Im not sure, but there is a commented column that could have 0(not
commented) or 1(commented) as values.
Is this right?
P.S.: I got it from voip--info.org on the realtime Static page...
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1610A
CEP: 80530-000 -
Very good news.
Really good to know about the success of companies(like Digium) and
developers(like all mentioned by Kevin) that are working with and for
the Asterisk community.
I just have one thing to complain:
When will Digium invite a developer to put the MFCR2 stack(channel)
on
Someone could help me on troubleshooting this error?
DEBUG[13314]: Didn't get a frame from channel: SIP/
When passing a fax over a PRI channel I got this error after the 4th
page. Evereything is ok if the fax has 3 pages, but on forth I got a
hangup and this message appeared on my full log:
Hi Kevin.
Where could I get more information about those boards?
Thanks,
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1610A
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977 r 101
http://www.isolve.com.br
On 25 de jun de 2006, at 07:07, Kevin P. Fleming
Hi all.
Is this possible to use an include parameter on zaptel.conf file?
I mean, I want to have a bunch of files with zaptel configurations,
each one with the configuration of one kind of board(TDM, analog, and
so on).
Thanks,
Denis Galvão
2006, at 14:32, Kevin P. Fleming wrote:
- Denis Galvão - iSolve [EMAIL PROTECTED] wrote:
Is this possible to use an include parameter on zaptel.conf file?
All Asterisk .conf files support #include, it's handled at the file-
reading level. It would have taken less time to just try
The worst thing on all Polycom IP phones is the speaker phone's poor
quality. You could not have a conference call using the speakers,
only the head phone.
Denis.
On 26 de mar de 2006, at 21:17, Avi Miller wrote:
Nick Hoffman wrote:
Hrm, well that's disappointing. If they're so slow,
Damned!
What is going on with voip-info.org this week?
I think Google Analytics is the cause...
Has anybody facing this problem too?
Denis.
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Could DUNDI help him?
Or maybe a OpenSER plus Asterisk environment...
Denis.
On 12 de dez de 2005, at 12:41, Kevin P. Fleming wrote:
Douglas Garstang wrote:
The issues of NAT, call limit handling and registration expiration
don't sound quite so bad. I think we can live with those, if we
Steve, Im receiving FAXes from an IP connection...
This is what Im talking about:
Asterisk - RxFAX - VoIP provider - PSTN - FAX
Denis.
On 16 de nov de 2005, at 12:34, Steve Underwood wrote:
app_rxfax and app_txfax do not work across VoIP channels.
Wait for the next UTStarCom version... Called F3000, Im not sure, but
something like that.
It will have better battery performance and will have 802.11g
support, and many other improvements. It will be available soon.
Denis.
On 07 de out de 2005, at 00:54, Andy Hamilton wrote:
Anyone
Put on the list the software version that you are using.
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977 r
http://www.isolve.com.br
On 09 de set de 2005, at 02:29, Le Van Khoa wrote:
Hi,
I run
Did you use the 1.1.x version of the patch and chan_unicall.c ?
Denis.
On 05 de set de 2005, at 20:57, Anton Krall wrote:
Guys.
Anybody gotten unicall to compile under cvs-head? I get a lot of
errors
while under 1.0.9 everything compiled without a hickup.
Any hints?
Hi Guilhermo.
Could you share with us your experience?
What is the hardware(CPU, RAM, etc) that are you using for this server?
What is your Linux distribution?
How many concurrent calls do you have in the high traffic moment?
Which is the unicall version that are you using?
Thanks a lot!
- Original Message - From: Denis Galvão - iSolve
[EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 9:47 AM
Subject: [Asterisk-Users] 100% CPU with Unicall and * head
Hi all.
When I place a call Im getting this error:
Jul 25 09:50:07 WARNING[3200
:
Hi denis
I am using Country ve,10,4Venezuela 10 ani 4 dnis
please let me know if I can do some test, or anything to help
Thanks
- Original Message - From: Denis Galvão - iSolve
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi all.
When I place a call Im getting this error:
Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on
'UniCall/13-1', but no exception handler
Lots of this messages appeared on my Asterisk full log and the CPU
got 100%.
Topology:
Analog Phone - Hicom300H E1 - E1 Asterisk -
Maybe we can have a wiki section with success stories using Asterisk
CVS HEAD. Some new features tested and succefully used.
It could be a point to start a 1.2 documentation.
I'm available to do it, or better, to put some success stories on it.
Denis.
On 23 de jul de 2005, at 09:52, Olle
I will have some extensions behind an E1. All of them will need the
features/applications of Asterisk.
Analog Extensions - PABX E1 - E1 Asterisk IP - VoIP trunk
^
|
|
Anybody using Asterisk HEAD with chan_unicall ?
Denis.
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To UNSUBSCRIBE or update options visit:
We are using it too, withouta problem.
