similar to: Asterisk FAx-to-Email

Hi, I have read a lot about the thread of faxing support in Asterisk as well as spanDSP. However, either I don't fully understand other people's applications or may be what I'm trying to do is different from what others are trying to do. I have a very simple setup. I have an asterisk server with a TE110P connected to the PSTN via T1 PRI (Asterisk A). I have another asterisk

Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb

Hello, I have installed FreeRadius server on Trixbox Server. My problem is mysql is not letting FreeRadius to login either locally or remotely. I also insert proper entries in HOST and USERS tables. But it does not work I always get ERROR 1045 (28000); Access Denied for user 'root'@'localhost' Thanks Wazb

Hi, I have Asterisk 12.7.1 installed through Asterisk@Home CD. and explicitly I have installed UnixODBC and FREETDS in order to access MS SQL 2000 Database which in on Windows 2003 Server on remote location. I tested connectivity through isql and tsql, both utilities are working fine. I need to access MS SQL 2000 Database through PHP. When I tired to check the connectivity through a Test PHP

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.

hello to all, i am geting this NOTICE while i am running asterisk. agents are able to here the customer voice but the customer is unable to here agent voice plz somebody help me #rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 -- M. VIDYASAGAR -------------- next part -------------- An HTML

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with "488 not acceptable here". I double check t38pt_udptl = yes in

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hello, I have the fowling scenario: fxs[asterisk1]-----iax-----[asterisk2]e1----e&m---PSTN I want to know the steps to transmit fax from a machine connected to the fxs to a fax machine on the PSTN. The same for dial-up's. Is it possible only with a/ulaw ? What configs I need in asterisk1? Thanks in advance Eduardo

I have been searching all over for the answer on all sources online and have come to the conclusion that it must be rudimentary or I am asking the wrong question. I cannot figure out how to configure the box to set the "from" address to a correct domain, as my outgoing isp will not pass mail from root@asterisk1.local, as I expect it wouldn't. Any help is appreciated, even just

hi, when VAD is enabled, can i make the decoder simply produce comfort noise in the event that no voice was detected? i'm working on a p2p voice app. when no voice is detected, i'm thinking that i can make the transmiting endpoint send a signal to notify the remote endpoint that VAD is in effect, instead of having to send the whole packet that doesn't have voice anyway. on the

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

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

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN

Hi all, We are an ISP in Switzerland and we propose VoIP with Asterisk. Everything works perfectly for all clients but one. In a conversation, they have no sound during 2 to 8 seconds using the G729 codec (I didn't make the test with G711). The Client configuration is perfect (QoS and bandwidth management). Do you know some issues with the G729 codec? Thanks a lot for your comments, Thomas

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

On Tue, 2003-04-15 at 11:31, Jean-Marc Valin wrote: > > How do i detect whether there is silence in media using speex? > > Is there any API which decides that the audio data only contains > > silence? > > Basically i will have PCM linear data, I want to know whether it is > > complete silence. > > Well, the best way is probably to turn VAD *and*