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 executes the command there as if nobody answered the call. I need this because I want to have a follow-me application that dials different phone numbers or extensions based on the call status. Thank you in advance for your help. Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490
Hi All, I am using Asterisk 1.0.7 with an X101P analog card which is connected to an Alcatel pbx. My problem is that when I place outbound calls on the zap channel, Asterisk returns a connect event as soon as the phone starts ringing. This means that Asterisk is not being able to do Call Progress analysis on the zap channels. I have tried setting 'callprogress=yes' in zapata.conf but it made no difference. This problem is not there with SIP and IAX channels. Here's my zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; default context=default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes busydetect=yes busycount=6 callprogress=yes progzone=uk group=1 callgroup=1 pickupgroup=1 immediate=no ; define channels signalling=fxs_ks channel => 1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051125/48893937/attachment.htm
Nitin Joshi wrote:> Hi All, > I am using Asterisk 1.0.7 with an X101P analog card which is connectedto an> Alcatel pbx. My problem is that when I place outbound calls on the zap > channel, Asterisk returns a connect event as soon as the phone start > ringing. This means that Asterisk is not being able to do Call Progress > analysis on the zap channels. I have tried setting 'callprogress=yes' in > zapata.conf but it made no difference. This problem is not there with SIP > and IAX channels.I have the same problem with Digium TDM cards. I've been doing pretty extensive research and found no solution. Look at my mail [Asterisk-Users] [Fwd: call status with FXO], few mails ahead of yours.