Asterisk voicemail, FWD, 403 Forbidden Error, etc


#1

Hi,
I have a few asterisk questions. I will try to put as much details as I can, but feel free to ask any questions, if you need more info. I am running “Asterisk CVS-HEAD-03/15/05-01:15:30” on a FC3 machine. I have configured a few extensions on it and this asterisk machine does not have a phone line terminated in it.

  1. In my office, we have Zultys MX250 as the main telephony exchange and asterisk communicates with it for PSTN numbers. Currently, people are able to dial from Zultys to Asterisk and vice versa. People calling my Zultys from a PSTN number can also reach my asterisk extension by dialing the extension number, but when they hangup, the call still continues on asterisk. Hence, recording hours and hours of voicemail, if the call is not answered. Is there a way to fix this problem? I saw a tutorial on your site about fixing this through zapata.conf, but I am not sure if that would work in my case, as calls are coming over the ethernet to my asterisk box from Zultys.

  2. All the users of asterisk extension are using a softphone, hence, when they are not online and their extension is dialed, the dialing person gets a fast busy tone. I have discovered that asterisk rejects the call by returning a 403 forbidden instead of forwarding the dialing person to the voicemail. How can this be fixed?

  3. Now this is a bigger problem. I will try to put as much detail as I can, but this problem can be complicated. Like I mentioned earlier that we have a Zultys MX250 that has the PSTN terminated in it. I also have an FWD account, which also has a US number matched with it (thanks to IPKall). I am trying to set up my asterisk box to forward the call coming on my FWD to my cellphone THROUGH Zultys MX250. I am not sure how to make this work and I would appreciate any help.

  4. Since, calls are going through Zultys MX250, I want some way to authenticate users on asterisk, before they are allowed to make PSTN calls from zultys. I have seen a script (on voip-info.org) that authenticates users by parsing the voicemail.conf, but I am not sure how to use it to my benefit. I want users to be able to dial normal Zultys extension but for PSTN access they will have to enter their voicemail password. Any guidance in this regard would be very helpful.

I am glad that I found this forum and I would appreciate all the help that I can get. Thanks again guys/gals.

Some Desi


#2

Please provide more details on your configuration (ie - /etc/asterisk/sip.conf), how you have your extensions configured (ie - /etc/asterisk/extensions.conf) and the verbose output from your Asterisk CLI when this problem occurs.

This would be primarily the same as the way you are doing outbound from the softphones connected to Asterisk via the Zultys to the PSTN. You could do something like:

[incoming_fwd] ; The context that FWD/IPKall comes to exten => s,1,Dial(ZAP/....)

Further, you could set up IPKall to forward directly to your Asterisk box and forgo FWD, simpligyfing the interconnection, although not required.

You could route them to a context (ie - use a Macro) that would collect DTMF digits to compare against the voicemail.conf PIN numbers (even easier if you have your voicemail table in a database such as MySQL). You may then ask them to authenticate before you release their request into the outbound context via the Zultys.


#3

Thanks for the prompt reply.

  1. From sip.conf file:

[code][user0]
type=friend
secret=1234
username=user0
callerid=“user0” <8534>
host=dynamic
;dtmfmode=inband
mailbox=8534@user0
nat=yes
allow=gsm
allow=ulaw
allow=alaw

[user1]
type=friend
secret=1234
username=user1
callerid=“user1” <8527>
host=dynamic
mailbox=8527@user1
nat=yes
allow=gsm
allow=ulaw
allow=alaw[/code]

From extensions.conf file:

[code][default]

include => voicemail
include => IT-Staff
include => training

[IT-Staff]
exten => 8534,1,dial(SIP/user0,31,rt)
exten => 8534,2,Goto(8534-${DIALSTATUS},1)
exten => 8534-BUSY,1,Voicemail(b8534)
exten => 8534-BUSY,2,Hangup
exten => 8534-NOANSWER,1,Voicemail(u8534)
exten => 8534-NOANSWER,2,Hangup
exten => 8534,3,Hangup
exten => user0,1,goto(8534,1)

exten => 8527,1,dial(SIP/user1,31,rt)
exten => 8527,2,Goto(8527-${DIALSTATUS},1)
exten => 8527-BUSY,1,Voicemail(b8527)
exten => 8527-BUSY,2,Hangup
exten => 8527-NOANSWER,1,Voicemail(u8527)
exten => 8527-NOANSWER,2,Hangup
exten => 8527,3,Hangup
exten => user1,1,goto(8527,1)[/code]

I just noticed something else also. The calls get directed to voicemail, if the softphone was last run < 6 hrs ago. Since, my softphone was running right now, I couldn’t get the error to duplicate and get the details from Asterisk CLI, even after shutting down the softphone.

I am not sure, if this is the best way to define extensions, but I would definately appreciate your input on the above *.conf files. I have seen some standard macros but, I am not exactly sure how to fit them in my scenario.

  1. You suggest a good point. I completely missed the fact that I can configure my asterisk with IPKall. I think I will have to discuss that with the network department, because my asterisk machine is on an internal vlan (behind the firewall and has no Nat’ed IP address). If that doesn’t work, I guess I will have to work with FWD. I believe FWD also causes problems when machines are behind firewall.

  2. I got the problem 4 sorted out. Thanks for the help though.

Some Desi


#4

I’m using sip extensions and I’ve noticed that when I call a person that is not on line * answers:
"The personal extension xxx is on the phone"
instead of:
"The personal extension xxx is unvailable"
Voicemail answers anyway, but if someone calls me, and I’m not on the phone but simply disconnected, he could think I’m on the phone.

I’m currently learning something more about the ‘hint priority’:

voip-info.org/wiki-Asterisk+ … extensions
voip-info.org/wiki/view/Asterisk+presence

maybe this is your problem also.
C.