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  #1 (permalink)  
Old 01-29-2007, 01:51 PM
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Posts: 97
Default Asterisk and Evolution and Asterisk Zimlet

Ok, I'm using a commercial distro of Asterisk called Evolution for Intuitive Voice.
http://www.intuitivevoice.com/

Management is uncomfortable with the idea of IT being in charge of phones also.... So we have paid supprot with them.

My question is I don't understand the fields in the Asterisk Zimlet.

1. I have no idea what my pbx username is I'm guessing its my phone extension?
2. Password? No Idea
3. My Conference Bridge - don't even know what that is.
4. My mobile number 757-xxx-xxxx
5. My Desk Number - I hope that is the same as my phone extension

I gather from the other posts that I need to manually edit the asterisk server config in the xml file and repackage and deploy.
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  #2 (permalink)  
Old 01-29-2007, 07:43 PM
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Posts: 5,606
Default

Hi Eric,
I've forwarded this onto the developers so that they can provide more information.

Thanks
john
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  #3 (permalink)  
Old 01-29-2007, 08:58 PM
Zimbra Employee
 
Posts: 4,792
Default

Quote:
Originally Posted by Ericx View Post
1. I have no idea what my pbx username is I'm guessing its my phone extension?
Phone ext is a good guess.
Quote:
Originally Posted by Ericx View Post
2. Password? No Idea
Same as what you use to get your voice mail.
Quote:
Originally Posted by Ericx View Post
3. My Conference Bridge - don't even know what that is.
A secondary number that can be used to bridge calls.
Quote:
Originally Posted by Ericx View Post
4. My mobile number 757-xxx-xxxx
yes
Quote:
Originally Posted by Ericx View Post
5. My Desk Number - I hope that is the same as my phone extension
your full desk phone so area code + number + ext
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  #4 (permalink)  
Old 01-31-2007, 12:08 PM
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Posts: 97
Default Config_template.xml asterisk

One more question?

sipHost: is the ip of the asterisk server correct?

myAddress: is the address of my phone?
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  #5 (permalink)  
Old 02-02-2007, 07:53 AM
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Posts: 97
Default Connection to server ok but commands incorrect

I can see that most of my configuration to Asterisk / Evolution is correct, because I'm can see stuff happening on Asterisk when I initiate a phone call.

I have posted two different phone calls below one form zimbra-asterisk-zimlet the other from my Polycom 501 phone.
----
Unsuccessful phone call from zimlet.
---
EvolutionPBX*CLI>
-- Executing NoOp("SIP/192.168.5.81-b7c46570",

"UNAUTHORIZED|UNAUTHORIZED") in new stack
-- Executing Answer("SIP/192.168.5.81-b7c46570", "") in new stack
-- Executing Playback("SIP/192.168.5.81-b7c46570", "699") in new stack
-- Playing '699' (language 'en')
Feb 1 19:17:18 WARNING[2811]: chan_sip.c:6668 get_refer_info: Referred-by:

Huh? Not a SIP header () Ignoring?
== Spawn extension (INVALID, 17575769301, 3) exited non-zero on

