Hi Bilal,
Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?
Another thing: u have idea if it is working on Microsoft Windows OS? As
most
of clients here are using Microsoft and not linux.
You can take alook here:
http://www.voip-info.org/wi
honestly I only did it for testing, and probably less than 30 minutes,
so I don't know how long it would stay active long term, but it worked
great during that test... definitely longer than 30 seconds.. I am
pretty sure I'm running straight v1.0, i have been reluctant to
upgrade because its worki
> I have echo training set on in my zapata.conf file for a X101P card:
> echocancel = yes
> echocancelwhenbridged = yes
> echotraining = yes
>
> Now, I know that echo cancellation is a black art, but I am finding that
> at the beginning of a call bridged between a SIP channel and a Zap
> channel t
Dear Dan;
Thanks alot for your kindly reply.
Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?
Another thing: u have idea if it is working on Microsoft Windows OS? As most
of clients here are using Microsoft and not linux.
Regards
Bilal
I have echo training set on in my zapata.conf file for a X101P card:
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes
Now, I know that echo cancellation is a black art, but I am finding that
at the beginning of a call bridged between a SIP channel and a Zap
channel the voice quality
Hahawell the MWI is the blinking blue LCD. The message button
is "reserved for future use" Hang in there. There will soon to be some
upgrades and rumor has it that the conferencing feature will soon be
introduced so that conference button on the phone will soon be
working.
The message b
On Sat, 2005-01-15 at 15:03, Philippe Daoust wrote:
> Hello list,
>
> I want to listen to voicemails on my * box from a phone that is not
> local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm
> aware that I can forward VM to email or use a web interface but that is
> not al
Hi,
Is DIAX supported for G723 codec and can work on Windows OS?
It supports just: alaw, ulaw, gsm, ilbc and speex.
G723 is not very usual in the Asterisk world.
Best regards,
Dan
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voip technocrat wrote:
hello list ,
Hello!
:-)
my set up is like this
ip device -->ser ---> asterisk(astcc) --> pstn gatewsy
my asterisk version is 1.0.2
Latest stable package is 1.03 with 1.04 to be released very shortly :-)
iam using the ser as registration and asterisk aa the
prepaid one with t
VoIP technocrat wrote:
hello list ,
iam using a simple setup as shown below
ip device ---> ser --> asterisk (astcc) --->pstn gatewsy
:-)
Are you having some problems? Is the voice quality good/bad?
What is your network load?
What is your PC load?
What PSTN Gateway are you using?
What codecs?
Snipp
I am trying to install [EMAIL PROTECTED] on a Dell 1650
server. The setup cd runs fine and completes with no
errors but when I try to connect to the web site
(http:myip/maint) to edit the configs nothing happens
just a page not found error.
I'm not sure if there is a problem with my dual on
board
hello list ,
my set up is like this
ip device -->ser ---> asterisk(astcc) --> pstn gatewsy
my asterisk version is 1.0.2
iam using the ser as registration and asterisk aa the
prepaid one with the help of the astcc.
now my problem is the destination people
i.e the pstn line s are listenin
hello list ,
iam using a simple setup as shown below
ip device ---> ser --> asterisk (astcc) --->pstn gatewsy
Yahoo! India Matrimony: Find your life partner online
Go to: http://yahoo.shaadi.com/india-matrimony
Eric Bishop wrote:
I logged a support issue with HP and their response was that it's not
their server that is the problem and if other cards show interrupts
(which they do) there's nothing more they can do
And you told them that this is the only server the cards don't work in?
--
Cheers,
Matt R
Michael George wrote:
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
no i was using line 1 for testing /w fxs module and i never changed it
back
Also, could you show us the contents of your [routing] context in
extensions.conf?
--
Cheers,
Matt Riddell
> I want to listen to voicemails on my * box from a phone that is not
> local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm
> aware that I can forward VM to email or use a web interface but that is
> not always practical.
