I am a newbie with *, but I have Suse 9.3 working with Asterisk 1.0.6, Capi
and Zaptel; very easy to configure with suse rpms.
From: YT Lim [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To:
Hmmm... Folks, I beg you pardon, if I'm telling something which was said
before, but actually I have not found this anywhere, neither on
Voip-info.org or in several Asterisk's docs.
So, here is the statement:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them
will
I am wanting to front-end a legacy PBX with an asterisk box. I have done
plenty
of asterisk work over the last 6 months to PRI circuits, but not with a PBX
being involved.
I know I can use asterisk and digium cards in this manner, but do I need
separate cards for the PRI - Asterisk
I looked into the source code of Asterisk to figure out how the printf()
statements were spaced. That's the power of open source, you can look under
the hood for these questions. It's easy to find, even for non-C-Gurus. Just
do a grep for the string that you want inside of the Asterisk source
Hi
So if I have this
queues.conf
[general]
[default]
[example_queue]
music = default
strategy = rrmemory
context = queue-out ; Here we go when the caller presses a single digit,
while in the queue
timeout = 20
wrapuptime=10
announce-frequency = 30
announce-holdtime = yes
joinempty = yes
member =
Hello folks,
I've did some tests with different phones and Asterisk last two days and
here are some results, which I want to share with audience.
Cisco's 79xx and Sipura's phones/adapters on INVITE always reply with their
preferred codec.
So, for example, if Cisco's/Sipura's phone has
Thanks Flynn.
Unfortunately the files aren't written by the voicemail application. I was
hoping that there was some little command-line utility which would return
basic sound information when passed the filename.
Malcolm
-Original Message-
From: El Flynn [mailto:[EMAIL PROTECTED]
Hi List,
we've made a litle script which is called /etc/init.d/zaptel. It scans
the pci bus and creates by request a /etc/zaptel.conf and a
/etc/asterisk/zapa.conf.
Also it loads the modules automagically.
If there are volunteers who want to try this out (it'll make first setup
of an
Hi,
I'm working with this issue for a while, Now I already solve the
dialplan issues, but I still have a question about the Callgroups,
I read at www.voip-info.org that , there is a 63 limit of callgroups.
And I'm wondering why?? and if the 1.2.0beta version supported more than
63 Groups?? (I
I got the same setup,sort of
I connected a single port sangoma to my pbx
My ony problem is,when a call comes in and it gets transfered back out
that it does not detect the hangup?So that channel keeps being open
Any ideas why
On Wed, 2005-09-07 at 01:40 -0600, Rich Adamson wrote:
I am wanting
Hello!
Hmmm... Folks, I beg you pardon, if I'm telling something which was said
before, but actually I have not found this anywhere, neither on
Voip-info.org or in several Asterisk's docs.
So, here is the statement:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them
Irakli Natsvlishvili wrote:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between
them will ALWAYS go via Asterisk.
Dial plan contexts has nothing to do with how we set up RTP traffic.
I.e. Asterisk WILL NOT issue Re-INVITE even if:
1. Both UAs have canreinvite=yes in their
Hi,
I tried to use txfax to send several faxes at the
same time.
It seams, that one can't send more than 3 faxes at once,
or one risks to get 50% and more aborted faxes due to
errors.
The CPU usage is below 97%.
I tried with Opteron and IntelP4: same result.
Ok, I know, that faxing via a
On Wed, Sep 07, 2005 at 09:32:58AM +0200, Christian Richter wrote:
Hi List,
we've made a litle script which is called /etc/init.d/zaptel. It scans
the pci bus and creates by request a /etc/zaptel.conf and a
/etc/asterisk/zapata.conf.
Also it loads the modules automagically.
If there
Tzafrir Cohen wrote:
On Wed, Sep 07, 2005 at 09:32:58AM +0200, Christian Richter wrote:
Hi List,
we've made a litle script which is called /etc/init.d/zaptel. It scans
the pci bus and creates by request a /etc/zaptel.conf and a
/etc/asterisk/zapata.conf.
Also it loads the modules
YT Lim wrote:
We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's useless if we can't
make outgoing calls on our ISDN line.
We are considering abandoning FC4
Hello!
If I have more than a hundred analog telephones (analog lines) that need
to be connected to Asterisk PBX, what kind of hardware do I need, and
where can I buy it?
Thanks in advance!
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Hi,
try to search with google for channelbank or something similar.
Giorgio
Josip Gracin wrote:
Hello!
If I have more than a hundred analog telephones (analog lines) that
need to be connected to Asterisk PBX, what kind of hardware do I need,
and where can I buy it?
Thanks in advance!
Hi,
can you be a little clearer???
Every VoIP hardphone can be connected to Ethernet except for USB models.
Giorgio
Alex wrote:
Is there any VoIP phones available which can be plugged directly to
the Ethernet network?
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try google for VoIP Phone ;-)
or here: http://www.voip-info.org/tiki-index.php?page=Asterisk+phones
On Wednesday 07 September 2005 11:19, Alex wrote:
Is there any VoIP phones available which can be plugged directly to the
Ethernet network?
___
Is there any VoIP phones available which can be plugged directly to the
Ethernet network?
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What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can
Il modo migliore è quello di utilizzare AMI (Asterisk Mang. Interface)
Buon lavoro
2005/9/7, Stefano Blasco [EMAIL PROTECTED]:
Hi all,
i have a question:
what about a CTI implementation with Asterisk.
I've been looking for info in www.voip-info.org and in google, but
On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:
YT Lim wrote:
We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's useless if we can't
make
Vahan Yerkanian wrote:
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten =
Brian Capouch wrote:
I am wondering if there are any tricks getting the Cisco ATAs to do
distinctive rings via the ALERT_INFO variable?
I have seen some contradictory information in the Wiki, and I tried the
example there. I then sniffed the connection between the server and the
ATA and
Hello list,
I am trying to connect an old ISDN PBX to my asterisk system.
The setup includes an asterisk (1.0.9) running on the Soekris
hardware, with an ISDN card (Billion BIPAC PCI), and I run
zaphfc-bristuff-0.2.0-RC8k kernel module in NT mode (modes=1).
When I connect an ISDN phone to the
Hy,
I have a network with WIFI communication and VHF/ HF channels.
I have integrated asterisk in the network using SIP, ZAP and IAX2
channels for WIFI communications, but I don't Know How I could integrate
the VHF/ HF channels.
I have heard speaking about app_rpt project, but I don't Know
The local CATV company is offering internet using packet cable, and they
have asked about using Asterisk in their office. Is there any working
packet cable interface?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo
hi
i get these messages every now and then
-- PROGRESS with cause code 34 received
wtf is this?
roy
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Tzafrir Cohen wrote:
On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:
YT Lim wrote:
We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's
Done. Not sure if picked categories under SIP Mantis correct but here it
is: http://bugs.digium.com/view.php?id=5149
VY
Olle E. Johansson wrote:
File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as
1 2. You could use the dial macro. Check out the screening macro on
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
More 1. To send tones, use SendDTMF:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SendDTMF
A little more 1. I'm not sure the best way to pause for a
I'm not sure about why, but it's it is hardcoded into asterisk. Back
when it was a limit of 31, I searched around and increased the value
on my box and recompiled. It did not seem to adversely affect the
system.
On 9/7/05, René Mayorga [EMAIL PROTECTED] wrote:
Hi,
I'm working with this issue
Buenos días quiero
que ya no me llegue mas correo electrónico de la lista Asterisk, como puedo
hacerlo
Gracias
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Unsubscribe directions are at the bottom of each email.
Translation via google: Las direcciones de unsubscribe están en el
fondo de cada email.
