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Hi,
Another question. With latest version of asterisk softwares am I ableusing rxfax? I had read some remarks about incompatibility between TDMcard and rxfax. Is it still exist?
I've been using rx for fax reception with TE110P as well as X100P (this only for tests and learning) with very
Hi,I didn't try that way, only tx fax in call file. But my experience is when u r working with FAX you MUST disable echocanceller!On 8/15/06, Andy Kuo
[EMAIL PROTECTED] wrote:Hi Marco,
I'm using T406P(with hardware EC) with a T1-PRI, and I'm havingtrouble sending fax out though SIP ATA
Hi , Please post here your extensions.conf in your central server only with that i can figured out or at least try to help u.Best regards,Marco MoutaOn 8/15/06,
Juan Luis Moyano [EMAIL PROTECTED] wrote:
Hi All, I have the following scenario: A central Asterisk server whereall the ATAs register
those files
jingle.conf and jabber.conf, i mean who is who, and their goals.-- Com os melhores cumprimentos,Marco Mouta
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for format_mp3.soWhat could be wrong?I've made already several asterisk installl and never got this problem...
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must be type=user or type=friend.friend=user+peer peers cannot place calls into the Asterisk server
http://www.asteriskpbx.com/Best regards,Marco MoutaOn 8/10/06, Shaun Hofer [EMAIL PROTECTED]
wrote:I have two trunks to the same machine (x.x.x.2), one is type=friend, other is
type=peer. Asterisk
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Hi!
I have two ISDN cards for my asterisk server, an AVM B1 (active
card) and a HFC.
I want to use the HFC card in NT mode, and the AVM B1 in TE.
Afair bristuff and vISDN doesn't support the AVM B1, so mISDN
should be my choice?
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GPG Key 0x1A6C95BA -- http://www.frankonia
Hi,Your Consultant has developed it with PHP scripts, so you must check those files in /var/lib/asterisk/agi-binYour application logic is there.Hope it helps,Best regards,Marco Mouta
On 8/3/06, Randy Paries [EMAIL PROTECTED] wrote:
On 8/2/06, Time Bandit [EMAIL PROTECTED] wrote: The problem
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,
Marco Mouta
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Erik,
What a great and detailled explanation! Thank you very much!
Ps. If you know anything about legal issues asked abouta g729 please
post it here:)
Best regards,
Marco Mouta
On 7/26/06, Erik [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
By the way could any one tell me wich
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my mistake you post it! could you pos it in file.conf format?
On 7/25/06, Marco Mouta [EMAIL PROTECTED] wrote:
It seems you didn't post any thing about you [general] sip.conf
neither allowed codecs
On 7/25/06, Carlos Alberto Bernat Orozco [EMAIL PROTECTED] wrote:
Hi group
Thanks Marty
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Chattanooga, and Montgomery.
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I must say that for mailing lists Gmail seems to me just perfect! I
apreciate your integration into Forum. But Gmail seems to me even more
friendly!
Best regards,
Marco Mouta
On 7/21/06, zoa [EMAIL PROTECTED] wrote:
There are some others out there, we did something similar at
http
that just
monitors your G729 licences, and keeps on track which codec is going
to be used: Ulaw or G729.
Don't know if this is a good idea, just a suggestion.
Best regards,
Marco Mouta
On 7/21/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote:
No, we aren't intending to check for available g729
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Could you post your sip.conf?
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.
On 21/07/06, Marco Mouta [EMAIL PROTECTED] wrote:
Did you port forwar in your router RTP ports ? 1-2 to your *Box ?
On 7/21/06
Hi,
I think i found your error. you are missing a context for your peer
PeopleCall , this way no context for incoming calls!
Am I wrong?
Hope it helps,
Marco Mouta
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Here is my SIP.conf. (just replaced psswds with *)
Thanks.
[general]
port
-users
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already have set language to 'se' in indications.conf.
Next question. If asterisk where to play a digit - does it look in
/sounds/se/digits or /sounds/digits/se ?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Marco Mouta
Skickat: den 19 juli
Try to active callwaiting in those unreachable extensions. You just
need to dial *70 from every SIP extension.
Be aware that *70 (call waiting ) may be disabled in your freepbx.
Hope it helps,
Marco Mouta
Please give me some feedback
On 7/17/06, Tim P [EMAIL PROTECTED] wrote:
Not sure where
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advice me on this settings ? or is this something worst?
Best regards,
Marco Mouta
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:
- Call party A - Call duration into my database
- then call party B and bridge it with A and keep CDR of the call
duration between A and B.
Does any of you has experience with this?
