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Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is

Hi, I notice that SJPhone is registering to asterisk with an expires of 120 secs. However, when I invoke the command "sip show peer [sip id]". I notice that the output indicates the expires 427 and the expiry is 900. Can someone explain these numbers to me? I also notice that just before SJPhone re-register, when I try to make a call to the SJPhone, asterisk will complain that

...sterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ------------------------------ > > Message: 6 > Date: Wed, 01 Dec 2004 22:36:22 +0800 > From: "HengWee Chin" <chinhw@hotmail.com> > Subject: [Asterisk-Users] SIP expiry time > To: asterisk-users@lists.digium.com > Message-ID: <BAY22-F3092A724859DC5FD7BF215DCBF0@phx.gbl> > Content-Type: text/plain; format=flowed > > Hi, > > I notice that SJPhone is registering to asterisk with an...

>Hi everyone, I'm a newbie to Asterisk. Will Asterisk run on RH9, easily >or >does it have to run on FreeBSD? Will the drivers for the Digium cards >work >on RH9? I am running Asterisk on RH9. I did not encounter any problems with both Asterisk and the drivers for the Digium card. _________________________________________________________________ Keep track of Singapore

Hi, I have a problem with the queue cmd. I am trying to redirect an incoming call to another phone when nobody in the queue answer it within 18 seconds. Somehow the incoming call keeps on retrying within the queue. The second part was never executed. Below is a part of my extensions.conf and the queue.conf. Is there someone out there who can help me? Thanks in advance. Regards,

Hi all, I have the following setup PSTN -> ASTERISK -> IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller