
Hi Erdem,
Can you try to put another dial command that points to the trunk afetr
the dial command to the SIP?
fro example:
exten => XXX,1, dial(sip/,20,r)
exten => XXX,2,dial(zap/) ->
note here that I am not sure if the order number should be 2 or 102 but
if this didn't work
As an asterisk server it is more than fine but asterisk prefers to be a
standalone machine.
You would have a lot less issues if you had 2 machines, one handling
file serving, SMTP and one dedicate machine for asterisk.
Voice isn't very tolerant of interrupts.
Cheers,
Dean
-Original
On Wed, June 22, 2005 13:39, Dean Collins said:
As an asterisk server it is more than fine but asterisk prefers to be a
standalone machine.
You would have a lot less issues if you had 2 machines, one handling
file serving, SMTP and one dedicate machine for asterisk.
Voice isn't very
This will work fine I have a traveling 700mhz duron that does
great. CUPS should not be running but test it your self.
Rule of thumb, if you are transcoding many channels then you need a
bigger machine. If you are just switching then you can use a smaller
machine.
On 6/22/05, Francesco
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel:
Il giorno mer, 22/06/2005 alle 07.39 -0400, Dean Collins ha scritto:
As an asterisk server it is more than fine but asterisk prefers to be a
standalone machine.
You would have a lot less issues if you had 2 machines, one handling
file serving, SMTP and one dedicate machine for asterisk.
On Wednesday 22 June 2005 07:27, Francesco Peeters wrote:
It is currently doing File/Printer serving.
Ideally I'd want it to do Asterisk (2 ISDN BRI 8 phones), File/Printer
server on a home network (3 clients) and some light SMTP ( 100 emails a
day)
Is this machine sufficient for the task?
Hi:
We have finished the translation of the FAQ of Digium to spanish.
They are already (in Spanish) available for download (in
http://ourproject.org/projects/asterix/):
* FAQ Frequently Asked Questions
* Features
* Hardware compatibility list
* Fast Installation Zaptel
All the documentation
If I try dmesg - no mention of a Wildcard TDM400.
Sorry I am fairly new to Linux. In Windows I suppose I would run some
hardware program which came with the card to see if I could manually set
IRQ's etc. What should I be looking at now?
Please feel free to point me to a good book or
hi,
is there any way to figure out what the mute status
is of the meetme conference participants?
i personally can no see any difference on the output:
kamikaze*CLI meetme
Conf Num PartiesMarked Activity Creation
5000 0002 N/A00:00:40 Static
*
What's wrong with ARA (asterisk realtime architecture)
from voip-info:
Asterisk, SER and MWI
http://mail.iptel.org/pipermail/serusers/2004-December/013727.html
Actually I wrote a patch for this and it supports
ast_data too. What you do is tell asterisk that all of
your phones IP addresses are
to work when I used fxs_ks signaling on them. However I changed to
fxs_ls signaling because the ks wasn’t detecting when people were
hanging up properly. Could changing the signaling be whats causing the
I added the following to my zapata.conf in the [channels] section:
busydetect=yes
nothing wrong with it, and ast_data and realtime are different unless I
have misunderstood.
Asterisk realtime and pulling Voicemail from a DB I have had problems
with, if it works great if someone could let me know, ast_data does I
believe work, so you could try that out, I just wanted to
Probably means that your perfectly good motherboard can't see the TDM card.
There are many motherboards that this card doesn't seem to work with,
Digium doesn't seem willing to address the issue or even acknowledge
that is the case, and usually answers try another motherboard rather
than
Someone else just reported what appears to be the same problem and he
narrowed it down to upper-lower case problem with sip digest. Don't
know if a fix has been applied to cvs-head for that as yet. Might
try an update.
md5 instead of plaintext?
Doesn't asterisk take
In article [EMAIL PROTECTED], bdz [EMAIL PROTECTED] wrote:
hi,
is there any way to figure out what the mute status
is of the meetme conference participants?
i personally can no see any difference on the output:
kamikaze*CLI meetme
Conf Num PartiesMarked Activity
I installed a new Digium TDM400P in a Dell 1750 server. The system
would not recognize the card. I took the FXS modules off of it and put
them on another TDM400P card I already had. Old card worked fine with
new modules. Old card is Rev. H and new card is Rev. I. Anyone else
having any issues
Hello
Here is what I find.