Best regards,
Marco Mouta
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Ganbaa
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It would be hard to bill all this calls, if you are using dialout call
files instead of Asterisk Manager API no ?
How would you colect the call duraction of both call legs?
Thks,
Marco Mouta
On 7/6/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
Also have a look at .call files.
You web
?
Marco Mouta
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as a TE port ? I think that might be the problem!
Best regards,
Marco Mouta
On 7/7/06, Andrea Spadaccini [EMAIL PROTECTED] wrote:
Ciao James,
Hello everyone,
I'm trying to set up an Asterisk machine with a quad-port BRI
Junghanns card, and I want to use the mISDN drivers.
I'm having
Sorry i didn't get your idea.
could you explain me what you mean? Are you saying to make CDR in only
one of the legs?
Best regards,
Marco Mouta
On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello
Just use NoCDR() in the non bridged local context.
Jon
-Oprindelig meddelelse
-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta
Sendt: 7. juli 2006 13:55
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] How to collect Call duration, Dialout Call files?
Sorry i didn't get your idea.
could you explain me what you
?
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta
Sendt: 7. juli 2006 14:15
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] How to collect Call duration, Dialout Call files?
did u try asterisk manager api
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by Ports i mean Spans :)
On 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote:
Newbie guess,
Don't you need to set one of the ports NT mode and the other one as TE mode?
hope it helps
Best regards,
PS. give me some feed back if it solved.
On 7/7/06, Ralph Liebessohn [EMAIL PROTECTED] wrote
and it didn't detect my audio board...
On 7/6/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jul 06, 2006 at 02:27:59AM +0100, Marco Mouta wrote:
it has happen to me , no sound after removing x100p board , and i found,
only because i was not accessing remote the server. i was localy
://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta
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Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will
be busy if you have already 2 calls running, so the caller party
should get busy indication from your Telco...
On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
You should handle correctly Dial
Hi,
I'm planning to develop a solution with SMS using Asterisk within
Portuguese PSTN landline.
Any one has made it before?
I'm looking for Telco's and details using Portugal Telecom landline.
Thanks in advance,
--
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Marco Mouta
Hope this could help,
Please note Inband DTMF won't work unless the codec is ulaw or alaw
(G711). Use out of band DTMF aka rfc2833 or info.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode
best regards,
Marco Mouta
ps.give me some feedback if it worked
On 6/29/06, Shane
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Bom dia,
On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
Sim é um software da Uplink, disponível para download gratuitamente, n
garanto q seja freeware (talvez tenha limitações esta versao free
Sorry to all,
Now only English speaking :)
Your translation was perfect.
Thanks once more
On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito
Asterisk handling My Skype Calls
This is for me, once more, Asterisk as the Future of Telephony.
Today I've integrated my Skype Account as SIP extension in my * Box.
This has been possible using Uplink Skype to SIP Adapter, available
for free at http://www.nch.com.au/skypetosip/index.html .
/problemas e soluções nas
implementações Asterisk.
Há spre detalhes que variam entre os Telco's de cada país, voice prompts, etc.
Se houver um número minimo de pessoas interessadas, podemos avançar com a ideia.
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don't know, i must say i'm not a
web expert.
I work with VoiceXML VoIP more related to communications.
Mailing list and blog or forum seems easy to start this, share and learn.
I hope i can help to this project grow.
Best regards,
Marco Mouta
On 6/23/06, Josué Conti [EMAIL PROTECTED] wrote
instead of using
Asterisk Users List.
This is not a rule, I mean a website may be created instead of the
blog. As i've written i'm not a web expert and this was the easiest
way to do the first step, some times the most important one :)
Best regards,
Marco Mouta
Obrigado a todos os que têm participado
works fine, without any echo. When I
make external calls (pstn with digium TDM400P) I ear an echo just at the begin
and at the end of any speech. If I say short words (sounds) then I hear a lot
of echos. Isn't it a sidetone effect?
Any other ideas?
Thanks again,
Marco
On Tue, 13 Jun 2006 12:02:20
of the echotraining, of the rx and of the tx
gain, but with no success.
Any idea or help?
Thank you in advance,
Marco
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First, thank you for your quick and kind answer.
I cannot change the TX gain on the Grandstream phones, or atleast I don't know
how to...
Can anybody help, please?
Thanks in advance,
Marco
On Tue, 13 Jun 2006 08:54:31 -0600, Colin Anderson wrote
Turn down your microphone TX gains on the phones
for section called [from-sip-external], there you need to paste your code to route the call to your meetme room.Hope it helps,Best regards,Marco MoutaPs. Please give me some feeback if it solved.
On 6/7/06, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all i have an asterisk working and i need to add
with Centos and ZaptelHope it helps!Best regards,Marco Mouta
ps. give me some feedbackOn 5/30/06, David K Parker [EMAIL PROTECTED] wrote:
Has anyone been able to compile Zaptel after upgrading to 2.6.9-34.0.1.EL kernel? I'm running CentOS and was unable to recompile Zaptel. I reverted back to 2.6.9-22.0.1
check [general] section of your /etc/asterisk/sip.confdisallow=allallow=alawallow=ulawallow=gsm This codecs depends on of your SIP provider as well as activation in your SIPphone
On 5/30/06, George A. Roberts IV [EMAIL PROTECTED] wrote:
I just put in a
new [EMAIL PROTECTED] 2.8 system. Trunk
did you check your verbose level for your console?On 5/29/06, Akpome Akpoguma [EMAIL PROTECTED] wrote:
Is there any reason why I cant see the environment dump display on asteriskconsole when call
agi-test.agi from my dialplan?reponses would be
I'm also not an expert, but could it as any relationship with your Telephony card drivers??Which Telephony boards do u use?On 5/29/06, Attilla de Groot
[EMAIL PROTECTED] wrote:Hi All,
First off all, this is my first mail to this mailing-list, so if I amdoing something wrong please tell me. And
configurations.Hope it helps!
Best regards,Marco MoutaPs.Please let me know if it worked.On 5/26/06, Werner Terreblanche
[EMAIL PROTECTED] wrote:
Hi
I'm very new to Asterisk and this is my first
posting to this mailing list. I got a [EMAIL PROTECTED] V2.8 working,
and now I'm trying to use
comes in to B, then B puts A in hold, then calls C asks if C wants the call from A and then simply bridge the call to A without using park , or hung the call with C???
Best regards,Marco Mouta
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Moises Silva,I've already tried to activate:features.confatxfer=*2as well as setDYNAMIC_FEATURE=atxfer in my [globals] of extensions.confBut I couldn't get it working, that's why I asked it in the mailing list.
Thanks for your help.Best regards,Marco MoutaOn 5/11/06, Moises Silva [EMAIL PROTECTED
QSIG was just the protocol communication between Legaccy PBX and Asterisk.My users connect to Asterisk through SIPOn 5/4/06, Olivier Krief
[EMAIL PROTECTED] wrote:
2006/5/3, Marco Mouta [EMAIL PROTECTED]:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using
to work in a standardt architecture:PSTN--E1---LegacyPBX---QSIG---AsteriskI hope it helps,Marco MoutaOn 5/3/06,
Asterisk User [EMAIL PROTECTED] wrote:
I am looking to get the info about QSIG support in Asterisk.
Does Asterisk have QSIG support?
Does Asterisk support QSIG SIP Tunneling or QSIG SIP
pay a SIP
Gateway to have a geographical number that points to my asterisk
(sorry if I do not use the correct terms).
Thanks a lot!
Cheers,
Marco
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their answers without arriving at my agents, and also keep them interested while they wait in queue.Is there any project or some one who has done this before?Any tips?
Best regards,Marco Mouta
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You must activate call waiting for those extensions, this way you will get correctly voicemail busy and unavailable.From the sip extension dial *70On 4/28/06,
Johnny Stork [EMAIL PROTECTED] wrote:
I have a fairly new, but functional install of [EMAIL PROTECTED] 2.7 with a TDM400 (1 FXS) and T101P
I've been asking about this problem in Asterisk channel... I didn't report it has a bug...Probably it is recommended... On 4/24/06, Thomas Winter
[EMAIL PROTECTED] wrote:Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta:
How do I report a Bug to Digium? or asterisk project?Did you report
Hi all,I've asterisk 1.2.5 , and what is happening is this:Sip user agent A calls a pstn phone BSip User agent Activates MOH.B starts listening.A doesn't hangup and just Disconnect Sipoftphone XLITE (exit)
B stills listenning Music on Hold and A has left Asterisk, who hangs the call? only when B
Asterisk shouldn't see that the specific SIP user agent isn't there any more?On 4/19/06, Doug Lytle
[EMAIL PROTECTED] wrote:Marco Mouta wrote: Hi all, I've asterisk
1.2.5 , and what is happening is this: Sip user agent A calls a pstn phone B SipUser agent Activates MOH. B starts listening.
A
I've tested maxexpirey=120 and even with this, asterisk didn't stop the call:Scenario: SIP user agent has left without telling to asterisk it was leaving...There was a call to pstn world with MOH running...
