search for: recives

Hi! How can I setup my Chan Dongle recived calls in my Asterisk? I have to setup in dongle.conf ? Or in extensions.conf? And the code for recive I found this site http://asterisk-service.com/page/chan-dongle-use I have to To save Subscriber Number before? See the error log in my Asterisk pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-0100000000' sent to invalid extension but no

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

http://bugs.freedesktop.org/show_bug.cgi?id=11492 riccardo at datahost.it changed: What |Removed |Added ---------------------------------------------------------------------------- QAContact|eric at anholt.net |swfdec at lists.freedesktop.org Status|NEEDINFO |NEW AssignedTo|eric at anholt.net

hi, mount -t ocfs2 -o datavolume,nointr -L "oracrsfile" /u02 when i mount linux partation using above command i recieve the following error mount.ocfs2: Invalid argument while mounting /dev/sdd1 on /u02. Check 'dmesg' for more information on this error. thanks in advance zeeshan

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

hi, i am new in ruby, i have to do the next Send a hash (key are strings and values are booleans) to the view, show it like check box in the view, using string like nome and the value like check or not checked, and recover this hash modified to the controller for work with it so i wrote this in the view <%@arquitectures.each do |key, value| %> <%=key.to_s%> <%=check_box(key,

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

Hi, all I want to configure a few FXS ports in an Antribank-16 to be able to receive faxes sent throught a PRI: E1 ==>Zap * ==>FXS * ==>Fax machine My asterisk box has a Digium TE120P (for the PRI). Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5 The Astribank is not configured yet, because I am a little bit confused about how to do it. Let's say I configure

I recived this: logonCount: 0 sAMAccountName: Guest sAMAccountType: 805306368 objectCategory: CN=Person,CN=Schema,CN=Configuration,DC=hebe,DC=us isCriticalSystemObject: TRUE memberOf: CN=Guests,CN=Builtin,DC=hebe,DC=us

Salve, Imagine ogg vorbis is used to produce radio with free software. A journalist would produce a report end send it with 24kBit/s out from a cricis place somewhere in the world. DRM is going to use MP4 - so his report has to be reconverted with loosing quality :-( Can you imagine to have an free codec someday that would work in embedded radio-reciver like MP4? If yes, should DRM not be open

I have a customer that is complaining that any call coming in from Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel 1.4.3 and all the MFC/R2 patches and libraries. All other calls go out and come in, just Nextel seems to have this problem. The phone company technician connected a PBX emulator on the line and that one could receive the calls from Nextel. The E1 is provided

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

On 02/12/14 19:51, steve wrote: > On 02/12/14 20:30, jacek burghardt wrote: >> I recived this: >> logonCount: 0 >> sAMAccountName: Guest >> sAMAccountType: 805306368 >> objectCategory: CN=Person,CN=Schema,CN=Configuration,DC=hebe,DC=us >> isCriticalSystemObject: TRUE >> memberOf: CN=Guests,CN=Builtin,DC=hebe,DC=us >> > Next step. Add: >

...t i = 0; i < data_in_length; i++ ) { output[i+offset] = out_put[i]; } offset+=FRAME_SIZE; } /// change the floats to shorts so it can be used by the sound out put. speex_bits_reset( bits ); } my main loop just caputures and sends and then recives with a wait function/poll of the open socket. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060802/f8d9703b/attachment.htm

hi! im implementing a login system for a messageboard. tooked from a book. pretty simple. everythings works great, but in the example the idea is, when i want to create a new message, the author name shall be tooked from the session[:user] def create params[:message][:date] = Time.now params[:message][:author_id] = @session[:user].id #here!! @message =

Dear Friends I use R in linux. I have problem by installing a package. when i use "R CMD" coomond, i recived a d a erroe line " /usr/lib/R/bin/Rcmd: line45: exec: INSTAL~1 not found. how can i solve my problem? thank you for your advise --------------------------------- [[alternative HTML version deleted]]

R users   Thanks in advance:   I need to use the function "mle.cp" in R.  When I run this function I recive   result <- mle.cp(y.hald~x.hald) Error: could not find function "mle.cp"  could some one  give me a help?   thanks [[alternative HTML version deleted]]

Am I doing something wrong here? GOt a SPC-841 the other day and have it registering properly. Can place and recive calls as expected but when on the phone, a second call is immediately dumped to "busy" voicemail. Does this thing not support call-waiting? Or, have I just got my configs wrong? Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

Hello, Sorry, if you have recived this message already, but I have problem with my email service. Could you tell am I right in case: I am not able to add other users/groups than unix's user: owner,owner's group and others to folders/files localized on Samba server, despite doing that as root from XP client station. I also work without nt acl support = off. This is normal? I am