>
> Other than doing an IVR type arrangement or a phone
In the same context as your Voicemail() dialplan execution add:
exten => a,1,VoicemailMain() ; you can optionally pass it a vm box number in
the ()
exten => a,2,Playback(goodbye)
exten => a,3,Hangup
Now when you dial in, while your greeting is playing hit '*' and it will
prompt you for a mbox nu
On Fri, 2005-01-14 at 23:03 -0500, Philippe Daoust wrote:
> Hello list,
>
> I want to listen to voicemails on my * box from a phone that is not
> local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm
> aware that I can forward VM to email or use a web interface but that is
>
It was a very misleading error. I had the DB name spelled wrong in my
/etc/odbc.ini... You would think it would give a more intuitive error than
that.
Michael Shuler, C.E.O.
BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
682 High Poin
I am interested in learning Asterisk and have DSL (1 static IP) and a
single POTS line at home. I have an Ethernet LAN running behind a
Linksys router using NAT. My question is only about the hardware
needed at this point. The software configuration I will read about
and learn. So what hardware
As has been mentioned earlier this is to do with the DIGITMAP that is
configured in the phone1.cfg or whatever you have called it.
Mine looks like this:
[129]xxT|0[34679]xxx|[1-4]xxx|02[1-9]xxT|111|12x|195x|*8|8xx|0[58]0[08]xxT
I will explain it bit by bit so you understand i
Chris,
I do not have VoicePulse Open Access, but I do have an incoming number through
VoicePulse Connect. You might want to give the following a try unless you get
a repsonse back from someone who uses Open Access specifically. (I found the
access1.voicepulse.com address from another posting.)
Hello list,
I want to listen to voicemails on my * box from a phone that is not
local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm
aware that I can forward VM to email or use a web interface but that is
not always practical.
Other than doing an IVR type arrangement or a ph
Has any messed with getting Asterisk to work using the Voice
Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is
a number that is assigned to their hardware device (Sipura SPA-2000), the other
is a Open Access number that uses SIP from any device (you must authenticat
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
> no i was using line 1 for testing /w fxs module and i never changed it
> back
does changing it back make a difference?
> On Fri, 2005-01-14 at 07:43 -0500, Michael George wrote:
> > On Mon, Jan 17, 2005 at 08:12:24AM -0600, Justin
On Sat, 15 Jan 2005 12:41:42 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
>On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote:
>> iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
>> It is distributed as part of Steve Kann's iaxclient library.
>>
>> I've just
Andrew Kohlsmith wrote:
On January 14, 2005 11:09 am, Matthew Boehm wrote:
check out my bug post, I have yet to recieve a successful fax using rxfax.
and I'm using newest versions of everything.
That's likely your problem. :-) I don't feel like registering Yet Another
Account just to see
On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote:
> iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
> It is distributed as part of Steve Kann's iaxclient library.
>
> I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
>
> The Windows bin
It's most definately something to do with the G4 series both DL360 and
DL380. Most G3 series owners are reporting it working OK.
On Fri, 14 Jan 2005 20:50:40 +1100, Adam Goryachev
<[EMAIL PROTECTED]> wrote:
> On Fri, 2005-01-14 at 09:23 +, Steve Hanselman wrote:
> > Has anyone also logged a s
I logged a support issue with HP and their response was that it's not
their server that is the problem and if other cards show interrupts
(which they do) there's nothing more they can do
On Fri, 14 Jan 2005 16:30:25 +1000, Joshua McAdam <[EMAIL PROTECTED]> wrote:
> Has anyone logged a suppor
Hi Peter,
Basically they told me that they have several people complaining of
the problem with G4 series servers and they their hardware engineers
are going to order some of these servers and look into it. Currenly
the only "solution" they have is to use a different motherboard.
On Fri, 14 J
Okay, I've got success!
What I did was I made the change that you said, but I disabled the secret in
sip.conf. I then made the SIP changes on the DTA, and then I upgraded the
firmware to 12.34. I was then able to call the 8006 extension from another
extension, and I was able to call back that e
Is it possible to listen on more than one port within a single instance
of *? I have an engineer in Iraq that we need voice comms with, but the
gov't limits traffic to ports 80,443, 25, and 110. Can I set up IAX to
listen on port 80 AND the regular IAX port?
