Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11
On Mon, Aug 29, 2005 at 09:54:11AM -0700, Anthony Rodgers wrote: > We are using * with an Option 11C - we tried all of the various > protocols and the only one we could get to work satisfactorily was > 5ESS, with the * as CO and the Nortel as remote. The one drawback of > this approach is getting name information for caller ID - because the > Nortel sees the * as CO, it won't send the name information. This is our experience as well: National ISDN-2 doesn't work (calls only go one way) because of limitations on the Meridian unit. 5ESS only works if the Meridian is CPE and Asterisk is NET. But you can only send CID names from Asterisk to Meridian, not vice versa. According to our phone system consultants (TAC Centre) there isn't any way to get the Meridian of this age to transmit CID names in this case. -- Karl A. Krueger <[EMAIL PROTECTED]> Network Security -- Linux/Unix Systems Support -- VoIP -- etc. Woods Hole Oceanographic Institution ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Won't hangup like Polycom 600 will
On Wed, Apr 13, 2005 at 03:08:43PM -0400, George Burt wrote: > I have been experimenting with GXP-2000 Grandstream. [snip] > Polycom>Asterisk>Grandstream > then Hangup the Polycom, the Grandstream plays a busy signal. This is my experience as well. The GXP2000 always seems to play a busy signal when the other end hangs up. It does not seem to matter which end placed the call, or whether the call is SIP-SIP or SIP-*-PSTN. This is not what any other phone does in my experience, so I think I'm reasonable in considering it a bug. I am overall unimpressed with the GXP2000 -- although the cost savings is significant, the lack of features by comparison with other "enterprise" marketed SIP phones (e.g. the SNOM or Cisco offerings) is also quite substantial ... and I do not think I will ever get used to the idiosyncratic Grandstream way of entering alphanumeric addresses. -- Karl A. Krueger <[EMAIL PROTECTED]> Network Security -- Linux/Unix Systems Support -- Etc. Woods Hole Oceanographic Institution ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote MWI via IAX?
On Mon, Feb 07, 2005 at 02:22:07AM -0800, George Pajari wrote: > We have a couple of Asterisk boxes with one being the main system with > everyone's voicemail and the other a slave used merely to link a couple > of remote phones to the main system using IAX. > > How can one propagate message waiting indication from the main system to > the remote phones? This is not quite the same set-up, but we have phones registered against an SER proxy with voice mail on Asterisk. We send MWI NOTIFYs using the Asterisk externnotify feature and a shellscript that composes the NOTIFY message and sends it using sipsak. Here's the script we are using with externnotify, and the template it uses. Please note that you have to set the configuration settings to your SIP environment ... CUT HERE check_voicemail.sh #!/bin/bash # # check_voicemail.sh -- Asterisk voice mail externnotify script # # This script traverses the voice mail for a user and sends a SIP NOTIFY # indicating how many voice mail messages they have. # # It can be triggered directly (e.g. from Asterisk dial plan) whenever it # is appropriate to think that the number of messages has changed. # # It can also be triggered periodically (e.g. from periodic_vm_check.sh) # just in case a user has missed a NOTIFY. # # Command-line parameters: # $1 -- $VM_CONTEXT -- the called context (ignored) # $2 -- $EXTEN -- the user account to check ([EMAIL PROTECTED]) # $3 -- $NUM_MSG -- ignored # # Configuration: # # THIS_HOST -- the name of the host we are running on THIS_HOST=some.host.example.com # # THIS_IP -- the IP address of the host we are running on THIS_IP=127.0.0.1 # # SIP_HOST -- the IP address of our SIP Proxy where the users are SIP_HOST=127.0.0.2 # # SIP_DOMAIN -- the domain into which the users are registered SIP_DOMAIN=example.com # # SIP_SENDER -- the identity to send the message as SIP_SENDER=1234 # users can call back to check voice mail # # LOG_FILE -- where to store our debugging log LOG_FILE=/tmp/voicemail-log # # VM_ROOT -- the root of the Asterisk voice mail spool VM_ROOT=/var/spool/asterisk/voicemail # # TEMPLATE -- the message template for a SIP NOTIFY message TEMPLATE=/etc/asterisk/notify.msg echo "RUN AT" `date` >> $LOG_FILE echo "$1 $2 $3" >> $LOG_FILE VM_CONTEXT=$1 E_AT_C=$2 NUM_MSG=$3 EXTEN=`expr substr $E_AT_C 1 \( \( index $E_AT_C @ \) - 1 \)` CONTEXT=`expr substr $E_AT_C \( \( index $E_AT_C @ \) + 1 \) 64` USER_DIR=$VM_ROOT/$CONTEXT/$EXTEN # Did we get OK syntax? if [ "$EXTENfoo" == "foo" ] then echo Error, no user specified > /dev/stderr exit 1 fi # Is there actually any such user? if ! [ -d $USER_DIR ] then echo No VM user $EXTEN in $CONTEXT > /dev/stderr exit 1 fi # Count the number of VM messages for the user. pushd $USER_DIR > /dev/null TOTAL_MESSAGES=`find . -name "*.txt" | wc -l | sed 's/^ *\(.*\) *$/\1/'` NEW_MESSAGES=`find INBOX -name "*.txt" | wc -l | sed 's/^ *\(.*\) *$/\1/'` popd > /dev/null OLD_MESSAGES=$(($TOTAL_MESSAGES - $NEW_MESSAGES)) if [ "$NEW_MESSAGES" == "0" ] then HAS_NEW="no" else HAS_NEW="yes" fi CONTENT_LENGTH=$((34 + `expr length $HAS_NEW` + `expr length $NEW_MESSAGES` + `expr length $OLD_MESSAGES`)) CMD="s/!SUBSCRIBER!/$EXTEN/g;s/!IP!/$THIS_IP/;s/!SENDER!/$SIP_SENDER/g;s/!DOMAIN!/$SIP_DOMAIN/g;s/!MAILBOX!/$EXTEN/g;s/!CONTENT_LENGTH!/$CONTENT_LENGTH/g;s/!HAS_MESSAGE!/$HAS_NEW/g;s/!NEW_COUNT!/$NEW_MESSAGES/g;s/!OLD_COUNT!/$OLD_MESSAGES/g" cat $TEMPLATE | sed $CMD | flip -m - - > /tmp/mwi-$$ sipsak -H $THIS_HOST -f /tmp/mwi-$$ -p $SIP_HOST -s sip:[EMAIL PROTECTED] rm /tmp/mwi-$$ CUT HERE end of check_voicemail.sh CUT HERE notify.msg NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP !IP!:5060 From: To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 1 NOTIFY User-Agent: VoiceMail Event: message-summary Content-Type: application/simple-message-summary Content-Length: !CONTENT_LENGTH! Messages-Waiting: !HAS_MESSAGE! Voicemail: !NEW_COUNT!/!OLD_COUNT! CUT HERE end of notify.msg -- Karl A. Krueger <[EMAIL PROTECTED]> Network Security -- Linux/Unix Systems Support -- Etc. Woods Hole Oceanographic Institution ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
