search for: ppers

Hi Guys, is it possible to use more than one host with the pgsql backend? Like this: connect = host=10.5.29.77 host=10.5.29.210 dbname=postfix [...] I read that this is possible with mysql but I want to use pgsql. Regards Patrick

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Hi List, we are using openh323 gatekeeper for voip telefony. We also have a voip over ss7 TELES Switch for voip into POSTN Network. Know we want to use Asterisk for converting SIP to h323. Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do we need openh323 GK for astrerisk, too?. And how can i tell asterisk to sent all none SIP-ip calls to the gatekeeper over h323? thx

I have been trying to compile Samba 3.0.4 on a SunFire v880 running Solaris 9 but "make" generates a fatal error. My goal is to get Samba compiled with the "--with-ldapsam" option so that I can use SunOne Directory Server 5.2 as the back-end repository for user authentication. I have gcc version 3.3.2 installed in /usr/local/bin/gcc which I have used to successfully compile

with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial("H323/ip$10.0.0.124:49638/18690", "SIP/0699073201") in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered

hi, i am using Asterisk CVS-05/31/04. i have the problem that sip clients can make calls over asterisk without registering befor. the xlite is not loged in with any username/secret bit still can make calls over asterisk. how can that be? thx for help. thomas

Dear users I am reposting this and hope it will be accepted this time. I am using copula package to fit my bivariate data and simulation. As explained in package documentation we can use our own data distribution to feed on copula as long as we have d, p and q (pdf, cdf and quantile) functions are available. Hence my code for those are: # Make the functions for data distribution

Hi ! I have a small problem. When switching a call (pstn -> sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn ------ Asterisk ------ sip phone Regards, Dave

Hi - I'm trying to get wireless running on CentOS 4.7 on a dual core laptop (latitude-e4500) for an employee. I'm having trouble with building dkms-ndiswraper-1.54-1.el4.rf - enclosed are the errors messages. The kernel was rebuilt to disable CONFIG_4KSTACKS. The errors prior to rebuilding the kernel are identical to the errors after rebuilding the kernel and rebooting - minus the error

Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers Now to Antony questions: On 06/06/2017 04:36 PM, Antony Stone wrote: > On Tuesday 06 June 2017 15:18:32 andre castro wrote: > >> I just installed asterisk in a debian server. >> All seems to be running fine, but the audio sent by the server. > >> But I hear nothing at...

any ideas why it won't compile? running /root/samba-3.0.28a/packaging/RHEL/makerpms.sh eventually it gets to: ./autogen.sh: running script/mkversion.sh ./script/mkversion.sh: 'include/version.h' created for Samba("3.0.28a") ./autogen.sh: running autoheader -I. -Ilib/replace ./autogen.sh: running autoconf -I. -Ilib/replace Now run ./configure and then make. +

Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks Neil -------------- next part -------------- An

hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints: Peer User/ANR Call ID

Dear friends of R, I want to produce a pdf file with the contents of a matrix. I employ the latex command in combination with dvi, both contained in the Hmisc package. It seems to me that the function does not correctly set the directory. > tbl.loc <- matrix(1:4, nc=2) > latex.obj <- latex(tbl.loc) > dvi(latex.obj) warning: extra args ignored after 'cd' H:\PROJECTS\data