Or will I have to set up some weir
iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
It is distributed as part of Steve Kann's iaxclient library.
I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
The Windows binary was compiled on WinXP.
The Linux binary was compiled on RedHat 9.
Erik, thanks for your replies.
I tried both ways, and I'm still getting the same messages in the console. Do
you get these or similar in your console? Do you know what firmware they are
using by any chance?
__
Dana Olson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECT
Yes iaxcomm is an IAX softphone. I know Xten is working on improving
their linux support for their SIP based shoftphones.
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Whoops, I meant auth=plain for a packet8 dta.
Erik
On Fri, 14 Jan 2005 16:52:06 -0800, Erik Espinoza
<[EMAIL PROTECTED]> wrote:
> Under your sip.conf change to this:
>
> [8006]
> type=friend
> host=dynamic
> auth=md5
> secret=1234
> dtmfmode=rfc2833
> context=sip
> callerid=8006
> [EMAIL PROTEC
Under your sip.conf change to this:
[8006]
type=friend
host=dynamic
auth=md5
secret=1234
dtmfmode=rfc2833
context=sip
callerid=8006
[EMAIL PROTECTED]
The key is auth=md5
I have set a few of my buddies who use to have packet8 on my asterisk
box just fine.
Erik
On Fri, 14 Jan 2005 19:08:52 -0500
According to the provider's techs, Asterisk isn't properly following the SIP
guidelines about codec negotiation.
>From my understanding of the convo, Asterisk is expecting multiple codec
capabilities to be in the Capabilities string, in the first header sent
over, if it doesn't find a match, it r
Try
stratagy = leastrecent
- Original Message -
From: "dean collins" <[EMAIL PROTECTED]>
To: "Denis Voitenko" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion"
Sent: Friday, January 14, 2005 5:16 PM
Subject: RE: [Asterisk-Users] Routing incoming calls to vari
I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware version
(Application Code Version: DTA version 1.0 US (8x8 00)) onto it via TFTP,
so I could access the SIP configuration.
Under the SIP config, I put the IP of my * system, the 5060 port, and for
Domain Name, I put def
On Sat, 2005-01-15 at 07:09, Adam Fineberg wrote:
> Howard Lowndes wrote:
>
> >Can anyone _recommend_ a downloadable OSS softphone that _works_ under
> >Linux and is compatible with Asterisk.
> >
> >So far I have tried kphone and linphone and had problems with both, and
> >I am still waiting to he
Do a search on ACD and agents, this is certainly achievable.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis
Voitenko
Sent: Friday, January 14, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Routing incomin
Hello List!
Several users of a german VoIP-Forum experience the following similar
problems when using CAPI with an active ISDN-Card AVM B1 PCI connected
to the PSTN and a HFC-S in NT-Mode used as an "internal" S0-Bus to
connect ISDN-Phones.
The problem is, that when making a phone call from an
I encountered the same problem today. 'lspci -nvv' showed that the
subsystem ID of the TE110p changed from 79fe to 79fa or 797e.
Powering off/on the machine restored the subsystem ID to 79fe and the
wcte11xp module could then load. I already emailed digium support for
this.
On Mon, 20 Dec 2004
Paul Rodan wrote:
Capabilities: us - 0x4(ULAW), peer - audio=0x100(G729A)/video=0x0(EMPTY),
combined - 0x0(EMPTY)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp: No compatible
codecs!
What throws me off i
I am setting up * to accept incoming calls and route them to our reps.
What I'd like to do route the call to the rep who has been idle the
most, thus distributing the load among the reps. I can't seem to find
this functionality. Can someone point me in the right direction?
Script Head
Ok,
My provider is sending a call to me via ULaw but Asterisk
isn’t picking up on this, I’ve only allowed ulaw, I disallow=all
and then allow=ulaw in my sip.conf and that’s the only thing I allow, but
when my provider sends me the requests, I get an error about No Compatible
Codecs:
> Is the reregistering normal behaviour for an external client ?
Yes, IAX default behavior is to register every minutes or so, external
or internal
If I'm wrong, please someone correct me
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h
Hello!
