Asterisk as Voicemail and AA system for CME

Unanswered Question
Apr 11th, 2007

Hi All

I won't bore you with the background whys and wherefores suffice to say we now have an enviroment where we have migrated from asterisk open source PBX to CME based system but cannot yet move to CUE.

We have the asterisk (extension range 1300 - 1399) and cme systems (extension range 1400-1499) talking to each other via sip trunks and all is well.

What we would like to do now is to use the voicemail functionality of asterisk to provide voicemail for our cme users.

I'm relatively new to CME but think the answer lies in translation profiles?

I created a dummy extention on asterisk box (51401) to store the voicemail for CME extention 1401.

The voicemail access number is 600. Which can be called from the CME system but of course it asks for the mailbox number presumably because CME is presenting the wrong caller-ID (1401 instead of 51401)?

Am I correct in thinking that a translation profile could be used to add the 5 to the outgoing called number of 600? If so would somebody mind providing me an example of a suitable translation profile.

Also how would mwi indicator work? I'm guessing that mwi sip-server 192.168.x.x unsolicited would work but would i have to have an incoming translation profile to remove the 5 from the mwi message?

Of course if I'm barking up the wrong trees and someone knows of a better way to solve this (apart from buying Unity!) - please let me know.

Thanks in advance

I have this problem too.
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Paolo Bevilacqua Wed, 04/11/2007 - 12:11

Hi Antony,

yes your problem can be in the numbers. If you can't configure the voicemail to work with the native numbers, then translation-profile can change them.

So for example;

dial-peer voice 600 voip

destination-pattern 600

session-protocol sipv2

translation-profile outgoing add5

voice translation-profile add5

translate calling 500

voice translation-rule 500

rule 1 // /5/

For the MWI, observe the numbers that are exchanged with "debug ccsip message", and configure rules for incoming / outgoing, calling / called accordingly.

Hope this helps, if so please rate post!

anthonyfear Wed, 04/11/2007 - 13:54

Thanks v. much

I'll try this out tomorrow and post back if successful.

anthonyfear Fri, 04/13/2007 - 01:24

Ok used your exmaple and it worked sort of as intended - i.e.

When I call the voicemail number directly it changed my calling number to 51401 which voicemail recognises. Great!

But - when i use a call-forward noans 600 it does not add the 5.

I see from debugs that this may be because it's now the called number and not the calling number?

How do i get round that? any ideas?

Paolo Bevilacqua Fri, 04/13/2007 - 01:42

Hi Anthony, try:

translation-profile xx

translate redirect-target yy

other otptions is redirect-called. But I thing 'target should be the right one.

Good luck!

anthonyfear Fri, 04/13/2007 - 05:45

Thanks again for that

I can see how the commands are changing the numbers sent which may be useful in the future. Unfortunately it didn't seem to resolve the problem.

I have instead setup each ephone-dn with a call-forward noan to an individual direct voicemail number which seems to get around the problem. i.e. ephone-dn 2 which is extension 1401 forwards to *51401 which is a direct voicemail number for the extension.

Now I need to get MWI working. Can you help with that? or should i post a new question?

n2lbt Fri, 04/13/2007 - 10:32

I am doing something similar with my asterisk 1.2.0 and CUCME 4.1. I have extensions 200-299 on asterisk and 300-399 on Cisco. I do not need a custom voicemail forward for each phone. I use the pilots of 105, 106, and 107. Attachments are of just the relevant sections of the cisco and asterisk config files.

Attachment: 
n2lbt Fri, 04/13/2007 - 10:45

That is where I started more than a year ago. This config has evolved quit a bit since then. I'm using the [email protected] distro which is the old TrixBox. Asterisk is Asterisk though underneath. I will upgrade to TrixBox someday, but I don't really need any of the GUI bells and whistles and there really was no big asterisk changes since then. After learning cisco CLI voice, I hang out in the shell on the asterisk box now anyway.

This config should provide you some reference, you'll just be using 4 digits instead of 3. My VM config looked like yours before I honed it down to what I have now.

anthonyfear Fri, 04/13/2007 - 12:37

i'm sorry but what ?!?

Reading between the lines - Are you saying that I can replace my config on Trixbox with your config from asterisk and it'll work? I'm just not sure that's true? But hey what the heck - i've got nothing to lose so i might give it go.

Lucky for you that your comfortable with both cme and asterisk cli - I haven't been able to find the time to learn asterisk cli as well as CME.

Thing is I'm keen to push CME as THE VoIP solution not asterisk, so I'm not going to bother learn it now (not in detail anyway) - however a lot of my clients are upgrading from Trixbox / [email protected] to CME so it'd be nice to figure out a stepped upgrade.

n2lbt Fri, 04/13/2007 - 13:44

Actually, no, I haven't tried this specific config on Asterisk 1.4(trixbox). If you shell into your asterisk box you'll see all these config files in /etc/asterisk/. You won't need many commands in there, just

cd /etc/asterisk/

ls -l (lists files)

pico -w filename.conf

You can see how FreePBX(AMP for me) modifies the individual (xxxx_additional.conf) files. I make changes in the main conf files(sip.conf extensions.conf) so AMP or FreePBX doesn't overwrite my changes. I was lucky that I bought an 1751v and had a test bed I could setup this offline without interrupting daily functions of my Cisco/Asterisk production setup.

Paolo Bevilacqua Fri, 04/13/2007 - 12:02

Are the phones SIP or SCCP?

If the latter, you need

sip-ua

mwi-server ipv4: expires 3600 port 5060 transport udp unsolicited

and, "mwi sip" under ephone-dn or ephone-dn-template.

n2lbt Fri, 04/13/2007 - 12:12

99 % SCCP. I like SIP and all but SCCP still hums along like a champ. The SIP phone is a playtoy Linksys SPA-942, just so I can follow along with what Cisco is doing. In general the 942 is fine, you just loose all the pretty functions of a native Cisco phone or even a SIP Cisco phone.

mwi commands are not needed for SCCP in my setup. The CME takes the MWI messages from the asterisk and turns on the light on the phone auto-magically. I was trying the dial method in the beginning and it was very unreliable. About a year ago one of the T releases fixed the unsolicited and now I use that.

anthonyfear Fri, 04/13/2007 - 12:29

i had already tried the sip-ua bit but it wasn't working.

I see i need to add it to the ephone-dn too?

I'll try that post back.

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