Any help would be greatly appreciated.
Angus
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 2:09 PM
Subject: Re:
On Wed, 22 Jun 2005, jurczak wrote:
You could try the new chan_capi-cm-0.5.1 in which you dont need to have any
msn defined in capi.conf
Thank you very much, this solved my problem!
Do you know a solution for non-capi but i4l devices? It's not an error but
a warning only, so it's not a real
Hi,
can you recommend cheap GSM Gateway which works with Asterisk? VoiceBlue
solution is quite expensive.
Thanks,
--
-
David
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
What could be th reason for that?
/usr/sbin/safe_asterisk: line 41: 4590 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
On Jun 21, 2005, at 5:05 PM, Adam Megacz wrote:
Matt King [EMAIL PROTECTED] writes:
I am familiar with the OSI definitiion. I've read it again, but I
can't work out exactly how asking for permission contravenes this
definition.
2) OrderlyCalls MAY NOT be used to provide or augment call
Hi all,
Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver.
[EMAIL PROTECTED] zaptel-1.0.7]# modprobe wctdm
ZT_CHANCONFIG failed on channel 3: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces
Yes it works :) but i need to add
voicemail option, how can i do this?
I want a configuration like this
exten = XXX,1, Dial(sip/,20,r)
exten = XXX,2,Dial(zap/)
exten = XXX,3,Voicemail(u)
exten = XXX,103,Voicemail(b) -what should order
number
[EMAIL PROTECTED] wrote:
Thanks for the info... I am considering getting these to
experiment with,
so I can do some testing *before* I actually get in to the
real thing. The
cheaper the test period, the better, so 2 of these (which later can
be
reused in less used area's) look pretty
Hi,
Please refer to my post here regarding a similar issue as you are trying to
achieve,
I am not sure how far you have gotten with your attempt, but I would
appreciate if you can share your experience with me and help me get mine
working.
Digium's analog cards do not support DID lines.
Syed Akbar
Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pascal
Mosimann
Sent: Wednesday, June 22, 2005 8:53 AM
To: Asterisk Users Mailing List -
As suggested by you, I have tried the timing parameter with 0 an 1 both but
to no avail.
Signalling I have tried are both em_w and featd (suggested by digium tech
support).
Zap show channels shows all 24 channels to be ok with no alarms.
Zttool also confirms that.
The problem is that the pbx
Hello,
I have downloaded and installed Flash Operator Panel. version 0.21. It
works pretty well and I have some questions about it.
1. The text label of the buttons are partially hidden by their icons. Is
there a way to adjust right margin for the buttons?
2. I would like to have the fop
We have the following versions:
App Version: 1.4.1.0040 (SIP)
Bootrom: 2.6.1.0003
I also noticed that the polycom IP600 phones are Rev 3.
--johann
Chris Coulthurst wrote:
Which software pack to you have for the IP600? Sip.ld, bootrom, etc...
Chris Coulthurst
[EMAIL PROTECTED]
Yes, I know.
In this case the agent is logging in from a remote phone (pots line) and
staying logged in. If they used agentcallbacklogin they could make
outbound calls, but the long distance bill would hit their line, not the
* box...
You could use Agentcallbacklogin instead - the queue will
In my little experience with Meetme, i have not found how to know if
certain user is muted or not, so im keeping track of the commands i
execute from the web interface, so i know if its muted or not. Its not
so hard to add a manager event, check manager.c to know how to add
events.
On 6/22/05,
404 - Not Found
:(
On 6/22/05, Leonardo F. Bauchwitz [EMAIL PROTECTED] wrote:
Hi:
We have finished the translation of the FAQ of Digium to spanish.
They are already (in Spanish) available for download (in
http://ourproject.org/projects/asterix/):
* FAQ Frequently Asked Questions
*
What about framing? ESF/B8ZS vs D4/AMI?
-Original Message-
From: Karthik Natarajan
Sent: Wed, June 22, 2005 10:01 am
As suggested by you, I have tried the timing parameter with 0 an 1 both
but
to no avail.