I am trying to set up multi-link PPP using two T100P cards in one
machine, and 1 T405P card (the 4-port one) in another machine. I have
previously been able to get PPP working between the two T100P cards
in separate machines
The 4-port card seems to be my problem currently. I am trying
Dear Dan;
Thanks for your kindly email and reply.
Is DIAX supported for G723 codec and can work on Windows OS?
Regards
Bilal
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Dear Sebastian;
Thanks a lot for your kindly advise to use ASTCC.
But can u advise me the link for ASTCC to download it and wether it is open
source (to download the source and work on it?
Regards
Bilal
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Hello,
My group at work has a test voip deployment, using a Cisco 2611XM with
FXO and FXS modules, and a small number of Cisco 7940 and 7960 ip
phones. This is working today, with outgoing calls going over a pair of
POTS lines on the 2611XM.
I would like to add an asterisk server to the mix, just
Look for "canreinvite=yes" to get Asterisk out of the RTP path. Since
SIP traffic is infrequent and low volume having Asterisk in that loop
shouldn't be a problem, it's the RTP traffic you really want going
point to point. Realize that Asterisk can't get out of the loop if you
use the t or T option
The message button can be programmed to dial an extension that checks voicemail
exten => 160,1,Voicemailmain(${CALLERIDNUM})
On Fri, 14 Jan 2005 18:57:41 +0100, Aldo Bergamini <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] is believed to have said:
>
> >I don't mean to be rude to everyone who re
On Fri, 2005-01-14 at 14:38 -0600, Matthew Boehm wrote:
> Time for you to learn some maners and not to bitch at/insult people for
> something they don't understand. Time for you to stop telling people what to
> do. Please add those 2 to you ToDo list.
Before your whining gets too out of hand, you
Matthew Boehm wrote:
Eric,
Thank you for explaining this to me instead of being rude and bitching at
me about my lack of GPL understanding.
You caught me in an unusally good mood, that's all.
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On Fri, 14 Jan 2005, Alejandro G wrote:
> Sorry for the delay. I had to reconfigure all again. I do I inbound call to
> asterisk and the result log is this (hope this is usefull):
> < Protocol Discriminator: Q.931 (8) len=47
> < Call Ref: len= 2 (reference 1280/0x500) (Originator)
> < Message ty
Title: Re: T100P with NEC C2400 IPX switch
Jerry,
I have made this work on a 2400 IPX and * 1.0 with the T100P setup as a
PRI. The PRI has to be programmed so that the IPX is set as the CPE
side and * as the NETWORK side.
--LJ
"Jerry Geis" <[EMAIL PROTECTED]> wrote in message
news:<[E
Eric,
Thank you for explaining this to me instead of being rude and bitching at
me about my lack of GPL understanding.
Sincerely,
Matthew
- Original Message -
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Frid
Time for you to learn some maners and not to bitch at/insult people for
something they don't understand. Time for you to stop telling people what to
do. Please add those 2 to you ToDo list.
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing Lis
Matthew Boehm wrote:
So are you telling me that you cannot use other commercial products in
conjunction with asterisk?
You cannot distribute a closed source add-on (except AGI) for Asterisk
without a commercial license for Asterisk. This is just standard GPL
stuff, not Asterisk sprcific.
___
Sirs,
I have the following situration:
1) AudioCodes Stretto 2000 media gateway running MGCP
2) E1 Digium card at a PC with Asterisk
3) My application running as Call Agent (CA) from Stretto 2000
| My app |--| Stretto 2000 |--| E1 card + Asterisk |
As my application is the
trying to set up and configure a polycom soundpoint ip 500 phone, when trying
to get it to register with sip, i get the following message
Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=
On Fri, 2005-01-14 at 14:09 -0600, Matthew Boehm wrote:
> So are you telling me that you cannot use other commercial products in
> conjunction with asterisk?
Time for you to go learn about the GPL. Time to go learn about proper
trimming of an email. Time to learn how to use the archives for
inform
trying to set up and configure a polycom soundpoint ip 500 phone, when trying
to get it to register with sip, i get the following message
Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=
Howard Lowndes wrote:
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.