SipGetHeader and realtime works like charm.
I just didn't get spandsp working... It compiled ok, but doesn't work.
Denis.
On 06 de jul de 2005, at 13:56, Kevin P. Fleming wrote:
Tony Mountifield wrote:
Anyone here in the know about when HEAD will
Hi all.
What is the best hardware configuration to handle this following
scenario?
- 4 IVR menu with conference applications for each option;
- Only SIP/g711 user access
- 3500 simultaneous users(800 at the beginning)
- No ZAP channels
Where is the most important point of failure? CPU?
Hi William.
On 07 de jul de 2005, at 18:39, William Boehlke wrote:
If your users are business people they ratio to 1100 simultaneous
business
calls and you will need 6-9 Lintel servers, again depending on the
conferencing load and the transcoding.
I think that I will be in this case. That
IAX doesn't use INBAND DTMF.
Denis Galvão.
On 01 de jul de 2005, at 03:23, Mark Edwards wrote:
Hi.
Probably been asked before, but my IAX provider assures me its not
their problem
I have a IAX connection to a peer providing a DID. I am dialing up
my number, seeing the DTMF tones
Where?
Denis.
On 28 de jun de 2005, at 19:42, Sjaak Nabuurs wrote:
Hello
Just for fun a rss newsreader for the asterisk users and biz list.
Easy to use and now with the complete history to search.
Just use it if you like
Thanks
Sjaak
___
Hi Steve.
I think the proxy authorization is just for WWW access(tcp 80 and 443),
if some VoIP port is open you will be able to access your provider
without auth.
Denis.
On 25 de jun de 2005, at 02:22, Steve wrote:
I keep getting asked by people if these types of wifi phones are
capable
in, no
traffic goes through from that mac/ip.
- Dan
Denis Galvão - iSolve wrote:
Hi Steve.
I think the proxy authorization is just for WWW access(tcp 80 and
443), if some VoIP port is open you will be able to access your
provider without auth.
Denis.
On 25 de jun de 2005, at 02:22, Steve wrote:
I keep
On 21 de jun de 2005, at 17:20, Andrew Kohlsmith wrote:
How would you have asterisk know which IP to ring if nobody is
registered
until the phone rings??
You're right Andrew. I didn't thought about the ring...
Honestly -- what's wrong with
SIP/location1SIP/location2SIP/location3 ?
For
On 21 de jun de 2005, at 14:18, Jay Milk wrote:
|Rich is indeed correct, Asterisk does not yet support multiple
|registrations for a single peer entry. Thus when you register
|the previous registration is discarded and the new one is
|used. Thus like he said, the last one that registered gets
I got the same error ona TDM04B...
Comment out this line on zaptel/zconfig.h and recompile zaptel.
/*
* Uncomment if you happen have an early TDM400P Rev H which
* sometimes forgets its PCI ID to have wcfxs match essentially all
* subvendor ID's
*/
/* #define TDM_REVH_MATCHALL */
Hope it
/viewcvs.cgi/asterisk/channels/chan_sip.c?
rev=1.713view=markup
Regards,
Denis Galvão
On 14 de jun de 2005, at 12:48, Charles Wang wrote:
Where is the function? On source codes or any config file?
On 6/14/05, Denis Galvão - iSolve [EMAIL PROTECTED] wrote:
Hi all.
Could someone point me an example
Hi all.
Could someone point me an example to use SIP_HEADER function!? I want
to read the To: and send this INVITE to an internal extension.
Tks.
Denis Galvão
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Asterisk-Users@lists.digium.com
Hi all.
Im participating of a project(a huge one) that will study Asterisk as its
PABX base system.
They ask me: Who is using Asterisk as its base PABX!?
Now I ask you: Anyone know about some important and big company that have
been implemented Asterisk!?
Im not talking about VoIP
If you speck portuguese, visit AsteriskBrasil.org:
http://www.asteriskbrasil.org
Regards.
Denis.
Em Qui 03 Mar 2005 22:23, Paul Davidson escreveu:
All-
I am considering an Asterisk implementation in Brazil. Unfortunately,
this presents something of a challenge to plan sitting in Chicago,
Em Qua 02 Mar 2005 16:52, skamp escreveu:
Thats kinda lame who uses their machine and runs apps as root
ughhh, can i install it as root and run it later as the user ?
I installed as normal user... But didnt get the app running Just dont
appear...
Is there anything else to do!?
Hi All!
I have the folowing need:
We have a project in Brazil called Quinta Livre(Free Thursday) where we have
one speech about some Open Source project, every last thursday of every
month...
We want to make this presentation avaliable to more people, so we have to
broadcast this
Send us your DIAX configuration.
Denis.
Em Seg 21 Fev 2005 07:29, Bartosz Wegrzyn - asterisk escreveu:
I did change the port 4569.
Also my router forwards those packets.
If I start tcpdump port 4569 on my server I receive:
04:25:36.061292 IP 192.168.1.253.4569
Hi Dan.
' - audio delay when IAX bridging inside Asterisk
Will it cover that problem of long delays that we talked before!?
Regards,
Denis Galvão.
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
With this version I cant use my ATCom usb phone.
I didnt see it at the USB Phone options at the DIAX softphone menu. Only
yealink and eutectics.
Denis.
Em Qui 17 Fev 2005 11:44, Dan escreveu:
Hi Denis,
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
' - audio
Im proud to announce that our email list is already working!!!
I want to invite all of you to participate in our community!
http://www.asteriskbrasil.org
We are almost complete with the development of our portal, that will include
a lot of resources(translation os white papers, howtos, digium
Em Qua 02 Fev 2005 01:05, [EMAIL PROTECTED] escreveu:
snip
Surely there has to be one soft phone that works under Linux.
I've tried:
kphone - it sometimes complains about the need to release the sound
device
linphone - lowww
iaxcomm - needs some strange
Hi Max.
We are providing a brazillian Asterisk comunity. Our domain is
asteriskbrasil.org, and as soon as possible we are providing brazillian
portuguese content of Asterisk and all of documents needed to assist you an
other brazillians to install/configure and use Asterisk.
Em Qua 02 Fev 2005 17:34, Daniel del Castillo escreveu:
I'm having problems trying to run zaptel. I don't have the hardware, I
first want to test out asterisk. The problem is the usb-uhci/usb-ohci
module, it isn't present on the system as same as usbcore and I don't
know why. Any tip?
Do you
Hi Cesar.
Try it out:
http://iaxclient.sourceforge.net
--
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 252-2977
http://www.isolve.com.br
Em Ter 01 Fev 2005 15:22, César Davi Ávila do Nascimento escreveu:
Hi
Em Ter 01 Fev 2005 16:27, Bruno Hertz escreveu:
On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote:
Unattended transfers just does nothing. I cannot get it to do anything.
Not sure about this, but I'm under the impression that the # transfer
might need some client support.
E.g. I tried
Hi Michael.
Any work to support some USB Phones!? The ability to dial using the phones
keypad!?
Thanks.
Denis.
Em Sáb 29 Jan 2005 01:11, Michael Van Donselaar escreveu:
iaxComm is an Open Source softphone for the Asterisk PBX.
iaxComm compiles and runs on Win32, Linux and Mac OS X
Em Qui 27 Jan 2005 05:18, Dan escreveu:
Hi Denis,
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
Hey I tried DIAX today and the speech quality was rather poor
compared to X-lite.
Dan, do you know wich iaxclient version firefly is build on!?
I got
Em Qua 26 Jan 2005 20:01, Dan escreveu:
Hi,
Hey I tried DIAX today and the speech quality was rather poor compared
to X-lite.
Dan, do you know wich iaxclient version firefly is build on!?
I got better results(voice quality) using firefly, doesn't matter what CODEC
I used.
Regards.
--
Same for me... No confirmation...
Denis.
Em Ter 25 Jan 2005 17:38, Keith Burns escreveu:
Ok, I signed up a few hours ago for the AMP mailing list, and no
confirmation.
If anyone on this list has installed AMP with SUSE 9.2, if you wouldn't
mind emailing me with any gotchas at [EMAIL
Em Sáb 22 Jan 2005 07:51, Dan escreveu:
Hi all,
There is someone on this list having latency issues with DIAX who can
do this trace? I'm not able to dupplicate this behaviour here and as I'm
behind
a NAT I cannot use 2 DIAX phones connected to an external Asterisk
server (or there is a
Hi all.
Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!?
Somehting that I need to know before update!? How is the best way to get my
system updated!?
Thanks.
Denis.
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Asterisk-Users mailing list
Hi all.
Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!?
Somehting that I need to know before update!? How is the best way to get my
system updated!?
Thanks.
Denis.
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Asterisk-Users mailing list
Sorry about the repost. I got an error in the first one.
Denis.
Em Qui 20 Jan 2005 14:48, Denis Galvão - iSolve escreveu:
Hi all.
Somebody knows if AMP will work with the newest version of
asterisk(1.0.3)!?
Somehting that I need to know before update!? How is the best way to get
my system
Em Seg 17 Jan 2005 19:13, Steve Kann escreveu:
I've already replied, asking for a trace.. If you get the trace, and
send it, we can look at what is actually happening:
Quote
What would really help, though, is a packet trace of the call. The
best way to get this is to use either ethereal
Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu:
beonice wrote:
Ouch ... error while writing audio data: : Broken
pipe
What are the messages before this?
Matt I think that is something related to mpg123...
