similar to: DAHDI and hangup issue when playing the IVR

Hi all, I have installed asterisk latest stable version 1.6.1.0, with dahdi driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now it wont work with 1.6. I managed to register my phone on asterisk. but i cant hear any dial tone on my phone. these are my configs. it will detect incoming calls and transfer the call to ext 312. but sip phone users voice is not clear..., but

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hi list, I have successfully authenticated active directory users with samba. Now I need to create some Active directory security groups and authenticate and redirect those users to a specific directory. Ex: IT_GROUP - user x , user y FIN_group - user a, user b If the user x , access the samba server, that user will be redirected to the specific directory (that's in the samba stanza). This

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

I have a PRI line and I am having problems setting the ringtimeout on the dial application to more than 29. If I set ringtimeout to 29 on the dial application call and I do not answer the ringing phone then I correctly get DIALSTATUS set to NOANSWER. If I set ringtimeout to any value over 29 on the dial application call and I do not answer the ringing phone then I go to extension h and have

Hi I have this problem: <script> var param1 = ''field3''; </script> <script> var param2 = ''field3''; </script> <script> var param3 = ''field3''; </script> <%= submit_tag ''Next'', {:onsubmit => ''return validate_options(param1,param2,param3);''} %> function

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

Hi all, Iam using asterik 1.4.8 and connected to google talk. When iam calling from my google talk account to sip phone i can hear the voice (2 way). (this happens only within the LAN). when my friend tries to call my asterisk server (connects to the public ip) using his googletalk client it comes to my sip phone but either party cant hear a voice. I have fully allowd both tcp,udp on my

Hi guys, I have installed samba with AD authentication. Ntlm_auth is working without any issue with the domain. But if I connect using my windows pc, to the samba share, it gives following error. Wbinfo -u / wbinfo -g giving the correct output. And ntlm_auth also working without any issue. If I try to connect from my windows PC to the samba share it gives following error. [2010/12/01

Hi all, OK, so this at first looked like a clear cut "Don''t do it, or at worst handle the results" issue my customer has come to me with, but the more we discuss it, the more it looks like we should have better ways of dealing with this issue. > We have user defined dtrace probe points in the application which use > as parameter 64 bit values: > > provider adv {

Hi there, I would like to know if it is possible to modify a text file with a R function. In fact I would like to know if a function "Search & Replace" exists. My problem is to create config files from a Base file in which I have to modify values of parameters. My Base File: #... #... Param1= V1_1 #... Param2 = V2_1 Param3 = V3_1 #... What I would like for each created file

Hello, I'm using the "sem" package to do a confirmatory factor analysis on data collected with a questionnaire. In the model, there is a unique factor G and 23 items. I would like to calculate the standardized residual variance of the observed variables. "Sem" only gives the residual variance with the "summary" function, or the standardized loadings with the

PLEASE REMOVE ME FROM THIS LIST SERVE IMMEDIATELY!!!!!! On Wed, Dec 12, 2012 at 6:41 PM, dada <paxkn@nottingham.ac.uk> wrote: > Hi > I would like to do neural netowrk analysis on my data. It look like this: > > drug param1 param2 param3 param4 param5 class > A 111 15 125 40 0.5 1 > B 347 13 280 55 3 2 >

hi, I''m trying to create meaningful urls with routing extracting all params as specified: map.testroute ":param1/:param2-for-:param3.html" However this approach totally fails - it says Routing Error, No route matches "/aaaa/bbbb-for-ccc.html" with {:method=>:get} Then while browsing source code and some googling I''ve found that there is sth called

Dear R-users, I am a dentist (so forgive me if my question looks stupid) and came across a problem when I did simulations to compare a few single level and two level regressions. The simulations were interrupted and an error message came out like 'Error in MEestimate(lmeSt, grps) : Singularity in backsolve at level 0, block 1'. My collegue suggested that this might be due to my codes

I have the following STI table: def self.up create_table :distributions do |t| t.string :type t.integer :simulation_id t.string :dist_name t.string :desc, :default=> ''fixed'' t.float :param1, :default => 5.0 t.float :param2, :param3 t.timestamps #fields for RscDist t.integer :resource_id #fields for LoadDist

Hi all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf