search for: mouta

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope...

...very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call This is timeout after pressing the first digit isn't it? -- best regards, Marco Mouta

...acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de cada pa?s, voice prompts, etc. Se houver um n?mero minimo de pessoas interessadas, podemos avan?ar com a ideia. -- Com os melhores cumprimentos, Marco Mouta

..."A" Zaptel call 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060511/a26cdbe8/attachment.htm

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

...y for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any tutorial? Probably someone around the world as already done this before. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente...

...sic on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 Everytime i press the music 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060418/b18b206a/attachment.htm

...t changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? -- Best regards Marco Mouta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061010/c5a322ff/attachment.htm

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

...gistred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

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

...ntly i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that Ast@home is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it... Why is this 5 seconnds? any one knows? best regards, Marco Mouta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060412/98ff6d3c/attachment.htm

...that in /usr/lib/asterisk/modules/ doesn't have cdr_addon_mysql.so even after compiling Asterisk Addons! In fact the cdr_addon_mysql.c exists, but it doesn't seems to be compile when i run Asterisk-Addons: make && make install Any one can help me on this? -- best regards Marco Mouta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061011/3a306fca/attachment.htm

...so I notice, with SIP debug peer 4214 on * CLI , that when i dial from my sip phone 4XXX numbers, nothing seems to reach the asterisk Server! I hope someone can point me out where is the problem! This server has only sip extensions. P4 - 1G RAM wiht TE110P with weekly reboot. Best regards, Marco Mouta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070111/7c407c44/attachment.htm

Hi I'm very new to Asterisk and this is my first posting to this mailing list. I got a Asterisk@home V2.8 working, and now I'm trying to use Asterisk.NET (http://sourceforge.net/projects/asterisk-dotnet) to get call events to my C# program. Asterisk.NET comes with a sample program called Asterisk.NET.Test and it uses the following default constants for login: const

...it. This way I've tested and seems to work fine in my Virtual Machine lab. The only issue i found was one module that is loaded with my modules.confthat I needed to copy from the backup /usr/lib/asterisk/modules and give the right permissions. Am I missing something? best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente...

Hi i use now iaxmodem for receive fax and that's work very good with Hylafax ;=) Do you know if we can sent fax using iaxmodem and Hylafax ? when i test: d?c 13 13:47:21.12: [13725]: SESSION BEGIN 000000014 330426690268 d?c 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0 d?c 13 13:47:21.12: [13725]: SEND FAX: JOB 2 DEST 0426690268 COMMID 000000014 DEVICE '/dev/iaxmodem1' FROM

...I get this error message on my Check Point Firewall: "sip reason:Attack Info - Malformed SIP datagram, OPTION message is out of State" By the way i've one client that is running all ok, the others all have this problem. I hope some one could help me with this. Best regards, Marco Mouta

...remove the entries or files yourself. AAH should do this as part of the script. But it does not, would probably cause more harm than good. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Marco Mouta > Sent: Tuesday, March 07, 2006 10:02 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Destroying a SIP extension doesn't > destroyvoicemail box?is this a bug? > > Hi all, > > I'm using Asterisk@home 2.5, and i've don...

...ll, I've bought a TE110P, and received it today. So i decided to install Asterisk@home 2.7 with this card. In the past i had experiencies with X100P (clone card) and it never take me so long to reboot the machine.... Machine: P4- 2,8Ghz 1GRAM TE110P What could be wrong? Best regards, Marco Mouta