Hi,I would like to know from where to download the patches?
Regards.
john
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On Wednesday 26 March 2003 16:59, Pavel Litvinenko wrote:
> I have same needs,
> there is good phone dlink dph-100m but * has no transfer in mgcp channel
are you using that phone? any problems? what software version?
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel:
To cut this:
- This is not a problem with chan_h323. This is a problem with pwlib.
- Sorry, Jermy.
On Wednesday 26 March 2003 17:37, Jeremy McNamara wrote:
> Brian Johnson wrote:
> >There seems to be a lot of confusion regarding the h323 channels
> >
> >It might not be possible but ..
> >is it p
Jeremy McNamara wrote:
Michael Manousos wrote:
Your resources were actually some feedback and run-time results
of the software. We get lots of these and help us make it better.
I beg to differ.. Before I got pissed off and hacked my own H.323
driver, I sent at least 6 patches and many many many
> >> Your resources were actually some feedback and run-time results
> >> of the software. We get lots of these and help us make it better.
> >
> >
> > I beg to differ.. Before I got pissed off and hacked my own H.323
> > driver, I sent at least 6 patches and many many many email questions to
>
> S
Hi,
>From what I have been able to work out * supports G.711 a+u, GSM
and LPC-10 for VoIP calls.
So far it seems that the Hardphones out there support G.711, G.729
and some times a few other codecs..
So the common denominator seems to be G.711, the problem with this
codec is that it requires app
Hi,
i´m still trying to debug my issue with audio only going one way in the
following configuration:
ATA186-SIP-Asterisk-isdnBRI-Asterisk-Voicemail
(Asterisk is the same machine, i call myself over ISDN)
Calling this chain i cant hear the prompts from the voicemail application.
Back to "WriteFo
On Thursday 27 March 2003 14:11, Christopher Arnold wrote:
> ATA186-SIP-Asterisk-isdnBRI-Asterisk-Voicemail
isdn4linux or capi?
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356
Computers are like air conditioners.
They stop working when you open Wind
- Original Message -
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Hemant Kumar"
<[EMAIL PROTECTED]>
Sent: Thursday, March 27, 2003 6:18 PM
Subject: Re: [Openh323gk-users] SIP Support?
we don't have any large number.
others may
ask the asterisk list at [EMAIL
On Thu, 27 Mar 2003, Roy Sigurd Karlsbakk wrote:
> On Thursday 27 March 2003 14:11, Christopher Arnold wrote:
> > ATA186-SIP-Asterisk-isdnBRI-Asterisk-Voicemail
> isdn4linux or capi?
isdn4linux since i unfortunatly is running isdn cards without capi
support. (Dynalink IS64PH ISDN Adapter, from a
On Thursday 27 March 2003 14:59, Christopher Arnold wrote:
> On Thu, 27 Mar 2003, Roy Sigurd Karlsbakk wrote:
> > On Thursday 27 March 2003 14:11, Christopher Arnold wrote:
> > > ATA186-SIP-Asterisk-isdnBRI-Asterisk-Voicemail
> >
> > isdn4linux or capi?
>
> isdn4linux since i unfortunatly is runnin
On Thu, 2003-03-27 at 06:22, WipeOut . wrote:
> Hi,
>
> >From what I have been able to work out * supports G.711 a+u, GSM
> and LPC-10 for VoIP calls.
>
> So far it seems that the Hardphones out there support G.711, G.729
> and some times a few other codecs..
>
> So the common denominator seems
What about G.723.1? Anybody have any experiece with this? I know I can order the
documentation (including C source) from the ITU, what does this entitle me to do? Are
there any licencing gotcha's to going with this approach?
Lenny
-Original Message-
From: Steven Critchfield [mailto:
On Thu, 2003-03-27 at 08:56, Lenny Post wrote:
> What about G.723.1? Anybody have any experiece with this? I know I
> can order the documentation (including C source) from the ITU, what
> does this entitle me to do? Are there any licencing gotcha's to
> going with this approach?
>
Check the a
Hello,
I have a very complicated system which contains 2 *'s.