Any tip to solve this?On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Marco Mouta wrote
qualify=yes may overload my network .. no?On 4/19/06, Gareth Blades [EMAIL PROTECTED]
wrote:Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualifyOn Wed, 2006-04-19 at 14:51, Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop the call: Scenario
How do I report a Bug to Digium? or asterisk project?On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call: Scenario: SIP user agent has left without telling to asterisk it was leaving... There was a call
on hold button it seems that it stops music on hold and starts imediately again.
Any one can guess what may be wrong?Best regards,Marco Mouta
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I forgot to write: When i hangup the call, it hangs correctly!On 4/18/06, Marco Mouta [EMAIL PROTECTED]
wrote:Hi all,I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press
=
_2,2,gotoif,$[${HANGUPCAUSE} = 16]?9|1exten =
9,1,HangupI'm not sure if this is possible neither recommended,
should be HangupCAUSE=16 or =98 ??Best regards,Marco Mouta
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to an extension in the Old PBX, that case if the called party Hangs, the Old Pbx immediately sends a DISCONNECT message to Asterisk and the call hangs...
I hope someone could help US.Best regards,Marco MoutaOn 4/12/06, Abhimanyu Rapria
[EMAIL PROTECTED] wrote:Hi,We are using Vicidial and sometime even
that Sjphone is giving timeout error because of it...Why is this 5 seconnds? any one knows?best regards,Marco Mouta
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minutes of busytone indicationsCould be the OldPBX that doesn't send the disconnect ?
Any tips?Best regards,Marco Mouta
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Hi,I've been watching my * Console and seems to be one call not well terminated or something:For 5 minutes at least my console is reporting this: ectory|general|ext-local|be: -- Playing 'letters/c' (language 'en')
directory|general|ext-local|be: -- Playing 'letters/o' (language 'en')
Hi found that it could happen just using Xlite and after dialing *411 , then change your Xlite to line2 without hanging up channel 1 My solution has been on CLI a soft hangup for the SIP channel that made this call.
I found the channel with show channels.On 4/10/06, Marco Mouta [EMAIL
= *411,1,Answer
exten = *411,2,AbsoluteTimeout(300) ; for 5 minutes
exten = *411,3,Wait(1)
exten = *411,5,AGI(directory,general,ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS})
exten = *411,6,Playback(vm-goodbye)
exten = *411,7,HangupBest regards,Marco Mouta
Instead of call-limit=1 try o use incominglimit=1.
Note that this is not a fix, but more a workaround... I guess the
queue module still has allot of bugs... or it isn't sufficiently documented
to be correctly used.
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
Sorry for my delay writting here. My SIP.conf is similar of yours, i
only don't use qualify=yes, is it compulsory? I have 100 users and if
i activate qualify it will increase the traffic in my network no?
Best regards,
Marco Mouta
On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi
say me where I can find the Italian version of
IPswitchboard or if there is a way to translate the its messages?
Thanks in advance,
Marco.
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Password and username are ok.
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
But server B never registers in server A
I always get this:
Tx-Frame Retry[000
Just perfect! Thank you very much for your help so fast and fully explained!!!
BTW, I'm using TE110P --- Digium board :)
Best regards,
Marco Mouta
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Just because it's easier I'll do my rant up here. Don't over complicate
things when you're doing
post your iax.conf?
On 4/4/06, Marco Mouta [EMAIL PROTECTED] wrote:
Password and username are ok.
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
HI all,
My asterisk for all my users, everything was fine for 3 days, but now
i can't access it.
But it is running...
Could any one help me on this?
Best regards,
Marco Mouta
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Asterisk-Users
have my users in calls
Best regards,
Marco Mouta
On 4/5/06, Pimjai Wesnarat [EMAIL PROTECTED] wrote:
i used to have this problem. i solved it by recompiled it and change
modify the asterisk/Makefile by changing the ASTVARRUNDIR to something
like this.
ASTVARRUNDIR=$(INSTALL_PREFIX)/var
time yet to understand the safe_asterisk, if any one
could summarize it would be very good
Thanks,
Best regards,
Marco Mouta
On 4/5/06, Noah Miller [EMAIL PROTECTED] wrote:
Hi Marco
My asterisk for all my users, everything was fine for 3 days, but now
i can't access it.
But it is running
?
Best regards,
Marco Mouta
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]
CAUSE : Registration Refused
CAUSE CODE : 29
Any tip?
Best regards,
Marco Mouta
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Hi all,
I've around 10 people on my network getting this error,
Critical transaction failed: Client non-INVITE transaction[trying]: Time out
I'm using Asterisk 1.2.5 and Sjphone.
Any tips???
Best regards,
Marco Mouta
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