Signalling I have tried are both em_w and featd (suggested by digium tech
Damon Estep wrote:
I assume the bandwidth is being donated or something, but surely
someone
would be willing to donate reliable bandwidth as the knowledge
hosted on
the site (which is also donated!) is worth way more than the
bandwidth.
Sure it's the bandwidth? If the wiki is loaded, I
What type of connection are you using to link their pots with * ?
For the inbound part, * would be calling them to connect the call. For
their outbound, could you not use the same mechanism that you are
currently using to login, but dial the outbound number instead (so it is
* doing the
can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS
uses tcp, Asterik UDP.
Someone, knows how to do the configuration beetwen LCS and SER , SER and
Asterisk? the function of asterisk is SIP-PSTN Gateway for the
On 21 de jun de 2005, at 17:20, Andrew Kohlsmith wrote:
How would you have asterisk know which IP to ring if nobody is
registered
until the phone rings??
You're right Andrew. I didn't thought about the ring...
Honestly -- what's wrong with
SIP/location1SIP/location2SIP/location3 ?
For
Hi
One question for you!
Which operation are allows by Asterisk Manager API?
Can I connect to Asterisk server, create a new Channel,add channel on Asterisk
server,specify a Voip protocol like SIP and generate Sip signaling?
Thanks Ale
___
Absolutely. Here is the CLI output. I made two attempts. First, I
dialed inbound into an extension and then tried using meetme room
0201 from Server B, which didn't work. Then I dialed inbound into the
same extension and then tried using meetme room 0215 which resides in
Server A. Note
Hello,
I've been planning to replace my aging CENTREX switch with a new PBX and am
seriously
considering Asterisk as my solution.
I work at a college, and we currently support just under 300 regular analog
lines to
the offices and whatnot.
I was wondering.. Is asterisk ready for such a
I installed the TDM400P and installed it on a system running
on Fedora Core1, please refer to the steps below. I connected an analogue phone
to the FXS port.
When I pick up the speaker I don't hear the dialtone although
when I press a number key I hear a DTMF generated. What could be
I installed the TDM400P and installed it on a system running
on Fedora Core1, please refer to the steps below. I connected an analogue phone
to the FXS port.
When I pick up the speaker I don't hear the dialtone
although when I press a number key I hear a DTMF generated. What could be
On Wed, 2005-06-22 at 09:36 -0400, Tom Rymes wrote:
On Jun 21, 2005, at 5:05 PM, Adam Megacz wrote:
Matt King [EMAIL PROTECTED] writes:
I am familiar with the OSI definitiion. I've read it again, but I
can't work out exactly how asking for permission contravenes this
definition.
can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS
uses tcp, Asterik UDP.
Someone, knows how to do the configuration beetwen LCS and SER , SER and
Asterisk? the function of asterisk is SIP-PSTN Gateway for the
Hi
One question for you!
Which operation are allows by Asterisk Manager API?
Can I connect to Asterisk server, create a new Channel,add channel on Asterisk
server,specify a Voip protocol like SIP and generate Sip signaling?
Thanks Ale
___
On Wed, June 22, 2005 17:17, Francesco Peeters said:
On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
[EMAIL PROTECTED] wrote:
Thanks for the info... I am considering getting these to
experiment with,
so I can do some testing *before* I actually get in to the
real thing. The
cheaper the
although it has been announced, to my knowledge,
the NEC pure SIP fone has not yet been released as of this date
Ramkumar wrote:
Hi,
How can I make calls from Asterisk client to NEC NEAX 2400 traditional
phone ?
Is it possible to have a connection between Asterisk and NEC NEAX
2400,
On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
[EMAIL PROTECTED] wrote:
Thanks for the info... I am considering getting these to
experiment with,
so I can do some testing *before* I actually get in to the
real thing. The
cheaper the test period, the better, so 2 of these (which later
On Wed, Jun 22, 2005 at 10:41:18AM -0300, Alessandro wrote:
Does somebody know why no load modules to FXS? I used zaptel-1.0.7
version driver.
[EMAIL PROTECTED] zaptel-1.0.7]# modprobe wctdm
Here's a clue:
ZT_CHANCONFIG failed on channel 3: Invalid argument (22)
Did you forget that
I think he was referring to load measurements on the server, not the amount
of * users. A guideline of how high load should be can be taken from the
amount of CPUs on the server we're talking about. If the server runs on a
single CPU, then ideal load is up to 1, it it's two CPUs, then up to 2 etc.
Calls into the asterisk box, including non-VoIP remote agents, are via a
ISDN/PRI on a Digium T1 card.
It is the same PRI that inbound and outbound calls come in on and go out
through, there are no IP dial tone providers.
-Original Message-
From: [EMAIL PROTECTED]
asterisk2*CLI show manager commands
Action Privilege Synopsis
AbsoluteTimeout callSet Absolute Timeout
ChangeMonitorcallChange monitoring filename of a channel
Command command
[EMAIL PROTECTED] ha scritto:
can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS uses tcp, Asterik UDP.
prova http://www.winton.org.uk/zebedee/
It should proxy udp over tcp, but I didn't try it yet
Sergio
look at ser projects:
asterisk is limited to 250 channels
You need cat 5e and manage qos if you setup ip phones
Harry
--- Don Brearley [EMAIL PROTECTED] a écrit :
Hello,
I've been planning to replace my aging CENTREX
switch with a new PBX and am seriously
considering Asterisk as my
I've had several users today inform me that whilst they were on a call,
the volume kept fading in and out to such an extent that they thought
the caller had hung up.
I would dismiss this if it were a single person mentioning it, but it
isn't ..
Has anyone else seen anything like this ? We
300 phones should not be a problem if you design the system correctly.
If they are all analog sets with no transcoding your should fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Don Brearley
Sent: Wednesday, June 22, 2005 10:51 AM
To:
On Wed, Jun 22, 2005 at 09:50:42AM -0500, Don Brearley wrote:
I've been planning to replace my aging CENTREX switch with a new PBX and am
seriously
considering Asterisk as my solution.
You have a CENTREX switch? I'm confused.
Do you have a PBX?
Do you pay for Centrex services from your
Digium logged in and fixed the problem. It seems they had to fix the zaptel
source code - so not really something I could easily have done. something
about adding the subvendors ID to the cards source. So I assume a bug.
I personally feel a little indebted to Digium for sorting the problem
Hi everybody !
I'm trying to setup a TE110P card. But i'm facind this
error... and * cannot start.
[chan_zap.so]Jun 22 11:50:33 WARNING[16384]:
loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so:
undefined symbol: pri_dump_infoJun 22 11:50:33 WARNING[16384]: loader.c:429
[EMAIL PROTECTED] wrote:
On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
[EMAIL PROTECTED] wrote:
Thanks for the info... I am considering getting these to
experiment
with, so I can do some testing *before* I actually get in to the
real thing. The
cheaper the test period, the better, so
I installed a TDM400P with 4 FXO modules. Before moving all of my
office phone lines to it, I decided to move only one for testing. I
plugged it into port 4 on the card.
In zaptel.conf I have:
fxsks=1-4
And zapata.conf:
context=incoming
signalling=fxs_ks
busydetect=yes
callprogress=no
I will be out of the office until Friday June 24th. If this is an emergency
please call our office and ask for customer service.
Thank you.
Steve Hearst
Accent Communication Services Inc.
740-548-7378
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Hello,
I have read about using redirect on a sip channel to get * to step out of the
voice path. Is this possible with ISDN or maybe a US T-1? I would like to
have the * box answer a call, do some niffty IVR/AGI stuff, and then redirect
that call back to the originating switch to be passed on
We do mambo projects all the time, contact me off lists and we can get
you rolling.
JD
Kristian Kielhofner wrote:
Hello everyone,
So, this isn't exactly what it seems. I am not looking to
integrate Asterisk and Mambo. I am the maintainer/creator of
AstLinux, and I have recently
For some reason a couple weeks ago users began experiencing garbled audio
in one direction when dialing out via our VoIP provider. This happened at
multiple sites simultaneously. The VoIP provider doesn't think it's their
problem. If I switch to another codec so that Asterisk transcodes
Hi All,
Does the T1 PCI cards supported by Digium support Performance Moinitoring.
If so are they in compliance with any of GR or ANSI standards of T1.
Thanks,
Somesh
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