similar to: analog call progress - simplified I hope

Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. [callprogress] exten =>

Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its

Hi, We are having a problem with asterisk detecting that an analog ext has been put down. This seems only to happen after a number of calls have been made. We have an FXO port (TDM400P with FXO module) connected to our PBX and are using this to test asterisk prior to rolling our for our small office. What happens is that we make a number of calls to this ext which 1st rings a phone (FXS)

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'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

Hi All; Asterisk version is 1.6.1.8 Dahdi version is: dahdi-linux-complete-2.2.0.2+2.2.0 Since long time, and I am facing this problem and I did all the trouble shooting that I know without any success. The problem that while we are talking with someone through the FXO (connected to the PSTN analoge line), suddenly the call disconnect (without any specific time). I tried callprogress=no and

Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and

Hi all. I am looking for some command line programs (pre made) that will connect to an ldap server and list out the users in question provided by the search argument given. I found some mention of it from oracle but I did not see where they can be downloaded. Is something like this available and I just havent found them? Thanks, Jerry

downloaded and compiled today's CVS (04/08/2004) tried using callprogress on Via mini-itx (running RedHat Linux 9) if callprogress is set to yes on x100p, an i call the line connected to x100p, asterisk would execute the first app and will wait forever. anyone had success using callprogress? thanks. __________________________________ Do you Yahoo!? Yahoo! Small Business $15K Web Design

I have 2 Digium 4 port FXO cards in my system. The system is a P4 2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1 for storage - whitebox - running RedHat 9. With pretty much any CVS HEAD version we are getting, what I will call, "phantom" calls on some lines. What I mean by a phantom call is that the line will ring, Asterisk will log that the Zap channel has been

> -----Original Message----- > From: Jerry Geis [mailto:geisj@pagestation.com] > Sent: September 8, 2004 2:19 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] T100P calls with playback starts speaking > before pickup > > > Hi > > I am using a T100P connected to a panasonic phone switch using T1 and the > switch has 4 analog lines

I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is there any good tricks for a fast and accurate hangup detect in this situation?

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

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

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

Hi all, I was wondering if someone knows how to "automatically" login as root with gdm and gnome. I have been doing some searching, and havent been able to find a method. I dont want to login automatically all the time - its just part of install setup. 1) kickstart install 2) post section do somethings 3) setup so on reboot auto login as root 4) complete installing some things and

Here is some more stuff to add to the confusion about the "callprogress" option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively set "callprogress=yes" in zapata.conf. The results were typical of what many

I have a desktop unit with a sempron 2500+ to play with. doing "more /proc/cpuinfo" indicates cpu Mhz as1044 not the 2500 I am familiar with cpuspeed. doing ps ax | grep cpuspeed resulting in nothing. I did service cpuspeed start. No errors reported. ps ax | grep cpuspeed resulted in nothing. I was going to do "killall -SIGUSR1 cpuspeed" to attempt to get 2500+ cpu speed. I

We have the following situation: We're ordering two PSTN lines for our new office (no broadband at all -- it's not even available). The first line is going to be our primary # and it will also serve as our fax line using distinctive ring. The second line is going to be for a dial-up ISP (*sigh*) but now we'd like it to do double duty as the second line of a hunt group (we can't

Hello, I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP phones and will be receiving a machine containing a Dialogic card for a development project (in a nutshell, the card receives analog calls while the accompanying software handles automated prompts, etc). The Dialogic card is not SIP-based but will work with an analog line, so I'm looking into adapters that act