similar to: Problem solved on Xonox Asterisk distribution

Hi! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rog?rio __________________________________________ Microrede - Tecnologias de Informa??o, Ltd. http://www.microrede.pt *** ? There are only two types of people in the world, those who have lost data and those who will. ? -- Richard Nixon

Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

Hi, I've been loosing my mind with NAT and read that IAX doesn't have problems about nat. Does anyone knows about hadware (routers and etc) support IAX? Best regards helder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041230/f689bdde/attachment.htm

Hi! Is there a way to avoid being "at the middle" of communications between two SIP endpoints? So that we can avoid loosing bandwidth with it? Is there a way to "forward" the authentication to a IAX provider and "transfer" the call to it, avoiding using my own bandwidth? I've tested it with SER with some results, I was wondering if it is possible with

Norm, thanks for the help. Applying the fix solves the issue I mentioned but now I have more issues. I can install clang, but when running I cannot compile and link files. If I compile with -c flag it works but compiling the following x.c file gives an error: x.c: int main(void) { return 0; } > ./clang x.c /project/helder/scratch/packages2/bin/ld: unrecognized option '-C'

Hi! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

On Mon, Jun 24, 2013 at 6:17 PM, Jorge Rodrigues <skeept at gmail.com> wrote: > Norm, > > thanks for the help. Applying the fix solves the issue I mentioned but now I > have more issues. > > I can install clang, but when running I cannot compile and link files. > If I compile with -c flag it works but compiling the following x.c file > gives an error: >

Is there anything I can do regarding the linker issue? The solaris linker is in /usr/ccs/bin/ld but I think llvm wants to use the gnu linker. gcc in my system was compiled with the solaris linker. Thanks, Jorge On Mon, Jun 24, 2013 at 6:18 PM, Stefan Teleman <stefan.teleman at gmail.com>wrote: > On Mon, Jun 24, 2013 at 6:17 PM, Jorge Rodrigues <skeept at gmail.com> wrote: >

Hello! Im trying to mount the ADMIN$ share of my windows XP and i cant! I can mount other shares. I do ./smbclient -L 192.168.0.18 added interface ip=192.168.0.1 bcast=192.168.0.255 nmask=255.255.255.0 session request to 192.168.0.18 failed (Called name not present) session request to 192 failed (Called name not present) Password: Domain=[WORKGROUP] OS=[Windows 5.1] Server=[Windows 2000 LAN

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Hi guys, I''m just beginning to use RSpec and I ran into the issue of testing a named_scope. I''m not actually trying to test its behavior, as it''s not my code, but I wanted to test at least that it''s defined. I tried doing this: describe Post, ".most_recent" do it "should be defined" do Post.method_defined?(:most_recent).should be_true

Hi, I'm using Wine to run on linux, an application build with vc++ 6.0. I've got no problems running it, except when i try to create a child process to call a a different executable that performs tasks needed to the main process, I get no error or system failure, the process simply fails to be created.. The code i've done to do this is shown below. Do i need to invoke wine on the

Hi, This is a thank you message for all that helped me including Max from www.asterisk-support.ru with whishes of a Happy New Year. Althought I still have a problem I'm happier I've 50% of my task complete. I'm using two TA from Draytek (router 2600V / router 2500V) 3 ADSL lines (2 for TA 1 for ASTERISK with a Draytek 2500 no voice model). If I call from one to another (using

I'm sorry as Patrick said I've made a post 4 times but didn't realize it until I've started received the messages back. The mail client hung (apparently) but it was in fact sending the msgs. I know probably some/all of you (and the owners of the list at Digium) were not very pleased for receiving such a bunch of copies of the same email, however it was sent in error and apologies

Hi, Is there some parameter that I should pay attention to when using externip parameter on sip.conf? I ask this because after using sipsak I've noticed maybe the reason that i don't get voice in one direction could be because at the SDP/SIP messages are references to the 192.168.1.50 (server internal ip) and not the ip from the provider. Is there a way to alter SDP messages to correct

Hi! There's any way to set up a call using G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it to PSTN broadband termination? I've put the following in sip.conf: disalow=all allow=gsm allow=g726 (my TAs use G726 32K) best regards, Helder -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am a developer from Oracle. I wanted to create a png file out of png device with a bit depth of 8. I am using R on linux. But I noticed that R automatically switches between 8 and 24 bit depth. i tried a lot of things from your grDevices package but unable to control the bitdepth. I would be really grateful to you guys if you can suggest me some way to solve this issue. -/Ag Anshul Gupta

I am getting the following from both of my win2k machines in the /var/samba/log/log.%m. [2001/11/09 08:21:39, 0] lib/util_sock.c:read_socket_data(478) read_socket_data: recv failure for 4. Error = Connection reset by peer [2001/11/09 08:21:39, 0] lib/util_sock.c:get_socket_addr(1038) getpeername failed. Error was Transport endpoint is not connected [2001/11/09 08:21:39, 0]

This is for the record, thanks for your patience. > Gerald (Jerry) Carter wrote: > >> >> Peter Kruse wrote: >> | >> | Say, I create a "distribution group" on Windows ADS named >> | "distgroup" add as a member a security group named "secgroup" with a >> | user "robert" in it. Then when I look at the groups

Hello, My first PDF report from rails. When i try to render "tab.render_on(pdf)", I get that error message. Thanks any help, Helder ================================================================= TypeError in ContasController#plano singleton can''t be dumped RAILS_ROOT: script/../config/.. Application Trace | Framework Trace | Full Trace