PSTN --- CISCO --- H.323 --- 1 Asterisk --- IAX --- 2 Asterisk --- PSTN
In the middle of conversation i get a Hangup. I get this logs when it
disconnect on the 2 Asterisk.
I must metion that the call is originated by Cisco, and i can'
They are merged with CVS now.
Mark
On Thu, 27 Mar 2003, it wrote:
> Hi,I would like to know from where to download the patches?
>
> Regards.
>
> john
>
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo
Hi.
I'm experiencing a little issue with chan_zap.
Seems that is slow to detect the off-hook state
(answered), so if you answer between the first
& second ring , when CID is transmitted, you
can hear the CID tones ;-)
Could be a slow cpu problem?
My system is a PIII 500Mhz, kernel
2.4.19+preempt
Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link.
There is somebody he know (has experienced) how many concurrent call (Classical
Phone->Voip) can handle Asterisk ?
Thanks !
--
__
http://www.linuxmail.org/
Now with e-mail for
My asterisk system is now working wonderfully (thanks to all of you for
your invaluable contribution to the software world and your assistance
on this list)!
Enough of that... On to my current issue. MeetMe is working just fine in
a basic sense, but when I try to assign a PIN to a conference room,
How would I go about implementing a voicemail callback
feature. The idea is that asterisk contacts any extensions that have voicemails
waiting, and plays them back. There would have to be parameters such as number
of callback retries, time between retries.
Thanks
Tan
On Thu, 2003-03-27 at 11:16, Ciocoiu Catalin wrote:
> Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link.
> There is somebody he know (has experienced) how many concurrent call (Classical
> Phone->Voip) can handle Asterisk ?
Which codec do you think you will be using?
--
On 27 Mar 2003, Matthew Farley wrote:
> My asterisk system is now working wonderfully (thanks to all of you for
> your invaluable contribution to the software world and your assistance
> on this list)!
>
> Enough of that... On to my current issue. MeetMe is working just fine in
> a basic sense
Lets say G.711. So what is maximum number calls between PRI-H323 ???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 27, 2003 9:27 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoIP Gateway Performance
On
This the error I receive when I try to record a GSM file
-- Executing Record("SIP/67.98.37.220:5060",
"intro|gsm") in new stack
-- Playing 'beep'
WARNING[15374]: File file.c, Line 602 (ast_writefile): No
such format ''
WARNING[15374]: File app_record.c, Line 143 (record_ex
I have my snom200 registering only if I set the passwords to blank in both *
and the snom...
sip.conf
secret=
SNOM/Home/Settings/SIP/Authentication
Realm=
username=1114
password=
...I'm wondering if there is a setting in the snom that requires an
encripted password?
Thanks,
John
This e-mail
Has anyone gotten the Packet8.net VoIP service working with
Asterisk? I'm behind a NAT firewall, but I'm curious to find
out if anyone has gotten it working at all (NAT or no NAT).
--Eric
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> So it looks like the best codec is the GSM codec as far and badwidth
> vs voice quality, but I can't seem to find which hard phones support
> the GSM codec or if * supports the G.729 codecs or others..
>
> Which phones do the * user commumity find work the best?? and which
> codecs do you use??
> Back to "WriteFormat and ReadFormat" output from show channel:
>
> Im trying to figure out what goes wrong in the above scenario and i guess
> show channel on each channel in the chain would be one way to do this.
>
> But im a bit confused regarding "WriteFormat and ReadFormat" are these for
> bo
I'm getting occasional seg faults when a call ends. Here is the
backtrace. Calls are going to a SIP device.
