Messages

Sounds like my issue might be elsewhere then, thanks

If it's internal that's causing you the issue, have you got your router set up to forward the correct ports (assuming you are on a single public IP with NAT)? You need to have ports 10000-20000 forwarded to your server as well as the 5060 control port.

I've got: (you'll need to change 1234567 and AAAAAAAA to your SIP telephone number minus the code, and secret respectively)

PEER DETAILS:

allow=alaw&alaw&ulaw&g729&gsm&slinear
auth=md5
authuser=1234567
callerid=1234567
canreinvite=no
context=from-trunk
dtmfmode=inband
fromdomain=sipgate.co.uk
fromuser=1234567
host=sipgate.co.uk
insecure=very
nat=yes
qualify=no
secret=AAAAAAAA
type=peer
user=1234567
username=1234567

USER DETAILS:

allow=alaw&alaw&ulaw&g729&gsm&slinear
auth=md5
authuser=1234567
callerid=1234567
canreinvite=no
context=from-trunk
dtmfmode=inband
fromdomain=sipgate.co.uk
fromuser=1234567
host=sipgate.co.uk
insecure=very
nat=yes
qualify=no
secret=AAAAAAAA
type=user
user=1234567
username=1234567

REGISTRATION:

1234567:AAAAAAAA@sipgate.co.uk/1234567

I have a sipgate connection, but calls are disconnecting on both inbound and outbound.

Have called sipgate, and they say that I am sending them a 'BYE' after 5 minutes.

Anyone know why it would be doing this?

I have a 'Higheasy 170X' 8 (or could be 16) channel H.264 capture card that I've put into my LMCE core.

Looking for the best way forward, in particular:

1. How to install drivers for it (and where to get them from??!) and test

2. What software is built into LMCE? (I know there is CCTV support, but is this 'out of the box', or should I be adding software to it. I have read about 'Zoneminder' which seems good, so should I install this if it isn't already?

Apologies if I'm wide of the mark, or missing info which is readily available here - I'm trying to assimilate info that I have picked up in various places and am now confusing myself; desperate for a little hand holding!

Thanks

Thanks - have enabled automatic config again, and can now select all extensions. However, I can only see the sipgate line; not the other two lines I have configured. I guess I need to recreate the trunks outside of FreePBX so that LMCE is aware of them. I've read this wiki article so can follow this, but don't see how this would give me an option to add a second line - am I missing something? Sorry if my actions are adding to your annoyance Thom, but as someone who has been installing PBX and VoIP equipment for well over a decade, I'd like to think that I'm 'knowledgeably adjusting settings via a commonly used GUI' rather than 'fucking shit up'. Everybody has different requirements, I know.

The IVR announcement is still playing after just a few rings, despite me having changed the answer time to 99. Any ideas? No offence Thom, but as an Englishman speaking 'original English', I can't understand a single word of what the IVR announcement is saying. It's dreadful. I guess this could be changed but would lose the text to speech functionality which I imagine is the reason it sounds synthesised. Could I do better? I don't know, but I'd rather just not use it anyway.

Really don't get this! Can someone please clear this up for me:

At present, only two phones ring on incoming trunk calls, and then go to the generic 'press x for all phones in house' etc.

I want to:

1) Make ALL phones ring
- I have looked at the default inbound route in FreePBX (custom app, with a string pointing to somewhere in pluto). Can't find any reference to this in Linux MCE. Have got a group that 'rings all', 600, but can't seem to point the inbound route to this as it won't let me??!

2) Either change the IVR or remove it.
- I've found the call routing section of LinuxMCE but cannot understand it as it refers to people rather than trunks etc. Maybe I'm missing something but...? Also, here I have tried setting '# of seconds to ring before ivr' to zero and also 99 in an attempt to disable it but nothing! Why am I feeling stupid right now?

Thanks Thom. Have had a fiddle and managed to get 'From Study' showing when I call someone - which param in sccp.conf controls the '201' in the top right of the 7970 screen?

Also, what freedom do I have in changing the pl_35 extension names? Don't want to break the core obviously...

Hi mtnbikeaz

Thanks for your reply; sorry for my slow response - didn't realise I hadn't subscribed to thread!

The Cisco SIP config includes Displayname and shortname.....shortname is what shows on the display.....Displayname is what ships with the Invite as your identity to display.

Not sure where you mean, can you expand on this please? I have had a look in the .cnf.xml files in /tftpboot/ but doesn't seem to be in there.

The background image can be served up from anywhere.  So if running Apache to serve up an HTML image will break anything, you can put it anywhere.  I use dropbox for some of my images or files needed anywhere.

Where is this configured please?

Thanks!

Hi all

Got my 7970 handsets working brilliantly (on SCCP) using the wiki article.

Am looking to see if I can customise them some more, and have been digging around and found snippits, but am wondering whether anyone has any experience they can help me with. In particular, I am trying to:

1 - Change the extension names (At the moment it says '201' and then '201' below it - would like it to say 'Kitchen' on screen, and that 'Kitchen' is calling when I ring someone). I've tried changing this in FreePBX and LinuxMCE but no joy.

2 - Disable push notifications to the orbiter function (am guessing this might be a generic orbiter option somewhere but I haven't found it yet? I know push is great BTW, just want to turn it off on one phone)

3 - Use the right hand context keys for direct extension dialling, AND line appearances if possible (ie I am extension 201, I can see 202 is on phone by looking at the screen, and when they come off the phone press it to call them)

4 - Any advice on background image - I read something somewhere about changing something in Asterisk but it concerned me the way it was worded in case it upsets LinuxMCE

5 - Also have read much on the 'SIP vs SCCP firmware' debate! Is SIP firmware working yet, or is it still best to stick with SCCP? It's working fine, so unless there's a reason to swap (call quality I think was mentioned on one page i read?) then I will stick with SCCP.

Thanks!
Tawny