My $0.02
Cisco pros:
1. Very easy to configure thru TFTP.
2. Great sound quality.
3. Great tech support.
4. High value even on used.
5. Power Over Ehternet support out of the box (even 802.3af, if you
play around).
6. Excellent handling of multiple lines (using one SIP account for all
buttons will
Just noticed something more interesting. I've added an ATP IAX account and
as expected it registers perfectly first time. But the strange thing is my
external IP address is 220.244.224.206 at the moment yet this is what and
IAX2 show registry reports
Host UsernamePerceived
Matthew:
I did the cvs checkout asterisk today. I think I have the latest
version:
The trace is:
(gdb) backtrace
#0 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6
#1 0xf6eb58c0 in build_peer (name=0x0, v=0x92a42d0, realtime=0) at
chan_sip.c:9255
#2 0xf6eb67b0 in find_peer (peer=0x0, s
Matthew:
The .version file in the asterisk folder reads: CVS-HEAD-03/17/05-15:43:44
pd: I opened chan_sip.c at line 9255 and that line reads: peer =
ASTOBJ_CONTAINER_FIND_UNLINK(&peerl, name); No strcasecmp there...
Thanks,
JO
Matthew Boehm wrote:
Jose R. Ortiz Ubarri wrote:
(gdb) backtrac
Put in two Cut statement for both the "@" and "-". It looks like if it
can't find the delimiter that it just simply sets the new variable to the
exact value of the old one. Just a thought.
B. J.
-Original Message-
From: Thomas Andrews [mailto:[EMAIL PROTECTED]
Sent: Thursday, March
I'm running Asterisk HEAD from March 4. I've Googled a bit but I can't
figure out what causes this error:
app_queue.c:374 in changethread: Can't change device '**Unknown**' with no
technology!
It doesn't seem to be causing any problems, but I'm curious what causes it.
I did a few Google searches
Well, this is getting more interesting. I started looking at this this
morning and realised that Asterisk had lost registration, yet my ADSL
connection has been up for almost 2 days - and it was working fine
yesterday. Therefore it doesn't appear to be related to the IP address
changing.
I'm thi
Install a smp kernel and you will use IO-APIC instead of XT-PIC you
typically will not share interrupts in APIC mode because it has twice
the numbers of interrupts to use.
Ronald Hartmann wrote:
Good Day list,
I am having some issues with my card in that it wants to
share IRQ’s
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to ;work
; with all telcos.
; outofband: Signal Busy/Congestion out of band with
;RELEASE/DISCONNECT
; inband: Sign
On Tue, Mar 15, 2005 at 03:23:54PM -0600, Matthew Boehm wrote:
> I compiled and installed and was unable to make outbound calls. Inbound was
> fine.
> (See my previous post about call length 0). The changelog in both dirs
> showed the version as 0.1.6.
> This "should" be the newest version of libpr
Thomas Andrews wrote:
How do I get the bit like "IAX2/white_phone" in extensions.conf eg from
pre-defined variables or variants thereof ?
What I *do* get is strings like this "IAX2/[EMAIL PROTECTED]"
from ${CHANNEL}, but that's the full channel name.
What am I missing here ?
Thanks,
Thomas
On Thu, Mar 17, 2005 at 02:06:24PM -0600, B. J. Bomar wrote:
> Try using the Cut application. For your example channel you can use the
> following.
>
> exten => whatever,n,Cut(my_variable=CHANNEL,@,1)
Thanks, I thought of that, but it doesn't account for cases like Zap/1
that becomes Zap/1-1 in
Hi,
When I dial from my sip device to the extension 1234 which is linked to
the ALSA console driver the call fails with the message "No channel type
registered for 'ALSA'" (see below).
I would like to have the console autoanswer for paging.
However when I call from the console to the sip device
Nick Stein wrote:
This is probably a stupid newbie question. Is there a way to search the
list archives?
http://www.mail-archive.com
http://www.mail-archive.com/asterisk-users%40lists.digium.com/
http://www.mail-archive.com/asterisk-dev%40lists.digium.com/
extensions.conf:
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Systems Inc
; dialparties.agi (http://www.sprackett.com/asterisk/)
; Asterisk::AGI (http://asterisk.gnuinter.net/)
; gsm (http://www.ibiblio.org/pub/Linux/utils/compress/!INDEX.short.html)
; loligo sounds (http://
I'm having some problems with Meetme2 and MOH. Here's what I've done, not
sure if it's correct.