(gdb) bt
#0 0x08057bae in ast_queue_frame (chan=0x8125240, fin=0xbf5fea0c, lock=1)
at channel.c:344
#1 0x08057dbf in ast_queue_hangup (chan=0x8125240, lock=1) at channel.c:380
#2 0
You can try to do that using authenticate application
regards
Martin
On Thu, 27 Mar 2003, James Golovich wrote:
>
>
>
> On 27 Mar 2003, Matthew Farley wrote:
>
> > My asterisk system is now working wonderfully (thanks to all of you for
> > your invaluable contribution to the software world and y
You have to call record like this
exten => 8000,1,Record,intro:gsm
(read show application record)
Martin
On Thu, 27 Mar 2003, Michael K. Rodriguez wrote:
> This the error I receive when I try to record a GSM file
>
>
>
>
>
> -- Executing Record("SIP/67.98.37.220:5060", "intro|gsm") in new
Do also "frame 0"
Martin
On Thu, 27 Mar 2003, Eric Wieling wrote:
> I'm getting occasional seg faults when a call ends. Here is the
> backtrace. Calls are going to a SIP device.
>
> (gdb) bt
> #0 0x08057bae in ast_queue_frame (chan=0x8125240, fin=0xbf5fea0c, lock=1)
> at channel.c:344
> #
You can use "debug channel " on asterisk console
to find out which channel sends disconnect/hangup
You have to trace all the channels on both of your systems.
regards
Martin
On Thu, 27 Mar 2003, diana wrote:
> Hello,
>
> I have a very complicated system which contains 2 *'s.
>
> PSTN --- CI
--On Monday, March 24, 2003 4:14 PM + Michael Bielicki
<[EMAIL PROTECTED]> wrote:
wait 2 weeks and you will be oable to order one from digium :)
Will to ports on this card be able to act as FXO as well, or just as FXS?
If the answer is yes, can we control which ports do which in any
combina
On Thu, 2003-03-27 at 11:55, George Lin wrote:
> Lets say G.711. So what is maximum number calls between PRI-H323 ???
G.711 shouldn't have any cost in cpu time for the codec as it is 8 bit
8000hz just like the PRI line. You would maybe have a little overhead in
the H323 channel driver and the netw
Quick question what happens if you go over
your channel licenses?
Mark Spencer wrote:
So it looks like the best codec is the GSM codec as far and badwidth
vs voice quality, but I can't seem to find which hard phones support
the GSM codec or if * supports the G.729 codecs or others..
Which phones
> My extension definition for the conference room looks like this:
>
> exten => 8600,1,Wait,1
> exten => 8600,2,Playback(wstconfbeta)
> exten => 8600,3,Meetmecount,8600
> exten => 8600,4,Meetme,8600|p|1234
PIN is not yet implemented apparently. You could in the mean-time use
"Authenticate" like t
Sunday Folayan wrote:
Its a pain not having H323 support out of the codes check out of CVS.
Ok... Lets do this diplomaticly...let the community decide.
Who thinks chan_h323 is ready to be distrubuted with Asterisk?
Jeremy McNamara
___
Asteris
> Will to ports on this card be able to act as FXO as well, or just as FXS?
Maybe later.
But there was some posting about "FXS to FXO converter" a few weeks before
???
> If the answer is yes, can we control which ports do which in any
> combination?
Why not ?
> Finally, can this card coexist
Shouldn't be. Can you send me a dump?
Mark
On Thu, 27 Mar 2003, John Harragin wrote:
> I have my snom200 registering only if I set the passwords to blank in both *
> and the snom...
>
> sip.conf
> secret=
>
> SNOM/Home/Settings/SIP/Authentication
> Realm=
> username=1114
> password=
>
> ...I'm
The same as you go over the number of PRI channels ?
regards
Martin
On Thu, 27 Mar 2003, James O. Sizemore III wrote:
> Quick question what happens if you go over
> your channel licenses?
>
> Mark Spencer wrote:
>
> >>So it looks like the best codec is the GSM codec as far and badwidth
> >>vs vo
Is there any way to require a caller to enter their "customer
number" when they call in AND have the info put in the CDR info?
Also is there a way do to the same for outbound calls?
--Eric
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For $10 a pop, I would buy 24 to cover upto a T1 (and I've already got my call into
Greg). This is exactly what I'm looking for.