How about iaxcomm?
http://iaxcli
Use valetparking :P
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kyle Hagan
> Sent: Friday, January 14, 2005 1:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Call Parking
>
> I
On Fri, 2005-01-14 at 14:13 -0500, Jeremy Hinton wrote:
> Maybe i missed this somewhere, but is it possible to define a variable
> with a scope of the current context? I know i can define a system wide
> variable, and i can define one that is valid for the duration of the
> channel, but is
So are you telling me that you cannot use other commercial products in
conjunction with asterisk?
Matthew
- Original Message -
From: "izo" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, January 14, 2005 1:32 PM
Subject: Re: [Asterisk-Users
On Fri, 2005-01-14 at 16:27 -0200, Denis GalvÃo - iSolve wrote:
> Em Sex 14 Jan 2005 16:11, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling VoceMail,
> EchoTest, etc. The problem is related with the conversation between two
> DIAX Softphones.
With * in the middle
On Sat, 2005-01-15 at 05:37 +1100, Howard Lowndes wrote:
> Can anyone _recommend_ a downloadable OSS softphone that _works_ under
> Linux and is compatible with Asterisk.
>
> So far I have tried kphone and linphone and had problems with both, and
> I am still waiting to hear back from the X-Lite
Brain:
I am still hanging with the same problem, although i tried this:
# iptables -t nat -A PREROUTING -p udp -i eth0 --dport 0 -j REDIRECT
--to-port 5060
from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20grandstream%20budgetone#comments
But still have same problems?
Any how
Has anybody compiled app_conference as of late?
I've already asked on the app_conference devel list but as I'm rather in
a hurry my thinking is somebody here has both run into and found a way
to get this compiled and running.
Using stable asterisk and the most recent app_conference from it's cvs
I tried IaxComm in two Linux boxes. Everything work fine, with USB Phones +
IaxComm.
So, the problem should be related to Windows OS!?
Wich version of Windows are you using Dan!?
Denis.
Em Sex 14 Jan 2005 17:16, Denis Galvão - iSolve escreveu:
> Same problem with jitterbuffer=no
>
> I tried I
Howard Lowndes wrote:
I have actually got a bit more cunning that this by using sipgetheader()
and sipaddheader().
The default user name is "asterisk", hard coded in chan_sip.c, so what I
did was to use sipgetheader() to get the From: header, then I cut() it
at the ":" character and the "@" charact
I am using a manager app to do redirects, when I redirect to 700, for
parking, the person on the other end hears the number its parked on.
How do I stop this?
Kyle
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Try with ASTCC is free.
Sebastian
-Mensaje original-
De: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Enviado el: Martes, 14 de Enero de 2003 14:56
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] PrePaid Applications
Hi;
Is the Prepaid Applications that we can use it with A
On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
> Why does it have to be commercially licenced?
Without it, the SS7 software would be linking to GPL software which
means they would have
to GPL the code too. So the only way to get commercial SS7 is to have
it with commercial
asterisk.
Maybe i missed this somewhere, but is it possible to define a variable
with a scope of the current context? I know i can define a system wide
variable, and i can define one that is valid for the duration of the
channel, but is it possible to define a variable that comes into scope
for every ch
Hi,
- Original Message -
From: "Bilal Ghayad" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 14, 2003 8:58 PM
Subject: [Asterisk-Users] PC to Phone
Can some one advise me an PC to Phone client software to be used under
Windows OS at the client side, to be communicated with Asterisk PBX?
Same problem with jitterbuffer=no
I tried IaxComm, same problem of DIAX.
This is related with iaxclient...
Denis.
Em Sex 14 Jan 2005 17:03, Dan escreveu:
> Hi,
>
> \> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> >> > I dont have problems when calling PSTN extensions, and calling
> >> > VoceMail, E
Weird, I haven't actually tried that, that may be part of my problem too.
