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Marco Mouta
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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,
--
Best regards,
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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[EMAIL PROTECTED] wrote:
Hi Marco,
Marco Mouta wrote:
Please feel free to contact me if you have more ideas to improve this
solution, currently i didn't test more than one simultaneous calls
incoming and outgoing through Skype.
get it running on unix so you can run it on the asterisk
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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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
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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
My guess would be to check your manager.conf[admin]secret = amp111deny=0.0.0.0/0.0.0.0permit=10.0.0.1/255.255.255.0read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,userThe line permit=10.0.0.1/255.255.255.0 should be adjust to your network
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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=Asterisk+config+features.confThank you very much!On 5/11/06, Marco Mouta
[EMAIL PROTECTED] wrote: Hi all, I've the current scenario: User A - Zaptel call incoming in my Asterisk to my SIP user B.
B gets the Call. A says : B i would like to call PSTN user C B places a call to user Cand asks if C
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
http://www.voip-info.org/wiki-Asterisk+config+zapata.confI've made some tests using this in Portugal and seems to work:---
switchtype=qsig ; you may try this in your
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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Hi all,My architecture is:PSTN-E1OldPBXE1-AsteriskI've a similar problem, SIP user agents using X-Lite:Sip User Agent A calls to PSTN user BB user hangs the call
A user starts listening busy indications on the phone, and if he doesn't hangup correctly on Xlite The calls seems to be
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
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
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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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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, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best regards,
Marco Mouta
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system does this PC have and is it up to date with security and bug patches.
Thanks
Marco Mouta wrote:
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best
the
users:)
Best regards,
Marco Mouta
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Hi all,
I've created this test.call file and it is not running outgoing call files:
i've made mv test.call /var/spool/asterisk/outgoing and nothing happens
Channel: SIP/200
MaxRetries: 3
RetryTime: 40
WaitTime: 25
Context: from-internal
Extension: 200
Priority: 1
My asterisk is running with
call on ZAP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 2)
-- Attempting call on ZAP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
Do I need to define context to outbound calls through my ZAP ?
Thanks in advance,
Marco Mouta
On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote:
I
it's working , the problem was:
Channel: ZAP/g1X
I changed to ZAP/g1/X
And it's working fine!
Thank you all
On 3/28/06, Marco Mouta [EMAIL PROTECTED] wrote:
Thank you for your fast reply!!!
It's working on for SIP:)
I've tried to my zapata and doesn't make the call, i get
In fact i've never done it. And i don't have any Cisco Phone...
If i find something i will report it here :)
Best regards,
Marco Mouta
On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote:
heh.. I just noticed that ;)
Heh, do you know maybe how to update time/date on all Cisco 7905 phones
I'm a bit newbie, could you tell me how to i apply the patch?
Thanks in advance
Marco Mouta
On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
http
it helps.
Best regards,
Marco Mouta
On 3/22/06, Charles Marcus [EMAIL PROTECTED] wrote:
C F wrote:
Polycoms are not the best if you want a phone that works behind NAT.
Do you mean in general? Or only if you are trying to interconnect
multiple offices?
Are Polycoms fine for just one office
,
Marco Mouta
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entered 0 (string) which is empty()
$agi-stream_file(dir-nomatch);
} // else, we timed out
Probably it's because i'm newbie, but is it correct 3 equals? ($digits
=== 0)) ?
Best regards,
Marco Mouta
regards,
Marco Mouta
On 3/10/06, Olle E Johansson [EMAIL PROTECTED] wrote:
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk development, things
that will change Asterisk and
make it work better in your organisation, make it easier to sell in
your area
suggestions?
Best regards,
Marco Mouta
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from users with
problems ( in fact i didn't find any sucessfull feedback).
I'm a bit afraid of doing all the tutorial and get in troubles with my
stable asterisks
Any one has tried it?
Best regards,
Marco Mouta
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Could it be Call Waiting Deactived?
On 3/7/06, Rolf Brusletto [EMAIL PROTECTED] wrote:
All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out,
this to a real Linux System.
Does any one could help me understanding what is going on?
Best regards,
Marco Mouta
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entering two diferent days on my conference room... Also I don't think
it is a good choice to contact Administrator to change my Meetme
password.
Best regards,
Marco Mouta
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connection 4Mbit.
I hope this Excellent mailing list could help me on giving me some
Feedback and or advices/tips.
Best regards,
Marco Mouta
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,
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On 1/18/06, yrving rivas [EMAIL PROTECTED] wrote:
Ok, thanks, it works for me.
Regards,
Yrving
Dovid Bender [EMAIL PROTECTED] escribió:
If you are new I would reccomend using [EMAIL PROTECTED]
http://asteriskathome.soundforge.net . It is a great
resource for beginers. Also
? Am I doing something wrong?
Best regards,
Marco Mouta
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Hi all,
I'm going to buy E1 digium110P ,any one knows how i can get faxtomail
working for three different channels?
I mean:
channel1--[EMAIL PROTECTED]
channel2--[EMAIL PROTECTED]
for 1 channel is not dificult [EMAIL PROTECTED] and NVfaxdetect
Using [EMAIL PROTECTED] 2.5 faxToPDFmail works,
regards,
Marco Mouta
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Hi i'm developing a solution with ASterisk, but in fact i don't know
which ATA SIP device should buy.
Could you give me some advices?
Marco Mouta
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-- Forwarded message --
From: Marco Mouta [EMAIL PROTECTED]
Date: Feb 15, 2006 1:58 PM
Subject: Which ATA device do you recommend?
To: [EMAIL PROTECTED]
Hello,
I'm developing a Voip Solution for a client, which ATA SIP do you
recommend? there are some ATA devices fully tested
regards,
Marco Mouta
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It seems to me, that your problem is that X100P is not detecting that
the caller has hangup through PSTN.
I really got lots of problems with disconnect detection, and currently
i only get it working on asterisk @home 1.5 , it doesn't work well on
later releases.
The main changes i've made in
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