On 9/7/05, Will Velez [EMAIL PROTECTED] wrote:
Buenos días quiero que ya no me llegue mas correo electrónico de la lista
Asterisk, como puedo
2 ways.
1) buy into the app_rpt system. They have a bespoke card for your PC
that can drive radio's etc. It's mainly aimed at repeater owners.
2) connect a phone patch between an ATA and your HF rig. This solution
is currently being used to provied phone services from a few Red Cross
Why do you care about an interface? The job of your cable modem/bridge
should be to convert from your local ethernet to their peculiar data
network.
/JFDI
Mark
Chris Mason (Lists) wrote:
The local CATV company is offering internet using packet cable, and they
have asked about using Asterisk
Being a German package this would make sense. ISDN is DT's circuit of
choice and can be found in the vast majority of businesses across Der
Fatherland.
John Daragon wrote:
YT Lim wrote:
We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to
Following on from my below email, things
have taken another bizarre twist
I have continued getting the error when
2092 tries to listen to messages by dialing .
--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack
--Playing vm-password
(language en)
WARNING:
I always get an unable to read password error if I hang up without
entering a password when prompted. Maybe is this what you are doing?
Even if you hear nothing, it is probably still expecting a password to
be entered.
On 9/7/05, Aisling [EMAIL PROTECTED] wrote:
Following on from my below
Current setup
2 x X100P cards connected to 2 analogue lines
Using prefix 7 and 8 before number
SIP gateway to SipGate to make VoIP calls
Using prefix 9 before number.
Is it possible so that if I dial a number:
0800 8000 8000
that it will try to route the call over the first analogue line, if
I hear absolutely nothing. The problem is I don't even get a chance to
enter the password. I dial and press send on my phone. Immediately
the following error appears on the asterisk console:
--Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
--Playing 'vm-password' (language 'en')
Hallo
There is the scenario:
client server
--- REGREQ with expires=60 ---
.
-- REGACK with expires=0
I did not see such situation previously, I mean PBX always responded with
expires!=0.
What does it mean? How should it be treated?
greetings
Hi,
I'm running Asterisk 1.0.7 and would like to add Speex support. I
downloaded Speex 1.0.5, installed and recompile Asterisk again.
Now trying from X-Lite to connect using Speex but getting lot of weird
erros on Asterisk console:
Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166
Hello,
I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the
reseller.
This is my first experience with Polycom and I cannot make them register in my
Asterisk Box.
I followed every advice I found (including separating [user] and [peer] in
sip.conf.
Using ethereal,
Angus - I have several mini-itx systems based on the Epia MII6000
(fanless) system.
They all run great, and I have no problems. I also run 'mpg123' with
several mp3s.
I run it in an embedded configuration (in house).
However, I do remember one board that I got where the heatsink on the
CPU
[EMAIL PROTECTED]
Simoni,
Thank you for your copersation. If you need routes in Brazil I have very
high quality ones ok...
Atenciosamente
Reduzimos ao mínimo a sua conta de Telefone
Liguetel - ITN Info - 15 anos em Telecomunicações
Diretoria Comercial - Newton Medina
PABX
Wow, first of all, if you have a hundred analog lines, you are doing
yourself a disservice.a 4 T1's would be much much cheaper, and much
easier to manage.
Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
them with FXO cards, and then buy 1 TE406 Quad T1 card for your
CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conf file coming from the realtime? It appears that RealTimefor the extensions.conf file is on a context by context basis, but you have to create each new context in the
extensions.conf file then add a switch = Realtime line
Is then possible using app_rpt solution for both VHF and HF channels?
Regards.
Mark Phillips escribió:
2 ways.
1) buy into the app_rpt system. They have a bespoke card for your PC
that can drive radio's etc. It's mainly aimed at repeater owners.
2) connect a phone patch between an ATA
Well, I just answer myself here:
Since the ISDN PBX is just the same as ISDN phone as far as the
asterisk is concerned, NT mode on the ISDN card should be used as
well.
The difference is that the phone uses p2mp (point to multi point)
protocol, as the PBX uses p2p (point to point) protocol.
The VHF or HF is determined by the radio equipment you have attached, not the
software.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy
Sent: Wednesday, September 07, 2005 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial
I use Centos 3.5 with great success. It is a RHEL3 binary compatible clone.
Don't know if I would use development cutting edge software in the
enterprise.
--- John Daragon [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:
YT Lim
Hi
I am testing a voip gateway product with Asterisk. We are experiencing
CONNECT ACK timer (T313) timing out on the Asterisk side when an incoming
call is received on the T1-PRI interface. The call is immediately routed to
voice mail.
This doesn't happen if I connect another PRI test equipment
Ive got some modifications Ive
made to asterisk that create a global switch. It essentially
just adds a check to the end of pbx_find_extension() that will try to look the
extension up in the database if its not found in one of the includes or in
any of the switches attached to the context
I fixed the problem using preg_replace but you are right, I completely
forgot We are using open source ! :) silly of me, I should have checked
that.
Thx for reopening my eyes Christoph
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Christoph
CVS HEAD/Asterisk 1.2: Is there a way to have the entire
extensions.conffile coming from the realtime?
Yes. Go read the wiki on RealTime Static.
-Matthew
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1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it
does 'ring-ri... ri ring... ri...)
and the 3rd one does not ring at all when Asterisk says 'Ringing Zap/6'.
However, when I do an 'off-hook' on this phone, I get tone signal and
can dial and talk perfectly.
I
I'm running Asterisk 1.0.7 and would like to add Speex support. I
downloaded Speex 1.0.5, installed and recompile Asterisk again.
Now trying from X-Lite to connect using Speex but getting lot of weird
erros on Asterisk console:
Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166
I am getting ready to spec out a replacement for a Merlin Legend system
with asterisk. There are a couple of things that holding me up that
hopefully someone here can answer.
1. How well do modems work through a channel back to a PRI/T1
interface?
2. Is there a decent receptionist phone (I
I don't know why Darren syas 3 Adits since each one can handle 48
FXO/FXS channels, so 2 make 96. Anyhow each Adit connects to 2 T1
ports on a TE405/6. With Adit 600 I don't see why TE406 is required
since I believe the Adit 600 will take care of the echo, I might be
wrong on this last one about
Sorry my mistake. The span to provider is pri_cpe, and the span to the
avaya is pri_net.
On 9/7/05, Rod Bacon [EMAIL PROTECTED] wrote:
It DOES help, thanks.
Except for this
the only difference between the first set of channels (1-23) and the
second set of channels (25-47) is:
It states that the conf file overrides the static db info, but what about the ael file? Does that override also?
BTW, RealTime Static...talk about oxymoron :-) Gotta love it!
Flobi
On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
CVS HEAD/Asterisk 1.2: Is there a way to have the entire
The issue appears to be between the Cisco 7940 and the
ArtDio IPF-2000, when a call is initiated between these phones the ArtDio phone
receives the audio stream fine from the Cisco, but the Cisco cant hear
anything from the ArtDio, until the Cisco user places the call on hold and then
Darren Wright wrote:
Wow, first of all, if you have a hundred analog lines, you are doing
yourself a disservice.a 4 T1's would be much much cheaper, and much
easier to manage.
Let me clear this up a little bit. There are hundreds of telephone
devices inside the building, all connected to
Indeed I do - but I read bug 2023 before posting and thought it was to
do with the system-wide problem, not with occasional occurrences. I'll
go back and read it again. Has the problem been solved with the 411P?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Hi Jorge -
I got 3 Polycom 300 phones, and upgraded to the latest firmware
provided by the reseller.
This is my first experience with Polycom and I cannot make them
register in my Asterisk Box.
I followed every advice I found (including separating [user] and
[peer] in sip.conf.
Using
Okay, this doesn't seem to be working. I've gone and deleted my ael
file also. I do know my MySQL is set up cause I have my sip, iax and
voicemail going through it too.
here's the line in extconfig.conf:
[settings]
extensions.conf = mysql,asterisk,pbx_realtime_extensions
in
Hi Robert,
Do you have the sample script for locking the extension?
Thanks,
Stephen
Robert Goodyear wrote:
On Aug 18, 2005, at 3:07 AM, Stephen wrote:
Hi All,
How can I lock the extension in Asterisk?
For example , my extension is 1000 and I am away for business trip.
I want to lock my
Olle E. Johansson wrote:
Try setting _ALERT_INFO
Worked perfectly, thanks.
B.
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Hi,
I am running Asterisk in production mode but unfortunately every few
days or so, it crashes, presumably...
Presumably because, when the phones stop working and I look for the
cause, asterisk is no longer running. Asterisk logs and
/var/log/messages contain no hints at all.
How can I
About my System:
2 * HFC Cards with misdn. 1 NT mode, 1 TE mode
1 * Sip-Provider (1und1)
On NT-Port Ritto
(Elmeg) PBX
On TE-Port NTBA
About my Problem:
When a SIP-Call from a phone connected to the Ritto PBX is
in progress and someone calls on the ISDN-Line, the greeting
I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS).
These connect to a Digium TE210P card. I'm running kernel 2.6.10
and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today.
The results are the same for all versions:
Right after I reboot, and modprobe wct4xxp, my
Okay, after noticing an error on this mysql statement after i switched to odbc:
SELECT * FROM
pbx_realtime_extensions
WHERE filename='extensions.conf' and commented=0
ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id
I added those fields and reloaded...* immediately
Nevermind, I figured out that the table is used way differently when
doing static. Here's my fixed table. I'll try to explain this in the
voip-info doc.
id cat_metric var_metric commented filename category var_name var_val
1 0 0 0 extensions.conf default exten
On Wed, September 7, 2005 18:11, Josip Gracin said:
Darren Wright wrote:
Wow, first of all, if you have a hundred analog lines, you are doing
yourself a disservice.a 4 T1's would be much much cheaper, and much
easier to manage.
Let me clear this up a little bit. There are hundreds of
You asked how to connect lines, so he answered that question. The answer
is basically the same just change the FXO in the channel bank to FXS.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005
I don't see your swich statement anywhere.
You must define a context [default] then add in the correct switch=
statement.
-Matthew
From: Flobi [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Wed, 7
The wiki doc's are correct. You are trying to combine two different methods
of pulling RealTime extensions and that is why it isn't working as you are
expecting.
Pick 1 method and all will be revealed. Both are very simple to do.
-Matthew
From: Flobi [EMAIL PROTECTED]
Reply-To: [EMAIL
Jonathan k. Creasy wrote:
You asked how to connect lines, so he answered that question. The answer
is basically the same just change the FXO in the channel bank to FXS.
Well, actually, I said: If I have more than a hundred analog telephones
(analog lines) that need... But, that doesn't help
Ohmy bad...I picked up the thread later :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to
It's not that, it's just that the wiki wasn't very clear on the fact
that all the tables for a static load had to be the same. I had
thought that I was supposed to use the table on this page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Extensions
even when doing realtime
Which version of * are you using? I had a problem with 1.0.7 crashing
unexplainably at one point, but 1.0.9 was out then and I upgraded and
it stopped.
On 9/7/05, Arik Funke [EMAIL PROTECTED] wrote:
Hi,
I am running Asterisk in production mode but unfortunately every few
days or so, it
Address lookup
canonical name asterisk.org.
aliases
addresses 216.27.40.102
Service scan
FTP - 21Error: TimedOut
SMTP - 25 Error: ConnectionRefused
HTTP - 80 Error: ConnectionRefused
POP3 - 110 Error: TimedOut
NNTP - 119 Error: TimedOut
digium.com
I'm not having any problems connecting to asterisk.org port 80.
