Asterisk PBX Integration Zimlet (new)

I was able to install the zimlet and get the configuration working and the zimlet looks brilliant! However, I can't get the calls to take place.

I've confirmed from Asterisk that the manager account is logging in and logging out:

Code:

---

  == Manager 'administrator' logged on from 192.168.111.124
  == Manager 'administrator' logged off from 192.168.111.124

---

But the handoff doesn't take place and I'm not sure where to look. When I click the number in Zimbra it says "Pickup phone to dial number #######" and then two seconds later, "Failed to dial #######. Extension not found."

Any suggestions on where to look? I've tried with multiple working phone extensions but it does not go. I've also verified that the manager "administrator" has the call and command attributes.

Thanks!

What does Asterisk show when this happens ?

Code:

---

asterisk -rvvvv

---

Asterisk just shows the logon and logoff of the manager users. It's what I posted earlier:

Code:

---

  == Manager 'administrator' logged on from 192.168.111.124
  == Manager 'administrator' logged off from 192.168.111.124

---

It doesn't show any call attempts or failures.

Hi bluecc
The "Extension not found." means what it says. It seems that asterisk does not know the extension of the phone you want to place a call for. Are you sure the dialplan is right? Please check the extension you configured in the zimlets prefences dialog (srcPhone). Then recheck the zimlet global settings to match your dialplan. Especially the properties: astDialChannelType, srcPhonePrefix, astDialContext

The source channel used to dial for is built the following way:

Code:

---

${astDialChannelType}/${srcPhonePrefix}${srcPhone}

---

Example:

Code:

---

<property name="astDialChannelType">SIP</property>
<property name="srcPhonePrefix">77</property>
<property name="astDialContext">internal</property>
srcPhone Property in zimlet's user config: 10

---

This means your voip phone you want to dial for needs to be configuerd in context internal and its channel is SIP/7710


Hope this helps

Thank you for the advice. It looks like I was handling the phone prefix and user preferences extension setting incorrectly. However, after changing I still have the same issue.

As an overview, the extension I'm trying to use is 200. So I now have the following set:

Code:

---

<property name="astDialChannelType">SIP</property>
<property name="srcPhonePrefix">2</property>
<property name="astDialContext">phones</property>
srcPhone Property in zimlet's user config: 00

---

This is what is in the 'phones' context for that user:

Code:

---

exten => 200,1,Dial(SIP/200,20)
exten => 200,n,VoiceMail(100,u)
exten => 200,n,Playback(vm-goodbye)
exten => 200,n,Hangup()

---

This is what the dial plan for the phones context is:

Code:

---

exten => _X.,1,Set(CALLERID(ani)=OURNUMBERMASKED)
exten => _X.,2,Dial(SIP/${EXTEN}@viatalk-2)
exten => _XXXXXXXXXXX,1,Dial(SIP/viatalk-2/${EXTEN})
exten => _011X.,1,Dial(SIP/${EXTEN}@viatalk-2)

---

Any other thoughts on where to look or does something look out of place?

Now you mixed up srcPhonePrefix :)
With the settings you posted the phone's channel will be SIP/002 BTW: Four your setting its easier to let srcPhonePrefix empty. Then just set srcPhone Property in zimlet's user config to 200

So srcPhonePrefix is essentially for if you need to dial a outside line eg. 9 ?

No the purpose of srcPhonePrefix is simply a string that is joined with zimelts users config (srcPhone). If you get a number block from your phone provider like 12 3400 - 12 3420 you can configure 1234 as srcPhonePrefix and the users need just to configure their extension that is used internally (means 00 - 20). E.g. user with extension 12 then configuers 12. The idea is that the extenal reachable direct number of that user will be 12 3412. But this depends on your dialplan.

Hope this helps to clarify.

I went ahead and left the srcPhonePrefix blank and just set the user preferences extension to 200. Unfortunately the problem still persists though.

Any other thoughts on what to check? Does it matter how the extension is setup in sip.conf?

Thanks for helping and I can't wait to get it working! Our users are really looking forward to this.

yes, sip.conf does matter... Whats the context you set there?