Topic: [Announce] SAIL-2.1.11-142 Beta

[Tib]

hello again

At the moment I have ...

Extention: 5000 ... spa1 = sipura3000
Extention: 5001 ... my computer = X-Lite
Extention: 5002 ... wife's computer = X-Lite
IVR Menu: Please pres 1 for so on and so forth

Call comes in ... goes to menu ... person presses 1 no one answers ... receptionist comes on and says ... sorry the person at extention 5001 is not available and there is silence ... then all of a sudden says thankyou.

The caller not knowing what to do just hangs up and wonders what that was all about.

Would make more sence if the receptionist said ... Sorry the person at extention 5001 is not available please leave a message ... then thankyou.

Now what I wouls like the system to do is ...

Call comes in ... menu (pre recorded message says what to do) .... this bit works.
Person decides to go for option 1 and call my computer .... I do not pickup but I am at home ... just not near computer ... this is where I would like the system to go back to a menu (new one maybe) and let the person select another extention or the system puts them in a queue and start going through the other extentions ...
If no one picks up then let the person have a choice to hang up or leave a message.

What would be the best way to execute this.

Regards,

Tib

[JonB]

Tib,

A couple of things.

The default voicemail has a tone at the end of the message however I suggest that you record your own greeting and say 'please leave a message after the tone'

You can create multi level IVR's but to do what you want you will have to create a custom app. Check out the asterisk documentation.

You can also change the SPA3k so that it answers the incoming call after so many sec's. This is how I have my SPA3k set up at home.

The home phones ring for 10 sec's so I can pick up the calls. If it is not answered after 10 sec then the SPA3k picks up the call and passes it to the 1st IVR message.

"we are unable to take your call......"
"press 1 to leave a message for Jon" (goes to my voicemail and email)
"press 2 to leave a message for Sue" (goes to Sues voicemail and email)
"press 3 to contact Computer Troubleshooters - Howick" (goes to a second IVR message)"

second message

"You have reached Computer Troubleshooters....."
"to contact the office press 1" (dials an office extension via a Sail2Sail trunk)
"to contact Jon by mobile press 2" (dials my mobile via a voip provider trunk)

Jon

[JonB]

Here are a couple of custom apps may you find useful in setting up an IVR.

First is to send an incoming caller directly to a mailbox.

Code: [Select]

[custom-vmxxxx]

exten => s,1,Voicemail,uxxxx
exten => s,2,Hangup()


where xxxx is the extension voicemail you want the call to go to

Second is giving an incoming caller access to DISA. DISA is internal dial tone. I make this a hidden option so that if I am away from the office at say a client site I can dial in on my PSTN (local calls in NZ are free), press the unannounced option for DISA, enter the password and get local dialtone from the office.

Code: [Select]

[custom-disa]

exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,Authenticate(****) ; replace **** with a numerical password
exten => s,5,DISA(no-password|from-internal)

The third is an example of dialling a number via a trunk

In this example I am calling my NZ mobile via Voipjet in the USA.

Code: [Select]

[custom-mobile]

exten => s,1,Dial(IAX2/user@voipjet/011642XXXXXXX,45,r)
exten => s,2,Hangup

011 is the number I use to access the Voipjet route, 64 is NZ country code, 2 is mobile code.

Jon

[dswillia]

Some more information about my caller ID issue, in the Asterisk CLI I get the following when doing a reload:

Mar 27 10:02:38 WARNING[9769]: chan_zap.c:10829 setup_zap: Ignoring caller_id
Mar 27 10:02:38 WARNING[9769]: chan_zap.c:10829 setup_zap: Ignoring signalling


Also, it may be interesting to note, when I reboot my server asterisk dies on the initial reboot, I have to do a selautosniff and then asterisk start to get the system back up and running

[SARK devs]

Hi DS

Thanks for this extra info, we're still looking at this.  It's an asterisk config problem of some description.  
Questions
1. Is CLID enabled on your analogue lines?  
2. Which country are you in and who is your telco carrier?

Thanks

[dswillia]

Yes, CID is enabled on those lines, as well as my VoIP circuit from MixNetworks, I am in the US and Southwestern Bell/ATT is my provider.

One other note pertaining to the GCP2000's:

The GXP2000 allows for BLF's (Busy Lamp Fields) so I can see if one of the extensions is currently on the phone.  Asterisk allows these to work via the "hint" command in the extensions.conf dialplan.  I was able to create a custom app named "Hint" for internal and put the following in:

exten => 5000,hint,SIP/5000
exten => 5001,hint,SIP/5001

This allows me to monitor the lines as well as one button speed dial the extension.  Just a FYI

Regards

[Tib]

Thanks JonB

I'll try this as soon as I get home.

I'm loving all this ... mind you I really don't need all this just for a home setup  but it's good to know just incase.

Looks like I need to do a bit more reading and searching ... asterisk is so powerfull.

Regards,

Tib

[dswillia]

Trying to setup a remote phone to my system, I have opened the correct IAX port and the softphone works fine internally.  I get the following error in the asterisk CLI:

AGI Script Executing Application: (Dial) Options: (IAX2/5004|30|twW)
Mar 28 11:21:18 NOTICE[3717]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)

Do I need to open a udp port on the other side?  Interesting to note, it shows the phone is registered correctly in both asterisk and on the remote phone.

Regards

[SARK devs]

HI

First of all, can we just say how astonished we were to release custom apps on friday evening and have Jon and DS put up working examples by monday!  You guys are amazing.

Other stuff (in no particular order).

Bugs..  

Asterisk fails to start after reboot

and

FOP only displays one conference room

We've created a "known-issues" section on the wiki and you can get fixes for both these issues there.

UDP Ports; - we aren't opening 5060 and  4569 correctly (thanks JonB).  We'll put a fix up onto the known-issues pages shortly.  

UDP 10000:20000, you should ONLY need to open these if you are running server-only behind a firewall.   You shouldn't need to open them in server-gateway mode unless you want to support a remote IP phone.  If you want to know why see

 http://selintra.com/docs/cgi-bin/view/Main/AstUdpPorts

CID not detected (reported by DS).  

We've been playing with this today DS.  We detect VOIP CID just fine on our SME7rc1-143 system here and it displays properly on all our test phones (gs101/102, GXP2000, spa-841, SJPhone, X-lite et al) .  

We don't detect analogue CID but you shouldn't read too much into this, BT CID is transmitted quite differently to the Bellcore CID used in the US and we have a few challenges with it at the moment.  As far as we know, we aren't doing anything to stop Bellcore CID recognition in our setup but we can't be sure.  You might want to check our generated zapata code with that of A@H (which you said worked OK).   Other than that you need to examine ${CALLERID} on inbound to see what, if anything, is in there.  

Sorry we can't be more forthcoming at the moment.

Thanks for your continued interest and support.  We really do appreciate it.

[dswillia]

Caller ID Issue Solved:

Ok looks like there is a misconfiguration in the zapata.conf file for my installastion anyways.  You have the line:

usecallerid=us

I assume this is automagicly set by setting in globals.  Anyways I changed it to:

usecallerid=yes

and it all works.

Regards

[SARK devs]

Ok looks like there is a misconfiguration in the zapata.conf file

Hi DS,

Thanks for spotting this.  We've posted a fix on the wiki at

http://selintra.com/docs/cgi-bin/view/Main/SysV21-143-SysZtl


Thanks again for all your help.


Selintra

[ldkeen]

Hi dswillia,
Thanks for that awesome tip with the blf's on the GXP2000. Finally managed to get around to setting it up. I had to upgrade the firmware in the phone though - but now I can't seem to get account2 to register?? Prior to the firmware upgrade I had both account1 and account2 registered as 5000 and 5003 respectively but now 5003 wont register. I notice that account2 uses udp5062. have you come across this problem??
Tib
I saw in a previous post by yourself you explained how to setup a transform mask for astratel (<:07>[35]xxxxxxx). I have an astratel trunk setup and I would like all my local calls to go through this trunk. Astratel require users to prefix all numbers with the area code. What I would like is for all 8 digit numbers dialed to be prefixed with 02 and routed through the astratel trunk. Is this correct <:02>xxxxxxxx and do I put it in the route-create panel or the trunk-change-transform mask panel?
Thanks everyone for the help and thanks JonB for the great custom apps
Lloyd

[JonB]

Lloyd,

Add port=5062 to the configuration for extension 5003.

Jon

[SARK devs]

What I would like is for all 8 digit numbers dialed to be prefixed with 02 and routed through the astratel trunk.

HI Lloyd,

In theory you should just be able to code a transform of

Code: [Select]


:02


However, running through the code in my mind (I don't have it here) I've got a horrible feeling it won't work.  

Of course if you also have other number sequences which need to go through the same trunk, then this won't work anyway.   One trick is to declare the same trunk twice with different names so you can route different number sequences through different transforms yet still punch them out through the same carrier.

Give it a try and let us know how you get on.  If it's broke, we'll fix it.  

Kind Regards

Selintra

[ldkeen]

Selintra,
<In theory you should just be able to code a transform>
I tried them all. I tried <:02>, :02, space:02 I tried putting the transform into the trunk-change panel as well as the route-create panel (by itself and combined with dial plan). I thought maybe Tib might have had a trick to get it to work.
<Of course if you also have other number sequences>
These 02 numbers probably account for well over 90% of the calls. I'm not to worried about others at the moment.
Thanks Lloyd