I added the following line to extensions_additional.conf
exten => 3000,1,MeetMe2
And I added conference rooms to Meetme_additional.conf
conf => 8005
conf => 8006
conf => 8007
If I dial directly into
Hi Thorben,
Thorben Jensen wrote:
Hi Kong,
No, I have no support for monitoring of Zap devices at the moment. If there
is great demand for it, I will make it.
I would also like some zap monitoring as well. Does it do IAX as well?
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Jose R. Ortiz Ubarri wrote:
> (gdb) backtrace
> #0 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6
> #1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0)
> at chan_sip.c:9255
> #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1)
> at chan_sip.c:1222
I do
Sean,
Thanks for the reply.
I was afraid this was the problem. It looks like the called server can only
negotiate on what codecs it wants to receive
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy
Sent: 17 March 2005 20:05
To: Asterisk
Oddly enough, I am in the middle of dealing with both a Cisco phone problem
and a Polycom phone problem. The Polycom problem I caused myself (oops
during a flash) and the Cisco problem came out of nowhere.
Polycom
During a flash upgrade I lost power (tripped over the cord) and the phone
was DEAD.
> #1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0)
> at chan_sip.c:9255
> #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1)
> at chan_sip.c:1222
I just updated to newest CVS and visited those line numbers above. 9255
is "in" build_peer but find_peer is n
Not really anything of use.. but here they are:
voicemail.conf:
201 => 5929,Matt Hoppes,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes
202 => 202202,Michael Eck,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes
203 => 203203,Matthew
Kiessling,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes
sip.conf:
Matt wrote:
Hi,
I have one phone on my network that just keeps ringing (when I call
it) and does not go to voicemail.
If the person there is on the phone, and someone calls it they get the
busy message, but they never seem to get the 'unavailable' message...
instead it will just ring and ring and r
Try using the Cut application. For your example channel you can use the
following.
exten => whatever,n,Cut(my_variable=CHANNEL,@,1)
That should give you your IAX2/white_phone. For more info, take a look at
either the wiki or CLI help.
B. J.
-Original Message-
From: Thomas Andrews
Please ignore,
intelligent messages will come later
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C. Tomlinson wrote:
Hi,
I have been trying to work this out and haven’t been able to.
I have some incoming numbers that come in over IAX, from the same
server, and wish to use different codecs for different calls. This
doesn’t seem to work for incoming either.
I cant seem to get any combination
On Thu, 2005-03-17 at 11:30 -0800, snacktime wrote:
> On Thu, 17 Mar 2005 11:14:01 -0700, Wiley Siler
> <[EMAIL PROTECTED]> wrote:
> > I recorded my last set of prompts over my Plantronics DSP 500 USB
> > Headset. I have also used a Logitech USB Headset. These and similar are
> > easiest to use al
As you have already started to debug this by yourself I would suggest a post
to the dev list as this might be a bug.
I'm not familiar with PRI signalling but I think you won't get any answer
here.
Jens
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David Brodbeck wrote:
This Postgres vs. MySQL business is ultimately just a religious debate, like
PC vs. Mac, Ford vs. Chevy, or Kirk vs. Picard.
With all due respect I disagree. It is much more like a public policy
debate. There are those of us in any of the Oracle, DB2, or PostgreSQL
camps
On Thu, 17 Mar 2005 11:14:01 -0700, Wiley Siler
<[EMAIL PROTECTED]> wrote:
> I recorded my last set of prompts over my Plantronics DSP 500 USB
> Headset. I have also used a Logitech USB Headset. These and similar are
> easiest to use along with X-lite or similar softphone. I used the
> suggested m
First, for an on-topic comment :-)
Which database you choose will largely have the most to do with what
applications you need to integrate your Asterisk databases with. If
those applications are based on MySQL, you may need to use that. Ditto
with Oracle, MS SQL, etc. My personal favorite is
Trevor Peirce wrote:
Anyhow, they are seeing the RELEASE COMPLETE message with cause code
1, however the tech told me they expect a PROGRESS indicator with a
value between 1 and 10.
Okay after printing off a dozen pages and taking up tons of floor space
I think I may have figured this out... but
Wow, thanks Brian! Everything I saw said the patch was only needed on
older releases. I've updated several times over the last week. I
patched two systems today, one 3/11/05 and one 3/17/05 and now they both
work. Should have posted here sooner!