I suspect nothing bad would happen if you went over, except for that pang of guilt you
may feel until you ante up for it's use :-)
I'm personally more interested in
as I understand it there are 2 h323 modules in the works (as well as others
for unrelated stuff.) It would be better for everyone if these were all in
one place (main cvs repository). The default modules.conf could simply be
not to load them, and notes should be provided as to their current stat
At 02:43 PM 3/27/2003 -0600, you wrote:
> Will to ports on this card be able to act as FXO as well, or just as FXS?
Maybe later.
But there was some posting about "FXS to FXO converter" a few weeks before
???
That is a separate device that works with any FXS port from my
understanding of it. F
Hello,
I'm totally new to telephony but I'm now in charge of doing some
cleaning in our school telephony architecture. I'm used to networking
stuff and was very happy to find Asterisk, a free application, into
these proprietary hardwares and softwares.
I cannot remove our "old" PBX (a Bosch
Hrm, interesting, is it possible in asterisk to authenticate VoIP users
based on called-from number, if so, could someone post sample syntax
(especially if I can auth from a list, mysql/flatfile it doesn't matter,
for a universal extension, this is outbound only).
Regards
MIKE
On Thu, 2003-03-27
how would that work with * since * sees them qs fxs ports ?
On Thursday 27 Mar 2003 20:43, Martin Pycko shaped the electrons to say:
> > Will to ports on this card be able to act as FXO as well, or just as FXS?
>
> Maybe later.
> But there was some posting about "FXS to FXO converter" a few weeks
That's my question exactly... How many concurrent calls can I run over
G.729 before I have to go out and buy a bigger processor? Does anyone
have some data? I've heard rumors on IRC, but I'd rather have some
"real world" data...
(Maybe I'll have to try it myself! Mark, is it possible to get the
On Thu, 2003-03-27 at 15:07, Jon Pounder wrote:
> as I understand it there are 2 h323 modules in the works (as well as others
> for unrelated stuff.) It would be better for everyone if these were all in
> one place (main cvs repository). The default modules.conf could simply be
> not to load the
If anybody buys this FXS to FXO converter and gets it to work with Asterisk,
please let us know how it goes. I may try it one of these days myself.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jon Pounder
Sent: Thursday, March 27, 2003 2:10 PM
To: [EMAIL P
On Thu, 2003-03-27 at 15:42, Michael Bielicki wrote:
> how would that work with * since * sees them qs fxs ports ?
You define the ports as immediate or not. Then the line to asterisk is
like a hotline phone, and it prompts for what is to be done. The device
just deals with the signaling conversion
I am no expert on it but I would assume when it sees the line side ring, it
answers, and goes off hook on the extension side.
probably little more than 600ohm transformer with a relay on each side, and
a ring detector on the line side that controls both relays, maybe some sort
of disconnect con
I am using a 7960 and it is registered to the *server, but I
keep getting this error. Does anyone know why?
NOTICE[5126]: File chan_sip.c, Line 3080 (handle_request):
Registration from 'sip:[EMAIL PROTECTED]' failed for '67.98.37.220'
Michael
K. Rodriguez
DialMex LLC
N
I don't follow - if the module is in cvs, what is the harm if it builds
from the main makefile ?
So it takes a few extra seconds to build - so what ?
As long as it is not loaded it is doing no harm just sitting there.
Maybe then a separate make target like "make experimental" to cover this
"unof
What are the dtmf= options in sip.conf.
--Eric
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I like the reasoning. In fact it could apply accross the
board: Have every channel type disabled unless specifically
enabled. I would assume most Asterisk sites don't use most
channel types.
No, you would not have to answer lots of "nothing works"
e-mail because the makefile could have a line
> Quick question what happens if you go over
> your channel licenses?
It cannot transcode.
Mark
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Using DISA.
Mark
On Thu, 27 Mar 2003, Eric Wieling wrote:
> Is there any way to require a caller to enter their "customer
> number" when they call in AND have the info put in the CDR info?
>
> Also is there a way do to the same for outbound calls?
>
> --Eric
> ___
> as I understand it there are 2 h323 modules in the works (as well as others
> for unrelated stuff.) It would be better for everyone if these were all in
> one place (main cvs repository). The default modules.conf could simply be
> not to load them, and notes should be provided as to their current
Hello list,
First, * kicks butt in about every way.