If i disconnect the line from the NT1 bristuff will not reconnect on every
occasion. Disconnecting the cord between the Nt1 and the HFC-S card makes
it lose connectivity occasionally. But I guess either way, the modules
sh
I don't have any card specific modules loaded. I don't have chan_zap loaded.
So how can the zaptel kernel module show usage? I just rebooted this server
due to defunct asterisk process.
How can I find out what 4 things are using zaptel?
[EMAIL PROTECTED] localhost]# lsmod
Module
Peter,
Sorry for the delay. I had to reconfigure all again. I do I inbound call to
asterisk and the result log is this (hope this is usefull):
Alejandro
Enabled EXTENSIVE debugging on span 1
*CLI> T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (27)
> [ 00
> Yes, it is called a Lucent MAX TNT or Cisco AS 5400 with DS-3 input
Perhaps you meant 5300? We have one of those and yes, it works fine. But
it costs about $20K for one.
> If you really want RJ-45 connections for your PRIs (ick) then you can
What else would you use to make a PRI cable
I have no idea how ODBC converts a prepared statement to MySQL when only up
until 4.1 did mysql support them. Have you tried using res_config_mysql
inside asterisk-addons?
-Matthew
- Original Message -
From: "Michael Shuler" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Com
Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
> VoceMail, EchoTest, etc. The problem is related with the conversation
> between two DIAX Softphones.
Between 2 DIAX phone and the delay is in one direction only??
Yes. One direction onl
Em Sex 14 Jan 2005 16:43, Dan escreveu:
> > I dont have problems when calling PSTN extensions, and calling
> > VoceMail, EchoTest, etc. The problem is related with the conversation
> > between two DIAX Softphones.
>
> Between 2 DIAX phone and the delay is in one direction only??
Yes. One direction
Dhennys Pestana wrote:
I'm trying to find a way to connect two (or more) extensions directly without
being kept in the middle during the conversation but it won't happen.
Asterisk will always stay in the SIP signaling path. It can get out of
the RTP path (only way to really see this is using some
Hi;
Is the Prepaid Applications that we can use it with Asterisk are Free or we
have to pay?
Regards
Bilal
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Hi;
Can some one advise me an PC to Phone client software to be used under
Windows OS at the client side, to be communicated with Asterisk PBX?
Regards
Bilal
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Hi,
I've never used those instructions, this is my skinny.conf, and I was
able to connect 3 12 sp+ (apparently the exact same as a VIP30 minus
the extra buttons) with different firmwares.
;
; Skinny Configuration for Asterisk
;
[general]
port = 2000 ; Port to bind to, default tcp/2000
I have modified the CallMe feature for DIAX to provide an Echo test.
Just use it with 0.9.9g and see the result. To pass the explanation or to
end the echo test just press '#'. You can still leave me a message after
that.
I got the echo test. The result was fine, just a very SHORT delay, but
noth
Thank you everyone for your help. It looks like my problem was
related to:
http://bugs.digium.com/bug_view_page.php?bug_id=0003332
I patched and all is well now.
-brian
On Fri, 14 Jan 2005 at 12:57 Brian S. Adelson ([EMAIL PROTECTED]) wrote:
> Muhammad,
> Could you possible share you confi
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.
--
Howard.
LANNet Computing Associates;
Your Linux peo
I'm currently trying to use a Radius server for acct and auth, cause
much of our systems are using it.
Anyone has an asterisk server working with Radius Auth and Acct?
Tkx, LTenorio
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I am looking at interface the T100P with a NEC C2400 IPX.
Has this been done before by anyone???
quick search did not bring anything up.
Thanks,
Jerry
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Em Sex 14 Jan 2005 16:11, Dan escreveu:
> I have modified the CallMe feature for DIAX to provide an Echo test.
> Just use it with 0.9.9g and see the result. To pass the explanation or to
> end the echo test just press '#'. You can still leave me a message after
> that.
I got the echo test. The re
DIAX and iaxComm both use the iaxclient library, which has been in flux
lately:
lots of new features added. If your iaxComm is version 0.99pre4 or later,
then
I *think* it might be using a more bleeding-edge version of the iaxclient
library.
Some of the recent library work has been on the jitte
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