Recent content by Duaneness

Here's the weird thing: if I program an autodial button on the SIP set to call my (H323) extension, it works. If I manually dial my five-digit extension, it waves off after the first digit. Further testing shows that this only happens when dialing extensions that start with a "1" or a "2"...

CM 7.0, SMGR 7.0, SessMgr 7.0. Two SIP test sets - one J179, one 9611G. Both have the latest SIP firmware for their model. I finally managed to get each set registered to the Session Manager. I can call each from my H323 set on my desk. On either SIP set, if I go off-hook and dial an internal...

Good day everyone. I am trying to connect an Avaya G3si v9.5 to a stand-alone FreePBX box, which will be used for voicemail only. I have a PRI T1 in place which mostly works - calls go to the correct mailbox, and messages are left. However I can't get MWI to work at all. I have tried a number of...

Still working this issue. I've tried two different Calista SMDI boxes, but neither seems to be able to properly connect to the Definity. So I'm going to give up on the PRI and go to an analog hunt group for the connection between the Definity and the Asterisk Box (being used for voicemail only)...

Hey all, I have a Connected Systems PBXLink connected between an Avaya G3siV9.5 and an Asterisk/FreePBX box being used exclusively for voicemail. I have ISDN/PRI connectivity between the Avaya and the Asterisk; Calls cover to voicemail with no issue. I have FreePBX setup to use SMDI to provide...

Solution: Needs to be: "External Notify" set to smdi.conf Once I made that change (submitted and applied config), I started getting SMDI data. - Duaneness

There is only one physical serial port on the box. Output of the setserial command: [root@freepbx ~]# setserial -g /dev/ttyS[0123] /dev/ttyS0, UART: 16550A, Port: 0x03f8, IRQ: 4 /dev/ttyS1, UART: unknown, Port: 0x02f8, IRQ: 3 /dev/ttyS2, UART: unknown, Port: 0x03e8, IRQ: 4 /dev/ttyS3, UART...

FreePBX 15.0.16.42 Asterisk 16.6.2 Getting the error: ERROR[2457] app_voicemail.c: No valid SMDI interface specified, disabling SMDI voicemail notification Voicemail.conf contains the following lines: [general] smdiport=/dev/ttyS0 smdienable=yes externnotify=SMDI smdi.conf contains the...

There are no phones on the FreePBX/Asterisk box. It is strictly being used for voicemail. It's an older Avaya system in a school environment. During class hours, calls to teachers are sent directly to voicemail. Need the indicator lights to alert teachers to new messages. Don't have...

I don't know if Asterisk will allow a pause to be programmed. Everything in my original post was found in a variety of different places on the web. If I dial 1062235 from another phone on the system, I turn on the MWI on the phone for 2235. I don't currently have any analog stations on the...

Asterisk 13.22.0 FreePBX 14.0.13.26 Avaya Definity v9.5si, IP enabled (but not SIP capable) I am an old Avaya guy, and a newbie to Asterisk and Perl scripting. And I apologize profusely for the length of this post. I am trying to set up FreePBX as a standalone voicemail server for an Avaya...

Hello, I had a need today to terminate an active conference running on Avaya Aura Conferencing (AAC) 8.0. I have no idea how to do so. I was watching the conference on Element Manager using the KPI / Monitoring and reporting. Short of deleting the user through the admin interface, I'm unsure if...

Thanks for the docs, they are a great reference. I'm trying to add a J139 SIP phone to our H323 environment. It looks like I'm running into an issue with the certificates. I included the System Manager CA certificate (SystemManagerCA.cacert.pem) to the TRUSTCERTS line in the settings file, but...

Avaya CM 7.0 / System Manager/Session Manager 7.0 Running 46xx/96xx/96x1 IP sets, all H323. Looking for a "quick setup guide" to add a SIP phone.I found lots of help on SIP trunking (which I'll need later, but we're not there yet), but nothing on adding a SIP station to an H323 environment...

Trunk Group type: wats; Trunk Type: loop-start Under "Outgoing Dial Type" my choices are "automatic", "mf", "rotary", and "tone". I don't see a way to turn this off. And I don't see any reference to "cut through access." - Duaneness