Next, I have a little problem I was hoping you might be able to help me
with. Some of you may recognize me from the IRC channel. This is my
first post, so I apologize in advance for its length.
I've acquired a T400P (Quad T1 card) and two Zho
We've done 60 channels on a dual 1.8 Ghz Xeon. Trial channels are *not*
available because we have to purchase keys from Voiceage, and they are
unwilling to make any trial keys available.
Mark
On 27 Mar 2003, Jared Smith wrote:
> That's my question exactly... How many concurrent calls can I run
Hello list,
First, * kicks butt in about every way. If you recieve this twice,
sorry about that. I accidently sent it from the wrong address the first
time.
Next, I have a little problem I was hoping you might be able to help me
with. Some of you may recognize me from the IRC channel. This is
The phone still appears to be "locked" as we get a similar error on our
locked Vista 350's. The unlocked 390's we have do not get this error
message.
You mentioned that you have an "unlock" procedure. Would you mind passing
this on? I could try it on my 350 to see if it works.
> I connected th
Is it possible to send a call from the asterisk server to a
gateway via sipv2 protocol.
I have some 7960 phones that can receive a call from a
5350 via sipv2 and the phone can send to the gateway via sipv2.
Is there an exten that dials to a gateways ?
Michael
K. Rodriguez
>I connected the phone and set up an extension for the GetCPEID app.
I have a 480 & it is completely *UNLOCKED* why dont you post that getCPEID
data & lets see if we have the same version Info
>The example configuration as well as the representative at Sayson (a 3rd
>party service that will progr
On Thu, Mar 27, 2003 at 07:14:22PM -0500, Wade Weppler wrote:
> The phone still appears to be "locked" as we get a similar error on our
> locked Vista 350's. The unlocked 390's we have do not get this error
> message.
Strange as it sounds, I've been able to get a third party programming
service to
On Thu, 27 Mar 2003, Mark Spencer wrote:
> > Back to "WriteFormat and ReadFormat" output from show channel:
> >
> > Im trying to figure out what goes wrong in the above scenario and i guess
> > show channel on each channel in the chain would be one way to do this.
> >
> > But im a bit confused r
You sent it to the wrong address bud.
Try
[EMAIL PROTECTED]
- Original Message -
From:
Percy Kwong
To: [EMAIL PROTECTED]
Sent: Friday, March 28, 2003 12:57
AM
Subject: [Asterisk-Users]
unsubscribe
Well, I almost cried... Here's why:
I followed the unlock procedure on my unlocked 390, and it worked as
outlined, so I did the big test; I tried to unlock my 350 that I've been
trying to unlock for months.
There isn't a Mute button, so I hit the Flash button instead. Voila, it
said "Memory eras
Thank you Michael. I've turned inbandDTMF to "no" and now is working.
Regards,
Carlos.
Carlos Crembil
Servicios Profesionales
http://openware.biz
eMail: [EMAIL PROTECTED]
> I have gotten relatively far with support from Sayson and Aastra, but
> the vibe I'm getting at this point is that "ADSI is a standard, we
> implement it but we're not responsible for helping you develop your
> implementation". The specs are available from Telcordia (and perhaps
> belcor?) for a
Does your link button work, after you program it with adsiprog? It broke on
my 350 .. had to clear the asterisk load out again to use it. Link/flash
is sorta important to an asterisk phone...
At 08:21 PM 3/27/2003 -0500, you wrote:
Well, I almost cried... Here's why:
I followed the unlock proc
The separator on app_record was ':' instead of '|'. I've modified it to
accept either one. Please try a cvs update.
Mark
On Thu, 27 Mar 2003, Michael K. Rodriguez wrote:
> This the error I receive when I try to record a GSM file
>
>
>
>
>
> -- Executing Record("SIP/67.98.37.220:5060", "int
> Will to ports on this card be able to act as FXO as well, or just as FXS?
> If the answer is yes, can we control which ports do which in any
> combination? Finally, can this card coexist with the X100P FXO card in the
> same PC and will Asterisk support them all at the same time?
They are FXS s
I've had good luck with chan_h323, but the dependancies are a PITA :(
I'd say as soon as there is some rational way to figure out if you've
got the required versions of pwlib & openh323 go for it.
Nick
On Thu, Mar 27, 2003 at 03:33:56PM -0500, Jeremy McNamara wrote:
> Sunday Folayan wrote:
Maybe I'm doing it wrong, but does the directory app in CVS work?
I can set up an extension and dial it. It gives the directory prompt
but never does anything when I dial three digits for the name. Is there
something I should know about setting it up (I did put in my user
extensions context as a
>Well, I almost cried... Here's why:
>
>I followed the unlock procedure on my unlocked 390, and it worked as
>outlined, so I did the big test; I tried to unlock my 350 that I've been
>trying to unlock for months.
>
>There isn't a Mute button, so I hit the Flash button instead. Voila, it
>said "M
That is an annoying, arguably misfeature, of the Aastra. The idea is that
the use of the programmed buttons should eliminate the need for the "Link"
button since manual flash hooks can get your phone out of sync. Don't
worry you can use manual flash hooks in the mean time.
Mark
On Thu, 27 Mar 2
On Thursday 27 March 2003 13:23, Michael K. Rodriguez wrote:
> This the error I receive when I try to record a GSM file
I don't know if this applies to your situation, but I get much
better quality if I record using record (22k sampling) and then
using sox to convert it to gsm. Here is my script
Mark Spencer wrote:>>So it looks like the best codec is the GSM codec as
far and badwidth
vs voice quality, but I can't seem to find which hard phones support
the GSM codec or if * supports the G.729 codecs or others..
Which phones do the * user commumity find work the best?? and which
codecs do y
> Not trying to be difficult, but is that a "purchase" or a time-bound
> license?
>
> Just curious. I'm very interested in using it, but the $$ commitment is
> something of a question.
That's a purchase.
Mark
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On Thu, Mar 27, 2003 at 04:45:20PM -0800, TC wrote:
> I have a 480 & it is completely *UNLOCKED* why dont you post that getCPEID
> data & lets see if we have the same version Info
Which security code did you use to program it? Just 0x (that is
the code for *UNLOCKED* phones, right?)?
Jays
Does anyone know if this unit works with Asterisk?
Thx.
B.
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I am using a clarent CPG-101 which I think is actually the same hardware.
MGCP works fine in the informal testing I have done. Hooking it up with an
X100P does give you some echo. I am just getting started with * so I can't
really tell you more than it seems to work fine.
- Original Message
I wonder what I'm doing wrong here, trying to use kphone with asterisk.
I have the following entry in sip.conf:
[kphone]
type=friend
username=brianc
secret=mysecret
host=192.168.1.4
context=home
When I go to the "Identity Tab" in kphone, and ask it to register with
my asterisk server, I see the
Thanks for your good advice.
> You shouldn't need ZERO_BATT_RING unless you're using a line coming off
> of a TA, and if you are then you may not get disconnect supervision,
> depending on your TA.
> In any case, the disconnect supervision (AKA CPC) on PSTN lines in
> Japan is very very short.
> L
> I like the reasoning. In fact it could apply accross the
> board: Have every channel type disabled unless specifically
> enabled. I would assume most Asterisk sites don't use most
> channel types.
>
I managed to set the following:
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload
On Fri, 28 Mar 2003, Brian Capouch wrote:
> I wonder what I'm doing wrong here, trying to use kphone with asterisk.
>
> I have the following entry in sip.conf:
how about s/[kphone]/[brianc] or conversely s/brianc/kphone
> [kphone]
> type=friend
> username=brianc
> secret=mysecret
> host=192.168
How do I do a manual flash hook? And sorry, I didn't elaborate enough - I'm
using the Nortel Vista 390. I don't see a flash button anywhere, only the
link. And the link doesn't work when I load it with *'s adsiprog. Once I
dump it, it's fine again . .
At 10:33 PM 3/27/2003 -0600, you wrote
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