Brian G.
On Thu, 2005-03-17 at 13:28, Brian Buh
Hi:
A very nice guy asked me for a trace: (I hope this is what I was
asked for)
from /var/log/messages:
Mar 17 14:04:23 NOTICE[24649]: Registered Config Engine mysql
Mar 17 14:04:23 WARNING[24649]: Unable to get our IP address, Skinny
disabled
Mar 17 14:04:57 NOTICE[24666]: Registered Config
On Thu, 2005-03-17 at 14:16 -0500, Matt wrote:
> Hi,
> I have one phone on my network that just keeps ringing (when I call
> it) and does not go to voicemail.
>
> If the person there is on the phone, and someone calls it they get the
> busy message, but they never seem to get the 'unavailable' mes
Title: Different codecs for different numbers via same IAX provider; how? Configs included.
Hi,
I have been trying to work this out and haven’t been able to.
I have some incoming numbers that come in over IAX, from the same server, and wish to use different codecs for different calls. This
On Thursday 17 March 2005 01:14 pm, Nick Stein wrote:
> This is probably a stupid newbie question. Is there a way to search the
> list archives?
google!
http://www.google.com/search?q=site:lists.digium.com+your+query+here
-Jeremy
--
Jeremy Kitchen ++ Systems Administrator ++ Inter7 Internet T
I am an Asterisk newby, and I cannot seem to get Caller ID information
from our T1 line. When calls appear at the phones, they say the call
came from "asterisk" and unknown number.
I know how Caller ID information is passed on an analog phone line
(between the rings) but with a T1 line, I don
Nick Stein wrote:
This is probably a stupid newbie question. Is there a way to search the
list archives?
http://www.google.com/custom?sitesearch=lists.digium.com
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Hi,
I have one phone on my network that just keeps ringing (when I call
it) and does not go to voicemail.
If the person there is on the phone, and someone calls it they get the
busy message, but they never seem to get the 'unavailable' message...
instead it will just ring and ring and ring... any
This is probably a stupid newbie question. Is there a way
to search the list archives?
Nick
Stein
NetPiano.com
[EMAIL PROTECTED]
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I think that should be
#INCLUDE numbers.conf
Note the caps and no ""
On Mar 17, 2005, at 12:31 PM, [EMAIL PROTECTED] wrote:
Hey guys. Thanks for the help on the Pattern matching, I got that
working pretty nicely.
the next problem I have is that I'm using an include file, but its not
really working
Because some of us, are already using radius for other purposes (radius also
has authentication and we could use it with other GWs vendors) as a single
solution, in our case, we already have radius for our GWs and Raidus using
Oracle and I don't want to use direct connection to Oracle at all.
LTe
Hello,
I just got off the phone with my PRI provider, and was told that I am
not sending an expected message when I reject a call with a Cause Code
for Unassigned(1) and Congestion (42). Busy works fine though...
Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1,
however t
Jose R. Ortiz Ubarri wrote:
> But if I
> load the module from boot or from the asterisk command load
> res_config_mysql.so then I get the Segmentation fault again.
Where is your backtrace? I don't see a backtrace anywhere.
"Hi. My phone isn't working but I'm not going to let you see what I did t
Matt wrote:
> Oh this is sad.. I'm familiar with radius.. and was hoping to be able
> to use asterisk with freeradius to be able to do call accounting and
> billing.. so you're telling me this is now not a good idea?
> Am I better off (for now) parsing the csv report each month?
>
>
> On Thu, 17 Ma
Doe anyone have Caller ID name working on incoming calls with the
following setup:
cisco 2611XM recieving calls on a PRI.
cisco talks SIP to the asterisk box.
Phones (cisco 7940s) talk SIP to asterisk.
I currently have verified the cisco router is receiving caller ID name
from the ILEC, and it a
Hey All,
I am looking for some advice on setting up a test lab for Asterisk systems
that I plan to install at client sites. Most clients will be using PRI. Does
anyone have any suggestions on equipping a lab to be able to test Asterisk
servers with T1 cards and SIP phones?
I have seen some test
set asterisk to log into database directly via there are mysql ,
postgresql and odbc drivers
available.
You dont need radius at all,
for billing and accounting all u need is a frontend to database
On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote:
> Oh this is sad.. I'm familiar with radius.. and
Hey guys. Thanks for the help on the Pattern matching, I got that
working pretty nicely.
the next problem I have is that I'm using an include file, but its not
really working...
In my extensions.conf:
[incoming]
exten => _NXXNXX,1,SetCallerID("Unknown Called Number")
#include "numbers.conf
Max Clark wrote:
Hi all,
I am working on building a new VoIP PBX. Looking at the current market
for phones it seems my best "enterprise" options are the Cisco and
Polycom phones. I have some experiance with the Cisco 7940G, but the
process of flashing the phone with the SIP firmware left a bad t
Same here - I record all my prompts either over a regular phone (Cisco
7960) or using the Plantronics DSP 500 and the "Record" application.
exten => 405,1,AGI(set-timestamp.agi)
exten => 405,2,Playback(rjs-voice-prompt-recorder) ; Play some instructions
exten => 405,3,Wait(2)
exten => 405,4,Rec
hi guys im getting this error when trying to load chan_h323 on my local
box
Mar 16 17:19:27 WARNING[2278]: libh323_linux_x86_r.so.1.12.2: cannot
open shared object file: No such file or directory
Mar 16 17:18:36 WARNING[2265]: Loading module chan_h323.so failed!
any ideas? everything compiled w
How do I get the bit like "IAX2/white_phone" in extensions.conf eg from
pre-defined variables or variants thereof ?
What I *do* get is strings like this "IAX2/[EMAIL PROTECTED]"
from ${CHANNEL}, but that's the full channel name.
What am I missing here ?
Thanks,
Thomas
___
I agree, why run to DBs. On the other hand, I have spoken with several
people asking about radius support for asterisk because they have a billing
solution that uses data from the radius servers to populate their billing
DB.
-Michael
On 3/17/05 11:00 AM, "Matthew Boehm" <[EMAIL PROTECTED]> wr
Fun things with MOH to remember...
Are the MP3 files you are using constant bitrate?
If transferred via FTP to the * machine, did you set Binary before the
transfer?
If this... (or similar)
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()
Gets you music you can hear, then the issue is m
I recorded my last set of prompts over my Plantronics DSP 500 USB
Headset. I have also used a Logitech USB Headset. These and similar are
easiest to use along with X-lite or similar softphone. I used the
suggested method of dialing an extension on the PBX and letting Asterisk
record for me direct
Daniel Burget wrote:
I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
connected via TE405P. Everything works great, except MOH. I added an
exten with MusicOnHold(30), and it plays just fine. Conferences have
music when no one is in. I have SIP phones. When I place a call on hol
Not sure the rquirements for your receptionist. I have found that the
IP600 does have most everything required to function properly. If you
do have an office without DID and a lot of traffic then you may want to
look at the tools to display status on her computer. I do have a Snom
inhouse for testi
Hi:
I had asterisk with RealTime database working perfectly in a RH 9.0
machine. I used the sip cache so I even had MWI working. The problem
is that I decided to move to Fedora Core 3. I installed the lastets cvs
version of asterisk and the RealTime addon from asterisk-addons. I at
first
I read about this option. But does it work on a h323 channel ?
(inAccessnetwork's one)
Brian C. Fertig wrote:
If you don't want to proxy the media through * the put this setting:
canreinvite=yes
this will allow the 2 end points to connect directly for the RTP
bypassing
you. otherwise I have noti
On Fri, 21 Jan 2005, Greg Boehnlein wrote:
> Hello,
> I've got a mixture of SPIP 300 and 500 phones in production for
> various clients. I've got the XML settings configured for local
> conferencing, but I'm not seeing the expected behavior from the phone when
> I attempt to conference tw
I did it and it worked. The problem was the national plan!
Thank you very much for your tip.
For those who run into this, here is the configuration of the voice port in
a Cisco AS5300 series:
interface Serial2:23
no ip address
isdn switch-type primary-ni
isdn incoming-voice modem
isdn map ad
What would be a minimum sound card/microphone combo for good voice
quality recording on a budget? This would be for * voice prompts.
Would a soundblaster live and a good mic do the job?
Chris
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I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
connected via TE405P. Everything works great, except MOH. I added an
exten with MusicOnHold(30), and it plays just fine. Conferences have
music when no one is in. I have SIP phones. When I place a call on hold,
the CLI give no in
The lack of full SIP suport and the cost of Ciscos license plus the
added base cost of their phones moved us away from Cisco and over to
Polycom. They have been working extremely well. Software updates are
free and the update process is relatively simple. I have found the
IP600 is a great desk
If you don't want to proxy the media through * the put this setting:
canreinvite=yes
this will allow the 2 end points to connect directly for the RTP
bypassing
you. otherwise I have noticed the same when I try to proxy I have to
make sure everyone is using the same codec or it doesn't work well
I think Cisco VoIP phones are absolute works of art. The first time I
saw one, I wanted them,
That being said, I use Polycom IP 500s and I absolutely love them.
The speakerphone is excellent, configs are pretty simple once you know
what you are doing with them, and the phone is very aesthetically
Oh this is sad.. I'm familiar with radius.. and was hoping to be able
to use asterisk with freeradius to be able to do call accounting and
billing.. so you're telling me this is now not a good idea?
Am I better off (for now) parsing the csv report each month?
On Thu, 17 Mar 2005 11:00:09 -0600, M
Hi,
I've got an Asterisk latest CVS head with oh323 installed. There is one
thing I can't understand about the codec negociation. I receive calls in
G723&G729, and send them to another gateway who can handle both codecs
too. So all I want to do is just passthrou, for both. It seems that *
only
Thanks for the pointer, makes sense.
Although I am not using zaptel, I am sending the calls via SIP to my Cisco
AS5300 which is connected via PRI to my provider and it happens to be set
like NI2.
I will test and will let you know.
Thanks!!!
Oswaldo
-Original Message-
From: [EMAIL PROTEC
On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote:
> I send a call with valid caller ID info (areacode+number); my provider gets
> the call and routes it properly, the end receiver gets the call and does not
> see the caller ID I sent, they just get 'Unknown Number'.
>
> This remains
Anybody knows how to patch the music on hold bug on a
bristuffed-0.2.0-RC7j 1.0.6-asterisk ?
Thanks
maxx
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Hi all,
I am working on building a new VoIP PBX. Looking at the current market
for phones it seems my best "enterprise" options are the Cisco and
Polycom phones. I have some experiance with the Cisco 7940G, but the
process of flashing the phone with the SIP firmware left a bad taste in
my mouth
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register => username:[EMAIL PROTECTED]
[netlogic]
type=friend
host=dynamic
conte
Kamran Ahmad wrote:
> i have written app for billing with asterisk. what is
> the problem in using radius.
>
> kamran
>
Its a pain and redundant. Why run two seperate databases when 1 will do
what you need? There is no native radius support for Asterisk. There is an
addon, (search the wiki) bu
Hi List
I've been using Asterisk for quite some time with no major problems, but
I've been facing this bug from the beginning and now I want to see if that
is fixable.
We have a provider who terminates our USA LD traffic and the problem comes
when relaying the caller ID I send them from my Asteri
Am Donnerstag 17 März 2005 17:49 schrieb John B Dunning:
> Daniel,
>
> Did you get much progress made on this? I'm new to asterisk - but we are
> a heavily invested LDAP shop and if I can demo an initial install that
> pulls telephony configs from LDAP it would really be nifty.
>
> I'd be happy to
Daniel,
Did you get much progress made on this?
I'm new to asterisk - but we are a heavily invested LDAP shop and
if I can demo an initial install that pulls telephony configs from LDAP
it would really be nifty.
I'd be happy to help in any way I can
- I'm not much of a developer - but have some
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended to the To: header as per requirements.
The "intercom=true" needs to be appended to the request URI, not to the
header as a whole -- your To: header should be:
To:
Mind you, I didn't get the phon
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended to the To: header as per requirements.
The "intercom=true" needs to be appended to the request URI, not to the
header as a whole -- your To: header should be:
To:
Mind you, I didn't get the phon
i have written app for billing with asterisk. what is
the problem in using radius.
kamran
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Do you Yahoo!?
Yahoo! Small Business - Try our new resources site!
http://smallbusiness.yahoo.com/resources/
I have tried everything to get BV working outbound. All worked fine
until the BV change last week. I called BV and they changed me to sip
gen with a new password. I stripped my Asterisk server to one phone on
Zap/1 until I get this working. The same BV account works fine with a
SPA-3000 so I do
Upgrade to kernel 2.6.9, there are supposed to be significant bugfixes for
CAPI support in 2.6.9.
