Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source (after removing the Ubuntu packages) have installed the following: asterisk-1.8.0 dahdi-linux-complete-2.4.0+2.4.0 libpri-1.4.11.5 I copied my config back into place and most seems to work, but I cannot get my phone that is plugged into the Wildcard TDM400P REV E/F card that I have to work. Basically, I don't hear the dial tone and Asterisk does not register off hook events. I have spent time reviewing my config but I don't see what the issue is. Is there anything I am missing, or can you suggest some additional things to look at? Tim chan_dahdi.conf grep -v "^;" /etc/asterisk/chan_dahdi.conf | grep -v "^$" [trunkgroups] [channels] language=en context=phones signalling=fxo_ks usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 dahdi-channels.conf: ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXOKS" signalling=fxo_ks callerid="Channel 1" <4001> mailbox=4001 group=5 context=phones channel => 1 calleridmailboxgroupcontext=default ;;; line="2 WCTDM/4/1 FXSKS" signalling=fxs_ks callerid=asreceived group=0 context=incoming-local channel => 2 calleridgroupcontext=default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101207/5a6b1dfd/attachment.htm
Timothy Legge wrote:> Hi > > I was using the delivered Ubuntu 1.6.x packages but I wanted to look > at gtalk integration so I downloaded, compiled and installed the > source (after removing the Ubuntu packages) have installed the following: > > asterisk-1.8.0 > dahdi-linux-complete-2.4.0+2.4.0 > libpri-1.4.11.5 > > I copied my config back into place and most seems to work, but I > cannot get my phone that is plugged into the Wildcard TDM400P REV E/F > card that I have to work. > > Basically, I don't hear the dial tone and Asterisk does not register > off hook events. I have spent time reviewing my config but I don't > see what the issue is. > > Is there anything I am missing, or can you suggest some additional > things to look at? > > Tim > > chan_dahdi.conf > grep -v "^;" /etc/asterisk/chan_dahdi.conf | grep -v "^$" > > [trunkgroups] > [channels] > language=en > context=phones > signalling=fxo_ks > usecallerid=yes > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > callreturn=yes > echocancel=no > echocancelwhenbridged=no > group=1 > callgroup=1 > pickupgroup=1 > > dahdi-channels.conf: > > ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) > ;;; line="1 WCTDM/4/0 FXOKS" > signalling=fxo_ks > callerid="Channel 1" <4001> > mailbox=4001 > group=5 > context=phones > channel => 1 > callerid> mailbox> group> context=default > > ;;; line="2 WCTDM/4/1 FXSKS" > signalling=fxs_ks > callerid=asreceived > group=0 > context=incoming-local > channel => 2 > callerid> group> context=default >I have no experience with 1.8, but unless things have changed channel= has to be the last line in a section. the remaining lines are ignored Don't you also need [line1] at the beginning of each section? using context=default has been a security issue in the past. Using a different context, and having the default context point to nothing more than a rude recording may save you in the case of a security breach John Novack -- Dog is my Co-pilot
Do you have any issues with getting audio to bridge? I am using 1.8 also. On Tue, Dec 7, 2010 at 12:38 PM, Timothy Legge <timlegge at gmail.com> wrote:> Hi > > I was using the delivered Ubuntu 1.6.x packages but I wanted to look at > gtalk integration so I downloaded, compiled and installed the source (after > removing the Ubuntu packages) have installed the following: > > asterisk-1.8.0 > dahdi-linux-complete-2.4.0+2.4.0 > libpri-1.4.11.5 > > I copied my config back into place and most seems to work, but I cannot get > my phone that is plugged into the Wildcard TDM400P REV E/F card that I have > to work. > > Basically, I don't hear the dial tone and Asterisk does not register off > hook events.? I have spent time reviewing my config but I don't see what the > issue is. > > Is there anything I am missing, or can you suggest some additional things to > look at? > > Tim > > chan_dahdi.conf > grep -v "^;" /etc/asterisk/chan_dahdi.conf | grep -v "^$" > > [trunkgroups] > [channels] > language=en > context=phones > signalling=fxo_ks > usecallerid=yes > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > callreturn=yes > echocancel=no > echocancelwhenbridged=no > group=1 > callgroup=1 > pickupgroup=1 > > dahdi-channels.conf: > > ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) > ;;; line="1 WCTDM/4/0 FXOKS" > signalling=fxo_ks > callerid="Channel 1" <4001> > mailbox=4001 > group=5 > context=phones > channel => 1 > callerid> mailbox> group> context=default > > ;;; line="2 WCTDM/4/1 FXSKS" > signalling=fxs_ks > callerid=asreceived > group=0 > context=incoming-local > channel => 2 > callerid> group> context=default > > > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ? ? ? ? ? ? ? http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >
On Tue, Dec 7, 2010 at 8:17 PM, Thomas Perron <thomas.perron at gmail.com>wrote:> Do you have any issues with getting audio to bridge? > I am using 1.8 also. >Not so far, but I am still pretty excited to have a dial tone ;-) Two hours last night (and a 1:30 am bed time) lost because I missed one line in a config file. So far I have only tested dialing out through the Dahdi interface from a connected analog phone and a sip phone. That works, but I need to do something about the echo on the sip phone. Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101207/c0150906/attachment.htm