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 e
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 c
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. --
__
- 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
- 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 bu
- 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.
--
Tilghma
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
permit=127.0.0.1
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 Ma
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
w
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 - Brasil+55
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
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 dahdi
en => s,1,Set(DB(fax/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
Gerent
- "Gordon Henderson" 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 HWEC, it enables
- "Daniel Leite de Abreu" 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 timing is
- "Joao Gomes Pereira" escreveu:
> Em 17-03-2010 20:51, Vinícius Fontes escreveu:
> > - "Joao Gomes Pereira" escreveu:
> >
> >
> >> Hello
> >> Im trying to receive FAXes with my Asterisk with "rxfax" command.
&g
- "Joao Gomes Pereira" 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 module 'ap
- "Kevin Sandy" 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 G729. In
> about
- "Tilghman Lesher" 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
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
C
- "Håkon Nessjøen" 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 example,
- "Jeff LaCoursiere" 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 .
>
>
- "Steve Underwood" escreveu:
> On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote:
> > Very informative post Vinícius !
> >
> > 2010/3/5 Vinícius Fontes > <mailto:vinic...@canall.com.br>>
> >
> > - "Chandrakant Sola
- "Chandrakant Solanki" 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,
>
> Chandrakant Solanki
S
- "Mark Adams" escreveu:
> Hi, thanks for your response.
>
> I'm not sure if I explained correctly. I need asterisk to provide an
> ISDN data function, whilst also routing voice calls over the same PRI.
> Is this possible?
>
> Regards,
> Mark
>
- "Mark Adams" 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 use the ISDN30 (
- "DHAVAL INDRODIYA" 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: Present and oper
- "Brian" escreveu:
> On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
> > - "DHAVAL INDRODIYA" escreveu:
> >
> > > Dear All,
> > >
> > > How can we know the On board supports echo cancellation
> > >
> &
- "DHAVAL INDRODIYA" 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 any way to k
- "Gordon Henderson" 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 Resoluti
- "Gordon Henderson" escreveu:
> On Thu, 25 Feb 2010, Vinícius Fontes wrote:
>
> > - "Shaun Ruffell" escreveu:
> >
> >> On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
> >>> I'm playing around with an ALIX 2D2 board
> &
- "Shaun Ruffell" 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 availabl
>
> [Feb 19 15:18:32] WARNING[3121]: chan_sip.c:3751 sip_write: Asked to
> transmit frame type 256, while native formats is 0x4 (ulaw)(4)
> read/write = 0x4 (ulaw)(4)/0x100 (g729)(256)
>
> Btw call does go through but transcoding is done by processor not by
> TC400P. Did anyone encount
- "Vinícius Fontes" escreveu:
> - "Steve Underwood" escreveu:
>
> > Hi Vinícius,
> >
> > Don't post big things, like wireshark traces, to a mailing list.
> They
> >
> > are likely to ban you.
> >
> >
ders to sort out T.38 issues,
> as
> many of them have very little understanding 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
- "Steve Underwood" 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 contains exactl
> > 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?
--
_
> 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 macr
>
> - "Kevin P. Fleming" 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&#
>
> - "Kevin P. Fleming" 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&#
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
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
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 Sec
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
er has 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 - R
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
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 - R
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
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'll 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
- "Kevin P. Fleming" 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 pee
- "Steve Underwood" escreveu:
> On 02/03/2010 12:45 AM, Vinícius Fontes wrote:
> > - "Kevin P. Fleming" escreveu:
> >
> >
> >> Vinícius Fontes wrote:
> >>
> >>
> >>> I couldn't agree more
- "Vinícius Fontes" escreveu:
> - "Kevin P. Fleming" escreveu:
>
> > Vinícius Fontes wrote:
> >
> > > I couldn't agree more Steve.
> > >
> > > Is there any other info I could provide in order to help you find
>
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
- "Kevin P. Fleming" 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
&g
- "Steve Underwood" 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 Fon
e 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 Segurança d
ecure=port,invite
qualify=yes
context=ddi
callgroup=1
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
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:
>
e's as few 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" 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
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
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" escreveu:
> Sorry if this is off topic
>
&g
apt-get install build-essential
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- "hadi motamedi" 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
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" escreveu:
> I have a pretty large setup on one of my customers. Digium TE420B
> (with echo cancelling module), 3 Xorcom Astribanks with 32
0 0 Thermal event interrupts
SPU: 0 0 Spurious interrupts
ERR: 0
MIS: 0
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
___
-- Bandwidth and Colocati
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" escreveu:
> I think that if I could go back and do this project over, I would have
> cho
ng list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
I second that. Spectralink 8002 phones are very good, specially when using a
managed wifi solution like the 3Com WX1200.
The only thing you must pay attention is no matter wha
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" escreveu:
> Hi,
> I just got a Xorcom Astribank with 8 FXS but it doe
l inter-server communication to be done via DUNDi and
IAX, because all IAX users/peers have trunk=yes, and use g729. In no
circunstances a call should be routed directly to a different server without
passing by DUNDi first. As you could see on the dialplan I attached, I first
tried dialing the extensi
verything and make the universe collapse into itself when I apply
the same principle on production?
I'll be happy 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 d
so it's 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.
up to minimize
latency on the IP side?
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- "Jason Baker" escreveu:
> Well I tried Doug's suggestion and the echo is now better, but when I
> call an outside analog line
rks much better than Asterisk's default MG2.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- "Jason Baker" escreveu:
> Well I tried Doug's suggestion and the echo is now better, but when I
> call an outside analog
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
- "K
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 Nich
ke 90%)
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" escreveu:
> Hi, we are experiencing a strange issue and I am hoping some
- "Ondrej Valousek" 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!
>
for all answers, I have tried hard to google out them, but
>
> no success so far.
> Ondrej
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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" escreveu:
> Any way to initiate a call and execute a playbac
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 -
oduct. 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
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 S
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 f
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
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
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
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
/2-1 is ringing
[Jun 10 16:36:05] -- Zap/2-1 answered SIP/200-b6f175c0
[Jun 10 16:36:09] -- Hungup 'Zap/2-1'
[Jun 10 16:36:09] == Spawn extension (macro-ramalzap, s, 5) exited non-zero
on 'SIP/200-b6f175c0' in macro 'ramalzap'
[Jun 10 16:36:09] == Spawn e
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 Comunicaçõe
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
Desenvolvi
ing]
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
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:
There were some really unstable Asterisk releases in the 1.4 branch. I
personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16
or higher I had problems.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- "Steve Totaro" <[EM
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 REA
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 w
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]> es
Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver?
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
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