--
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu,
Em Ter 18 Jan 2005 21:27, beonice escreveu:
That _seems_ to be a possibility. But I'm not really
sure. I made sure that there is a symbolic link in
/usr/bin to mpg123 ... the actual version is in
/usr/local/bin.
Thanks. By the way, I accidentally created a new post
with the details of the
Hi Dan, Steve, Michael, Bruno and others.
I will try to describe my VoIP environment below:
SERVER:
- FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17
- iax.conf
[general]
bindport = 4569
bindaddr = 0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
tos=lowdelay
jitterbuffer=no
2005 11:51, Denis Galvão - iSolve escreveu:
Hi Dan, Steve, Michael, Bruno and others.
I will try to describe my VoIP environment below:
SERVER:
- FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17
- iax.conf
[general]
bindport = 4569
bindaddr = 0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
Em Seg 17 Jan 2005 13:51, Steve Kann escreveu:
Yes, it sounds like there's a discontinuity in the timestamps when you
set up your call, but it seems Dan can't reproduce this.
The fix is probably:
a) The jitterbuffer needs to be reset after the transfer, or
b) The timestamps sent need to be
Em Seg 17 Jan 2005 13:43, Dan escreveu:
Hi Denis,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
...
- Same problem with DIAX oldest DLL;
It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0
Please try an older version of DIAX, like 0.9.8c.
You can still download it from
Digium is the company behind the Hardware to Asterisk.
Try its website:
http://www.digium.com
They have a developers kit that could reach your needs.
Denis.
Em Seg 17 Jan 2005 14:13, [EMAIL PROTECTED] escreveu:
Hello All,
Please forgive the lack of understanding as of yet but I have been
escreveu:
Hi Denis,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
...
- Same problem with DIAX oldest DLL;
It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0
Please try an older version of DIAX, like 0.9.8c.
You can still download it from:
http://www.laser.com/dante/diax
recusada.
Denis.
Em Seg 17 Jan 2005 15:59, Dan escreveu:
Hi,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
Same problem with version 0.9.8c
After one minute aprox, delay disappear.
Any ideas!?
Can you check the older ones too?
http://www.laser.com/dante/diax/diax097a.zip
http
Exactly the same problem for all of them(097a, 096d and 095).
The delay is getting down by the time of conversation. After aprox 1 minute,
or even less, the delay is totaly off.
Denis.
Em Seg 17 Jan 2005 15:59, Dan escreveu:
Hi,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
Same problem
Here in Brazil we are creating our language email list. For us it will be
great because we have a lot of different terms, technologies of US and
Canada, even Europe.
I think that for brazillians, the first place to solve a problem could be
the brazillian list, after that a universal list(this
Em Seg 17 Jan 2005 16:47, Dan escreveu:
Hi Denis,
Same problem with version 0.9.8c
After one minute aprox, delay disappear.
Any ideas!?
What's very strange is that I cannot reproduce this behaviour, trying
with different PCs and different DIAX versions and settings.
Which Asterisk version
Hi Dan.
I have the same delay problem with diax 0.9.9g.
This problem just happen with DIAX softphone, with others(iaxcomm, firefly,
etc.) doesn't occur.
Im using an ATCOM compatible USB Phone.
Is there anything that I can do to solve this issue!?
Thanks in advance.
Denis.
Em Sex 14 Jan
Hi Matt.
Same problem with 0.9.9g...
Thanks.
Denis.
Em Sex 14 Jan 2005 03:17, Matt Riddell escreveu:
Denis Galvão - iSolve wrote:
Hi all!
Somebody knows something to do with a high delay using Asterisk +
DIAX!?
Try grabbing 099g released today...
QUOTE: DIAX 0.9.9g is available
Em Sex 14 Jan 2005 15:11, Michael Van Donselaar escreveu:
The iaxclient default latency for windows was changed about two months
ago to 40.
There were a couple of reports of audio distortion, so it was kicked up
to 67.
I think you can get pretty agressive with this, just remember to check
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
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
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, EchoTest,
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
Hi all!
Somebody knows something to do with a high delay using Asterisk + DIAX!?
When I used IAXComm(Linux) in both sides(peer and me) no problems.
Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the
voice coming from the person that I called. I don't have delay in my
Em Qui 13 Jan 2005 21:06, Steven Critchfield escreveu:
On Thu, 2005-01-13 at 16:41 -0600, Matthew Boehm wrote:
OMG! 1 hour?!?! I just now got this at 4:40PM. It takes an hour for my
emails to get posted to the list? Geez..
If you need responses in faster time than 1 hour you need to
Dan, I'm using DIAX(Portuguese Language) for one week too... Without any
problems.
I've tested with all suported CODECs... But Im using with a-law and u-law
for now.
If you need some help to translate to Brazillian Portuguese, call me!
I like the incoming calls ring... ;)
Denis.
Em Seg 10
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