search for: isamarmaia

Hi all. I've been lurking here for a couple weeks just trying to get an idea of how to install asterisk. I'm running Debian with a custom kernel version 2.4.18. I think I've got all the dependencies installed - debian packages readline4 and openssl are installed. It's a very barebones install with only the packages relevant to building a kernel and asterisk installed. The

I have a Quicknet Linejack. Actually, I want to make a standalone(no internet connections) call to my asterisk box running GNU/Linux. When I call the system it says: -- Executing BackGround("Phone/phone0", "demo-congrats") in new stack NOTICE[14349]: File channel.c, Line 1212 (ast_set_write_format): Unable to find a path from 2 to 1 WARNING[14349]: File file.c, Line 553

Woooow! Kick this guy out from this list and pls filter *@uol.com.br It's very annoying.... Isamar ---------- Forwarded message ---------- Date: Fri, 26 Sep 2003 08:48:45 -0300 (BRT) From: AntiSpam UOL <andersoncbr.sspam@uol.com.br> To: isamar@isamarmaia.org Subject: RE:RE: [Asterisk-Users] G729 experiences.. [antispam_txt.gif] Ol=E1, Voc=EA enviou uma mensagem para andersoncbr@uol.com.br Para que sua mensagem seja encaminhada, por favor, clique aqui Esta confirma=E7=E3o =E9 necess=E1ria porque andersoncbr@uol.com.br usa o Antispam UOL, um pro...

I get this a lot. It eventually succeeds in registering, but can take a minute or two. Michael -----Original Message----- From: Isamar Maia <isamar@isamarmaia.org> Sent: Dec 21, 2003 10:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FWD / Timed out Hi Folks, I can ping fwd.pulver.com with no problem but not getting to register with them... NOTICE[4101]: File chan_sip.c, Line 2476 (sip_reg_timeout): Registration for '73921@...

...and then include it at the very end of the [default] context (or wherever you want to use it). That would then pick up anything that drops through. If you do it any other way it will get sorted to the top and you will have trouble! Peter -----Original Message----- From: Isamar Maia [mailto:isamar@isamarmaia.org] Sent: 29 June 2004 15:54 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Playing the invalid extension input I'm trying to do the following: exten => i,1,Saydigits(${EXTEN}) My intention is to play the invalid input to the user, but it doesn't work. Any suggesti...

...on ebay! Whilst the digium card requires more from the host processor and may not be approved in as many countries as the equivelent dialogic, I think that for most cases the advice was sound. Actually, I kept the dialogic card, but that was for support purposes. Tim. Isamar Maia <isamar@isamarmaia.org> wrote: __________ > >I'm planning to buy Dialogic licenses for one of my dialogic boards to use >with *. I have already that in the drawer and it's boring me to keep it >there with no use. >Although, I have heard that it doesn't work for dialout and I would like &...

...ionists with a penalty of 1 and us propeller heads in technical support with a penalty of 2). The technical support people would only be offered a call from the sales queue if all the sales people and the receptionists were busy. Steve -----Original Message----- From: Isamar Maia [mailto:isamar@isamarmaia.org] Sent: 04 July 2004 10:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Penalty in queues.conf I have already read explanation about that in some places but I don't have still a clear image about the meaning of Penalty parameter inside of queues.conf What means that? Th...

Hi FOlks, I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar

I was told the X100P will have issues if installed on a line with a DSL connection. Is there a card that will work correctly on a DSL connection? Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/6de6226b/attachment.htm

I have a setup with a polycom 601 and an act p160s. All on local segment, no NAT. Can call the act p160s, from the polycom, rings, connects, and a conversation can take place. The reverse is not true, Dialing from the act to the polycom does not work. SIP debug shows, at the end, "Incoming call: got sip response 416 "unsupported URI Scheme" back from 192.168.0.xxx. Which is

Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP:

I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2

I am trying to use spandsp 0.0.1k-whole. I have 2 X100P working well for inbound/outbound calls. I have tried libtiff 3.6.1 and 3.5.7 With 3.6.1 I get only faxes all black, and 3.5.7, I get blank vertical lines and the rest all black also. During the transmittion, apparently there is nothing that can indicate any error except some "Training errors". The sender fax thinks that the fax

Dear All, I need to bind two different ports at the same time for SIP. 5060 and another port number. Is it possible ? It would be something like port=5060,5062 Isamar

Hi FOlks, I am using for research purposes Kvin's codecs available at http://kvin.lv/pub/Linux/Asterisk/ G729 is working very well but g723 has a very poor audio quality. I recompiled everything with gcc4 and the distro used is Slackware 11. Anyone with some experience on that? Thanks in advance for any help. Isamar Maia +55-71-9146-8575