I'm planning to migrate 500+ extensions from a legacy PBX to Asterisk. Some
will be SIP, some will be DAHDI FXS.
I want to deploy a load-balancing cluster using DUNDi with regcontext so all
servers will know where to find all extensions. DAHDI extensions will have
their dedicated server, SIP
Hello.
Considering the following setup:
Legacy PBX --(ISDN)-- Asterisk --(MFC/R2)-- PSTN
When a user dials out, Asterisk receive overlap digits, matches them to an
extension and dial the PSTN, completing the call. So far so good.
The issue I'm trying to solve (or at least improve) is the
I'm trying to run a shell command from AMI, but I guess I'm doing something
wrong or there's a bug because no matter what command I try I always get a null
response. Running the latest 1.6.2 release.
On manager.conf I have:
[test]
secret = test
deny = 0.0.0.0/0.0.0.0
- Mensagem original -
On Wednesday 16 March 2011 13:14:40 Vinícius Fontes wrote:
action: command
command: ! /bin/ls -l /
For security reasons, you cannot do this. This is intentional, not a bug.
Consider the command 'rm -rf /' for the reason why.
--
Tilghman
I understand
- Mensagem original -
On Wednesday 16 March 2011 14:11:21 Vinícius Fontes wrote:
I understand the concern with security but why not create a separate
authorization allowing that instead of hard-coding it?
I understand the concern with security but why not create a separate
- Mensagem original -
On 17/03/11 9:53 AM, Vinícius Fontes wrote:
No increased security, lots of hassle, all because there's an
undocumented feature that is supposed to increase security but just
takes functionality away.
If you really want to you could add some dialplan like
But what about if asterisk running with non-privilege user?
Still it is not a good idea.
Yes I forgot to say that I also run Asterisk as a regular user, which also
helps with security.
But I really don't see much of a threat on this. AGI does almost the same. --
That makes two of us. I tried asking on asterisk-dev but had no reply.
- Mensagem original -
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
--
Ishfaq
I'm setting up a queue for two independent operator phones that are capable of
answering multiple calls at once. It's currently working with the following
settings and Asterisk 1.4:
queues.conf:
[telefonistas]
strategy=roundrobin
;strategy=leastrecent
music=default
timeout=60
retry=0
maxlen=0
Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS -
You probably want "core show channels verbose".Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brazil+55 54 2104-7000Hi,for
There's no such concept in Asterisk. Everything is a call, doesn't matter its direction.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo
/count)=$[${DB(fax/count)} + 1])
exten = s,n,Set(FAXCOUNT=${DB(fax/count)})
exten = s,n,Set(FAXFILE=fax-${DB(fax/count)}-rx)
exten = s,n,Set(LOCALSTATIONID=5421047008)
exten = s,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE}.tif)
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia
- Gordon Henderson gordon+aster...@drogon.net escreveu:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo
cancellation in
asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I
understand that
when Dahdi detects no
- Daniel Leite de Abreu dlab...@gmail.com escreveu:
Hi there , are you using any king of Iax trunk or Duguim interface on
this VM?
Because if is just for sip you dont need dahdi you can compile
asterisk and work on it.
He will need DAHDI if he plans on using MeetMe().
Also, internal
- Joao Gomes Pereira gomespere...@startel.pt escreveu:
Em 17-03-2010 20:51, Vinícius Fontes escreveu:
- Joao Gomes Pereiragomespere...@startel.pt escreveu:
Hello
Im trying to receive FAXes with my Asterisk with rxfax command.
To do that, Im trying to load
- Kevin Sandy kevin.sa...@snohio.net escreveu:
We're having an odd issue with codec negotiation from one of our SIP
providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and
G723. In our response, we send back that we support G711 and
- Joao Gomes Pereira gomespere...@startel.pt escreveu:
Hello
Im trying to receive FAXes with my Asterisk with rxfax command.
To do that, Im trying to load the app_fax.so module but asterisk
says:
[Mar 17 20:06:04] WARNING[11907]: loader.c:359 load_dynamic_module:
Error loading
Does the application PGSQL has been removed from Asterisk? Couldn't find it on
Asterisk source and addons.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall
- Tilghman Lesher tles...@digium.com escreveu:
On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote:
Does the application PGSQL has been removed from Asterisk? Couldn't
find it
on Asterisk source and addons.
That application has never been a part of Asterisk in the first
place
- Steve Underwood ste...@coppice.org escreveu:
On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote:
Very informative post Vinícius !
2010/3/5 Vinícius Fontes vinic...@canall.com.br
mailto:vinic...@canall.com.br
- Chandrakant Solanki solanki.chandrak...@gmail.com
- Jeff LaCoursiere j...@jeff.net escreveu:
On Thu, 4 Mar 2010, Steve Howes wrote:
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank
- Håkon Nessjøen haa...@avelia.no escreveu:
Hi,
I have read that DAHDI automagically turns of echo cansellation when
it sees that it is a FAX.
So I checked this out. I have a fax call into asterisk which is
immediately called out to an external fax machine via DAHDI again..
For
, Vinícius Fontes vinic...@canall.com.br
wrote:
- Mark Adams m...@campbell-lange.net escreveu:
Hi All,
I'm about to setup an Asterisk install to take over an old legacy
PBX
system. At present, the legacy system has modules in it which
provides
4
* data ISDN links to the video
- Chandrakant Solanki solanki.chandrak...@gmail.com escreveu:
Hello
I have successfully compiled OSLEC for echo cancellation for DAHDI
channel.
Is there any way to do echo cancellation for SIP Channel.
Is any, please suggest me.??
Thanks in advance..
--
Regards,
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Hi,
Carlos
I checked dmesg on my server and i found following message
what is meaning for this ? i cant understand
VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450:
- Mark Adams m...@campbell-lange.net escreveu:
Hi All,
I'm about to setup an Asterisk install to take over an old legacy PBX
system. At present, the legacy system has modules in it which provides
4
* data ISDN links to the video conferencing unit (Tandberg 3000 MXP)
on
site, these
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02) board
all working fine but sometimes i got echo when user are calling a PRI.
is there
- Brian brel.astersik100...@copperproductions.co.uk escreveu:
On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard
- Gordon Henderson gordon+aster...@drogon.net escreveu:
On Thu, 25 Feb 2010, Vinícius Fontes wrote:
Just checked and I'm using the high res timer as well:
Feb 25 17:42:32 voyage vmunix: [ 27.028798] dahdi_dummy: Trying to
load High Resolution Timer
Feb 25 17:42:32 voyage vmunix
- Shaun Ruffell sruff...@digium.com escreveu:
On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
I'm playing around with an ALIX 2D2 board
(http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system
using an AMD Geode processor with 256MB of RAM. Also available are two
network
- Gordon Henderson gordon+aster...@drogon.net escreveu:
On Thu, 25 Feb 2010, Vinícius Fontes wrote:
- Shaun Ruffell sruff...@digium.com escreveu:
On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
I'm playing around with an ALIX 2D2 board
(http://www.pcengines.ch/alix2d2.htm
anyone encounter such problem?
I never used a TC400 card so I don't have much knowledge on that, but you
really shouldn't be using Zaptel anymore. Upgrade to DAHDI, that might solve
your problem.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
- Steve Underwood ste...@coppice.org escreveu:
Hi Vinícius,
Don't post big things, like wireshark traces, to a mailing list. They
are likely to ban you.
The first two calls in your wireshark log decode to the attached
images.
There were no lost packets. The wireshark logs
of the systems they have.
Even big carriers can be very unresponsive, because they just don't
know
what to do.
Regards,
Steve
On 02/18/2010 12:19 AM, Vinícius Fontes wrote:
Can you try the attached version of spandsp with the T.38 gateway
you
are using? It behaves a lot more like FFA, and I
- Vinícius Fontes vinic...@canall.com.br escreveu:
- Steve Underwood ste...@coppice.org escreveu:
Hi Vinícius,
Don't post big things, like wireshark traces, to a mailing list.
They
are likely to ban you.
The first two calls in your wireshark log decode to the attached
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
Will do. You guys will have my feedback on monday. If everything
goes okay with that change, I'll post a patch on Mantis.
No need for the patch; it's already on my radar, and if you confirm
You could try defining the same identity string for app_fax that you
have defined for FFA. Trying to make the other things more similar
would
require additional work. Maybe you should try that change first, as it
is very simple, and requires no code changes.
My receiving fax macro,
He probably means AgentCallbackLogin
While it has been deprecated, that hasn't been removed, either. If
an
enterprising person would like to try to fix it, I don't have an
objection.
Wasn't AgentCallBackLogin() removed in 1.6.1?
--
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
Will do. You guys will have my feedback on monday. If everything
goes okay with that change, I'll post a patch on Mantis.
No need for the patch; it's already on my radar, and if you confirm
Unfortunely it didn't solve the problem. Here's the session packet capture
after editing app_fax.c. http://www.canall.com.br/wireshark_t38_jbig.gz
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
You want to set it like this on Asterisk:
tos_sip=cs3
tos_audio=ef
tos_video=cs4
And in Polycom config:
qos.ip.rtp.dscp=EF
qos.ip.callControl.dscp=24
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104
Will do. You guys will have my feedback on monday. If everything goes okay with
that change, I'll post a patch on Mantis.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security
be happy to provide you any info that could help solve this issue.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS
Have you tried to set srvlookup=no on your sip.conf?
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54
I solved similar issues by setting srvlookup=no, having bind running locally
and just the line nameserver 127.0.0.1 on /etc/resolv.conf.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information
Yes, the fax machine only transmits at 9600. That's normal and expected. I'll
capture the packets and will provide you with a link to download it in a few
minutes.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
Could be. Important thing is the problem was solved :)
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55
the address 10.150.65.16 and my
box has the address 10.153.66.146.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
There you go: http://www.canall.com.br/wireshark_trace_t38_ffa.gz
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS
transmit_t38: Shut down T.38 on
SIP/voxip-
Note how items like T38FaxUdpEC are listed as OK on one call and unsupported on
another one. Could that be a bug? I can show the entire SIP conversations if
that's necessary for debugging this.
Atenciosamente,
Vinícius Fontes
Gerente de
- Steve Underwood ste...@coppice.org escreveu:
On 02/02/2010 10:11 PM, Kevin P. Fleming wrote:
Steve Underwood wrote:
Hi Kevin,
On 02/02/2010 09:12 PM, Kevin P. Fleming wrote:
Vinícius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you find
out what's wrong? I could even open an issue on Mantis if the Digium
staff think it's worth it.
Post a 'sip
You'll find out the benefit when you change anything on your dialplan, making
it necessary to alter the dialplan on every FXS port of your gateway. :)
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
- Vinícius Fontes vinic...@canall.com.br escreveu:
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you find
out what's wrong? I could even open an issue
- Steve Underwood ste...@coppice.org escreveu:
On 02/03/2010 12:45 AM, Vinícius Fontes wrote:
- Kevin P. Flemingkpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I've put it on pastebin because is was a lot of text. Here's the
link: http://pastebin.com/m7467cea1. That's all the information on the
CLI with verbose=3 and sip set debug peer voxip.
OK, with the complete
pickupgroup=1
call-limit=10
disallow=all
allow=g729
allow=alaw
t38pt_udptl=yes
t38pt_usertpsource=yes
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
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That's a pretty crappy phone huh? :)
Anyway you should be able to do it on features.conf, in the applicationmap
section. I'm not entirely sure there's a dialplan app that allows you to put a
channel on hold and take it back later.
Vinícius Fontes
www.asteriskforum.com.br - Informações e
I have never used that card myself, but I have never seen an analog board
reporting a RED alarm. Probably there is something incorrect in your
configuration. Please post your /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf.
Vinícius Fontes
www.asteriskforum.com.br - Informações e
transcoding as
possible, or no transcoding at all. With that in mind, you can have as many
cards/ports as your hardware can physically handle.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- das sandesh sandesh...@gmail.com escreveu
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
signalling=fxs_ks
channel = 1
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- listu...@spamomania.co.uk escreveu:
On Mon, 2009-12-14 at 14:17 -0300, Vinícius Fontes wrote:
I have never
That will increase the gain on the tranmission side of the phone. That's
exactly what you need.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Robert Grignon rgrig...@fleetone.com escreveu:
Sorry if this is off topic
I have
apt-get install build-essential
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- hadi motamedi motamed...@gmail.com escreveu:
Sorry . I tried to install gcc but I got the following error :
#apt-get update
#apt-get install gcc
Try installing DAHDI from source in the guest, and instead of starting it as
usual try fooling Asterisk with the /dev hack you did.
That way you would have all the dependencies for compiling Asterisk and could
still use the devices you made available in /dev.
Vinícius Fontes
interrupts
ERR: 0
MIS: 0
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
___
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AstriCon 2009 - October 13 - 15 Phoenix
And of course I forgot the most important stuff:
Asterisk version: 1.4.22
DAHDI Linux: 2.2.0.2
DAHDI Tools: 2.2.0
- Vinícius Fontes vinic...@canall.com.br escreveu:
I have a pretty large setup on one of my customers. Digium TE420B
(with echo cancelling module), 3 Xorcom Astribanks with 32
Just out of curiosity, what managed switch you used on this project?
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Jason Baker jba...@glastender.com escreveu:
I think that if I could go back and do this project over, I would have
.
The only thing you must pay attention is no matter what kind of access point
you have, they *must* support WMM or else the phones won't work at all.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
This is the new Astribank2 unit. It will only work with DAHDI 2.2.0 or higher.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Loic Didelot ldide...@mixvoip.com escreveu:
Hi,
I just got a Xorcom Astribank with 8 FXS but it does
thinking of centralizing CDR as
well, storing it in a MySQL database and setting up the accountcode on the SIP
extensions to make it easy to retrieve all extensions on the same PBX for
example.
If you see any other problem with my toughts on this, any feedback would be
much appreciated.
Vinícius
to provide more details in case there are any doubts. I really
appreciate your feedback, no matter what is it. :)
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
___
-- Bandwidth and Colocation Provided
a matter of
interoperability.
Also, IMHO, G.729 has the best voice quality for a compressed codec. Many users
can't notice the difference in a call using G.711 and G.729. The same can not
always be said for other compressed codecs like GSM or iLBC.
Vinícius Fontes
www.asteriskforum.com.br
than Asterisk's default MG2.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Jason Baker jba...@glastender.com escreveu:
Well I tried Doug's suggestion and the echo is now better, but when I
call an outside analog line I still get some
latency on the IP side?
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Jason Baker jba...@glastender.com escreveu:
Well I tried Doug's suggestion and the echo is now better, but when I
call an outside analog line I still get some echo
jbmaxsize=80 is way overkill. If your jitter is really close to 80ms then you
have some serious issues on your link and it's not suitable for VoIP at all.
Try jbmaxsize=40.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Kelvin Chan kelv
That's not bizarre at all. Blind transfers will always forward the other end's
CID. Attended transfers will always forward the CID of the phone doing it.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Danny Nicholas da...@debsinc.com
%)
you would need a dedicated transcoder:
http://www.digium.com/en/products/voice/tc400b.php
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Deepak dlal...@gmail.com escreveu:
Hi, we are experiencing a strange issue and I am hoping someone
- Ondrej Valousek webs...@s3group.cz escreveu:
Hi Vinicius.
/ 1. To enable jitter buffer on SIP channels it seems I have to
enable
// and
// force it, right?
/
Not sure about the forcing part (don't know exacly how it works),
but I always set jbforce=yes to be sure.
Ok, thanks!
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre
You can use console dial num...@context. If the Asterisk box has a soundcard,
you will hear the audio and will be able to speak on the microphone.
Vinícius Fontes
www.asteriskforum.com.br
- Joseph L. Casale jcas...@activenetwerx.com escreveu:
Any way to initiate a call and execute
Downloading right now, thank you very much for sharing it with us.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS
Sure it is:
exten = blah,1,Dial(SIP/blah,30)
Where 30 is the time in seconds the application will wait before quitting and
setting the DIALSTATUS variable to NOANSWER.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS
. Too bad it's the only TDMoE channel bank
(that I know of, at least).
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS
canreinvite=yes
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
- Arno Scholz [EMAIL
Make host=dynamic.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
- michel freiha
On zapata.conf:
faxdetect=incoming
The detected fax calls will be redirected to the 'fax' extension on the context
set to the group of channels.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
I'm having problems exactly with that tone detection. I even submitted a bug
report (http://bugs.digium.com/view.php?id=13286) but it still has not been
viewed yet, I guess.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS
As RTP packets have a sequential number, is there some logging/debugging option
in Asterisk to monitor how many packets have been lost on a SIP call?
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
When people release software under the GPL license, like Steve Underwood did
with libunicall, spandsp and so on, they were supposed to know that other
people has the right to use their code.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
non-zero
on 'SIP/200-b6f175c0'
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
You could simply short-circuit the two wires of the line. The telco will
interpret that as a busy line.
Other than that, you could do this on extensions.conf:
[context]
exten = s,1,Answer()
exten = s,n,Busy()
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda
Two things you could consider trying:
1) In sip.conf, set the externip and localnet parameters correctly.
2) Also in sip.conf, try the following on the PAP2's sections:
disallow=all
allow=alaw:10
In case that fails, try also
disallow=all
allow=alaw:20
Att
Vinícius Fontes
Desenvolvimento
Mostly SIP, some of my clients have queues and everything is working fine by
now.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Benoit Plessis [EMAIL PROTECTED] escreveu:
lordfuknowsyou a écrit :
Vinícius Fontes wrote:
I use 1.4.18 with no problems
]
exten =
_00[2-6]XXX,1,MixMonitor(${CALLERID(num)}-${STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
exten = _00[2-6]XXX,n,Dial(Zap/g1/${EXTEN:1})
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Rahul Yadav [EMAIL PROTECTED
Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver?
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
There is no single reference to TDM410P in that page. Plus, genzaptelconf
detects the card and sets the driver to wctdm24xxp, that's why I asked.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Steve Totaro [EMAIL PROTECTED] escreveu:
wctdm http://www.voip
setups there is 3Com WXR100 that supports up to 3 MAPs (Managed
Access Points).
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Brent Davidson [EMAIL PROTECTED] escreveu:
A friend of mine recently told me about a phone system his office was
considering that did
Sorry, my fault. I did a
$ grep -R -i TE410P *
before asking, but in the README it was listed as TE410, so no match.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Eric Wieling [EMAIL PROTECTED] escreveu:
The README file in the Zaptel source directory
Oops, seems like I didn't realized something: the queue size can't be zero. I
solved the problem by setting maxlen=1 and defining a timeout on the Queue()
app. That way when all the agents are busy, the call gets diverted after
[TIMEOUT] seconds, which is ok to me.
Att
Vinícius Fontes
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