'SIP/192.168.5.81-b7c46570'
EvolutionPBX*CLI>




-------
Successful call from Polycom 501 phone
--------

-- Executing Set("SIP/1014-1798", "__DIALED=17575769301") in new stack
-- Executing Set("SIP/1014-1798", "__SOURCE=1014") in new stack
-- Executing Set("SIP/1014-1798", "__TYPE=LONG DISTANCE") in new stack
-- Executing Macro("SIP/1014-1798", "reporting") in new stack
-- Executing SetCDRUserField("SIP/1014-1798", "1170375287.1281|17575769301|1014|||LONG DISTANCE") in new stack
-- Executing Macro("SIP/1014-1798", "callrecord|1014") in new stack
-- Executing NoOp("SIP/1014-1798", "Call Record Macro "REC1014" ") in new stack
-- Executing GotoIf("SIP/1014-1798", "0?4:3") in new stack
-- Goto (macro-callrecord,s,3)
-- Executing GotoIf("SIP/1014-1798", "0?4:8") in new stack
-- Goto (macro-callrecord,s,8)
-- Executing Goto("SIP/1014-1798", "20") in new stack
-- Goto (INT-LOC-TOLL-BYPASS,17575769301,20)
-- Executing Gosub("SIP/1014-1798", "out-TRUNKGROUP1|s|1") in new stack
-- Executing NoOp("SIP/1014-1798", "Outbound") in new stack
-- Executing Set("SIP/1014-1798", "GROUP()=TRUNKGROUP1") in new stack
-- Executing Set("SIP/1014-1798", "__DESTINATION=TRUNKGROUP1") in new stack
-- Executing Set("SIP/1014-1798", "__STATUS=ANSWERED") in new stack
-- Executing NoOp("SIP/1014-1798", "") in new stack
-- Executing Macro("SIP/1014-1798", "reporting") in new stack
-- Executing SetCDRUserField("SIP/1014-1798", "1170375287.1281|17575769301|1014|TRUNKGROUP1|ANSW ERED|LONG DISTANCE") in new stack
-- Executing Set("SIP/1014-1798", "__TRANSDIALED=17575769301") in new stack
-- Executing Goto("SIP/1014-1798", "17575769301|1") in new stack
-- Goto (out-TRUNKGROUP1,17575769301,1)
-- Sent into invalid extension '17575769301' in context 'out-TRUNKGROUP1' on SIP/1014-1798
-- Executing Goto("SIP/1014-1798", "s|20") in new stack
-- Goto (out-TRUNKGROUP1,s,20)
-- Executing Dial("SIP/1014-1798", "ZAP/g1/ww17575769301||TWWgM(callsetup^TDM400P/2)") in new stack
-- Called g1/ww17575769301
-- Zap/1-1 answered SIP/1014-1798
-- Executing NoOp("Zap/1-1", "Call Setup Macro") in new stack
-- Hungup 'Zap/1-1'
== Spawn extension (out-TRUNKGROUP1, s, 20) exited non-zero on 'SIP/1014-1798'
EvolutionPBX*CLI>
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  #6 (permalink)  
Old 02-02-2007, 08:01 AM
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Posts: 97
Default My zimlet setting that produced the above output.

Config Template
-----
<zimletConfig name="com_zimbra_asterisk" version="1.0">
- <global>
<property name="sipHost">192.168.5.81</property>
<property name="myAddress">192.168.0.79</property>
<property name="debug">true</property>
</global>
</zimletConfig>

---
Preferences in the asterisk zimlet
---

PBX Username: evolution - Copied from /etc/asterisk/manager.conf
PBX Password: ******* - Copied from /etc/asterisk/manager.conf
My Conference Bridge: 1014
My Mobile Number: 757-576-9999
My Desk Number: 1014
Default Number: Desk


------
/etc/asterisk/manager.conf is
------

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
;displayconnects = yes

[evolution]
secret = *******
;deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
permit=192.168.5.0/255.255.255.0
permit=192.168.0.0/255.255.255.0
permit=192.168.0.89/255.255.255.0
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  #7 (permalink)  
Old 02-05-2007, 10:26 AM
Junior Member
 
Posts: 8
Default SIP passwords in Evolution

Hi Eric,

We're using Evolution, too. Extension passwords are stored in the clear (for the most part), in different files, depending what kind of phones you're using.

Polycom IP501s have their passwords listed on the Evolution box in /evo/tftp/[MAC address of phone]-auto.cfg in the reg.1.auth.password field.
Cisco 7940/7960s show theirs in /evo/tftp/SIP[mac address of phone].cnf

The config files will also clue you into appropriate auth names and so on -- they contain everything the hard phones need to know to get on the system.

SSH into the Evolution box, cat the file, and there's your password.

The Evo web interface will show you the password for a generic SIP extension in a pop-up window when you make a change to the extension.

I've found the Evolution guys to be forthcoming in giving access to supported systems. You could try emailing them at support@intuitivevoice.com, but please feel free to drop me a PM if you need a hand.

- Adam
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  #8 (permalink)  
Old 02-05-2007, 03:33 PM
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Posts: 97
Default It Worked!

Thank you! That worked like a charm!

How many phones do you have on Evolution? Currently we have 15 connected between two different sites. We want to expand that to 75 phones between the two different sites, with a dedicated server at each. I think the Zimbra integration will help seal the deal.
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  #9 (permalink)  
Old 02-05-2007, 03:56 PM
Junior Member
 
Posts: 8
Default

Glad to help.

Yeah, integrated messaging is kind of a big deal. Evolution specifics get kind of OT for the Zimbra forum, so look for a reply in your email. - Adam.
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  #10 (permalink)  
Old 02-06-2007, 03:27 PM
Loyal Member
 
Posts: 97
Default Evolution and Voicemail

Voicemail via email was having some trouble getting through to me. One thing I noticed from the /var/forgot/zimbra.log file was that id was rejecting email with .wav attachments. I've removed that attachment from attachment filter list in the Zimbra Admin console.

Hopefully this will work.
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