On 9/7/05, Martin [EMAIL PROTECTED] wrote:
Address lookup
canonical name asterisk.org.
aliases
addresses 216.27.40.102
Service scan
FTP - 21Error: TimedOut
SMTP - 25 Error: ConnectionRefused
HTTP -
I don't believe 2023 has anything to do with the 411P; it was basically
an digium analog card issue (eg, TDM04b x100p).
Based on my tests and findings, the issue is the digium cards record
voicemail messages at a very low audio level (very different from
recording a voicemail from a sip phone).
On Wednesday 07 September 2005 13:47, Flobi wrote:
I'm not having any problems connecting to asterisk.org port 80.
They came up again. Finally. That check wasn't from where I am but another
location once I couldn't get onto the site. Nothing more to see here...move
on ;-0
On Wednesday 07 September 2005 14:41, Rich Adamson wrote:
I don't believe 2023 has anything to do with the 411P; it was basically
an digium analog card issue (eg, TDM04b x100p).
Based on my tests and findings, the issue is the digium cards record
voicemail messages at a very low audio level
[EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM:
I was able to raise the volume from inaudible to acceptable by
increasing the RxGain in zapata.conf by 5db. I'd rather not go the
uncomressed wav route, as it will chew up storage in my email system.
I know I'm way behind on reading this,
At 16:16 9/6/2005 -0700, Jesse Keating wrote:
On Tue, 2005-09-06 at 17:41 -0500, Doug wrote:
After I did this, it appears that the Web interface
for the phone won't change the settings, nor will
it actually reboot the phone now. What do I have
to set on the phone itself, so I can update
Hello.
Just rx'd the sip - aastra 9133i.
Haven't done sip before.
My initial attempt at setup has failed.
No Service
Anyone want to contact me off-list or on irc ?
Regards...Martin
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[EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM:
I was able to raise the volume from inaudible to acceptable by
increasing the RxGain in zapata.conf by 5db. I'd rather not go the
uncomressed wav route, as it will chew up storage in my email system.
I know I'm way behind on reading
On Wed, 2005-09-07 at 14:18 -0500, Doug wrote:
I again followed instructions here:
http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501
So yeah, the instructions are a bit misleading. I had to set register
to yes prior to the line information stuff. Without that the phone
On Wed, Sep 07, 2005 at 01:47:49PM +0200, Roy Sigurd Karlsbakk wrote:
hi
i get these messages every now and then
-- PROGRESS with cause code 34 received
wtf is this?
Nothing to see here, move along :-)
Seriously though, it's basically just and interesting message to see. The cause
I can't understand why anyone would use Fedora Core. Sure it 'can be' quite
stable depending on what your doing but it is not considered a production
ready OS.
Any of the Red Hat Enterprise edition clones such as CentOS or White Box
Enterprise Linux are a MUCH better alternative IMHO. I don't
Can anyone suggest where I might begin looking for an answer please?
I have just installed a TDM400P (2x FXS and 1x FXO modules installed)
The first problem is that it does not seem to be able to detect if the
remote party has hung up when a call comes through on the FXO. For example,
if someone
Hi All,
For sometime now I've been searching the wiki and googling, but I think I'm
missing some of the very important answers. So I'll have to ask this to the
list.
I'm trying to decide on the right motherboard and processor. Here are my
questions:
1. Would I have problems with
Hi
Can you give me any hint on with file of the source you modify that
Value???
tnx
On Wed, 2005-09-07 at 08:18 -0400, Flobi wrote:
I'm not sure about why, but it's it is hardcoded into asterisk. Back
when it was a limit of 31, I searched around and increased the value
on my box and
Does anyone currently use Vonage with Asterisk? I've tried to set
it up but it looks like Asterisk (at least the version that I have)
does not handle well the SIP call dialog, sending a BYE with the wrong
tag. As a result, when I hang up, Vonage sends back a 400 Bad
Request and the call on the
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