All of my CAPI systems use FC2, 2.6.9. I tried to go 2.6.10 but had
problems.
Craig
- Original Message -
From: "Kib Eki" <[EMAIL PROTECTED]>
To:
Sent: Thursday, March 17, 2005 7:02 PM
Sub
has session border control been added to asterisk yet? i remember hearing
about it, but i haven't been able to find any information on it on wiki.
Thanks,
daniel
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Thank you John,
Max Blackmer
> >I would like to create an Intercom extension that will dial a group of
> >extensions which are connected to SIP phones. The SIP phones are setup
> >to auto answer a particular extension assigned to one of the lines in
> >the phone. All phones must answer and broa
Steve Underwood wrote:
> Matthew Boehm wrote:
>
>> Steve Underwood wrote:
>>
>>
>>
>>> The bigegst holdup has really been the poor state of T.38 support in
>>> current equipment. Few ATAs do it. Fewer do it right.
>>>
>>>
>>
>>Care to share which ones do it right? We purchased 2 ATA's all
>> wh
Gilbert Abboud wrote:
> I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk
> through SIP. Can you please send me the Dial-peer configuration that
> creates a trunk between the Cisco router and Asterisk.
You can try something like this:
dial-peer voice 900 voip
destination-
you need to get openssl-dev package too
for most dependencies problems you need respecitive dev libriaries
regards
m.
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On Thu, 17 Mar 2005 14:53:27 +0800, XinTai Wang
<[EMAIL PROTECTED]> wrote:
> who have been fabricated their own cards from Tormenta 2 PCI Card?
>
govarion.com
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hi
thanks all who helped me in making this success.
i am using latest asterisk from CVS.
asterisk-oh323-0.7.1,
pwlib-Janus_patch4-src-tar.gz,
openh323-Janus_patch4-src-tar.gz
GnuGatekeeper
it is working asterisk is routing calls to GNUGK
successfully
extensions.conf
exten => _40X,1,Dia
I've got echo problems.
*** I'm looking for paid support. ***
I'll accept free support, but don't mind paying if someone really knows
what they are doing.
I've read the wiki, etc.
Played with the settings in zapata.conf
Using V400P
PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTNECHO ON CALLED PHONE
Hello!
I have bin trying to set up trunking between some of my Asterisk
boxes but had no luck...
I use Asterisk 2.0.6 on SuSE 6.2, but I have had the same problem with
erlier releases. I have a working connection and can place
multiple calls in both direktions. Than I set "trunk=yes" on
both sid
Matthew Boehm wrote:
Steve Underwood wrote:
The bigegst holdup has really been the poor state of T.38 support in
current equipment. Few ATAs do it. Fewer do it right.
Care to share which ones do it right? We purchased 2 ATA's all which
claim to do T38 and they don't. (Azatel, WorldAxx)
-
Kamran Ahmad wrote:
> hi
>
> Any one give me any hint how to start radius with
> asterisk.
> Is there any addon available for asterisk+radius.
> Please provide me helpfull link which could help me.
> i am new to radius.
>
> regrads
> kamran
>
If you are new to radius then I will suggest and highly
Steve Underwood wrote:
> The bigegst holdup has really been the poor state of T.38 support in
> current equipment. Few ATAs do it. Fewer do it right.
Care to share which ones do it right? We purchased 2 ATA's all which
claim to do T38 and they don't. (Azatel, WorldAxx)
-Matthew
had the same thin with 729
I had to go
disallow=all
allow=g279
On Thu, 2005-03-17 at 16:37, Matt wrote:
> Hi,
> What do I need to do to get Asterisk to allow me to use codec G-726?
> I've already tried allow=all in my sip.conf config.. didn't work...
> _
Hi
I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through
SIP. Can you please send me the Dial-peer configuration that creates a trunk
between the Cisco router and Asterisk.
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
Hi,
What do I need to do to get Asterisk to allow me to use codec G-726?
I've already tried allow=all in my sip.conf config.. didn't work...
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[EMAIL PROTECTED] wrote:
I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to
be hands on for each new phone number deployed... so I would like to set
up some administrative extensions that can record greetings... lets say:
[admin]
exten => 8(NXXNXX),1,Record($1|-greeting.
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