similar to: problem with polarity reverse

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

Hi all... I have a problem with caller id on my asterisk server. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id. My extensions.conf : [incoming1] exten =>

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Hi, I have both codec_g726.so and format_g726.so loaded: root at test:~# asterisk -r -x "module show" | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE ..... [SIP headers omitted] v=0

Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten => 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Hi, I am taking over an asterisk system from another person and having an issue where a sip trunk is restricting the outgoing codecs to just g.729 I am dialing in from a Cisco 7960. The Invite from the Cisco has the folowing M line: m=audio 17022 RTP/AVP 18 0 8 101. So it is allowing g.729, ulaw and alaw. Asterisk is tandeming the call out over a SIP trunk sip.conf tandem trunk config:

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

Hello all We are running an Asterisk server using 24 extensions with an AB1 channel bank and a T1 digium card, plus a TDM400 with 4 fxo modules (using only modules 2,3,4). We have been using the pbx with this configuration for some time (asterisk 1.0rc1,1.0rc2). A few weeks ago we have installed a mirrored disk (raid 1) in this server and we have been having some problems with sporadic

I'm using Asterisk@Home with the AMP interface and I'm having troubles getting incoming calls working properly. In AMP, I have it set to take incoming calls from PSTN, during regular business hours, to be sent to extension 201. The include statement for extentions-additional.conf is uncommented in extensions.conf; and I've verified AMP successfully changes the config files. However

---------- Forwarded message ---------- From: noreply at rubyforge.org <noreply at rubyforge.org> Date: 04-Jul-2007 19:21 Subject: [ mocha-Bugs-12001 ] Method call count is not reported correctly on error To: noreply at rubyforge.org Bugs item #12001, was opened at 2007-07-04 15:21 You can respond by visiting:

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

Is the new Isolinux series effecting hardware in a different way as 2.xx series? I'm getting a system crash in VMware. I guess this is a Vmware issue, but thought you might want to know. 2.xx series works OK for me. I get to see 8 dots when loading a 360KB imagefile, following by crash message mentioned below. Replacing MEMDISK by old version does not matter, only replacing isolinux.bin 3.05

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

I just installed my RoR application on my ISP''s server. This is a Cpanel environment and it creates a .htaccess file: RewriteEngine on RewriteCond %{HTTP_HOST} ^xyz.com$ [OR] RewriteCond %{HTTP_HOST} ^www.xyz.com$ RewriteRule ^(.*)$ "http\:\/\/127\.0\.0\.1\:12001\/$1" [P,L] in a public_html directory. (I changed my domain name in the text above to xyz because we are

Hello together, perhabs anyone of you, has an ideal, how i can do this: I have a matrix, like this one: [,1] [,2] [,3] [,4] abnr2 11425 11425 11555 11888 TIME 2 1 1 2 Cat 1 2 1

Hey guys, I need some help. Firstly: Running Rails 2.0.x cPanel Version 11.16.0-RELEASE Shared hosting. My App is not in public_html/ but under home/usr/rails_apps/(rubyapp) Please tell me how I get my rails app which is using the port eg http://www.domain.com:12001/ to be accessible without having to have users type the port, or even see the port ( the :12001 bit) Basically, in CPANEL,

I see MANY of these in my log files: [Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from '"202" <sip:202 at X:5060>' failed for '37.8.12.147:26832' - Wrong password [Jan 15 03:06:19] NOTICE[14129] chan_sip.c: Registration from '"5001" <sip:5001 at X:5060>' failed for '37.8.12.147:21268' - Wrong password [Jan 15 03:06:23]

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000