Re: [asterisk-users] Calls are dropped after 15 minutes

[Andrew Colin]

 I had a similar issue and i set a timeout which fixed the issue
SIP/trunk/ ${EXTEN},216,t

We only had this on one of our providers the rest we havent had the issue

- Original Message -
From: Steve Edwards 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Sat, 30 Jul 2016 20:27:45 +0200 (SAST)
Subject: Re: [asterisk-users] Calls are dropped after 15 minutes

On Sat, 30 Jul 2016, Keith Heppner wrote:

> We have a problem in that calls are dropped after 15 minutes (on both 
> internal and out going calls, incoming calls do not seem to have that 
> limit) How do we fix it?

You may gain some insight from viewing the console output after bumping up 
the debug and verbose levels.

You will probably resolve this by using tcpdump to capture packets and 
wireshark to see what's happening.

I had a problem with a similar description that was resolved by refusing 
SIP session timers.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[D'Arcy J.M. Cain]

On Sun, 31 Jul 2016 06:19:13 +0100
Nabeel  wrote:
> I tried your extension definition as suggested:

But did you understand every line and what it was doing?

> exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail)
> same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s)
> same => n,Hangup
> 
> But there was no change in the prompts asked, ie. the voice first
> asked for 'mailbox', and then 'password' as before. The prompts are
> not removed.

Too much information missing.  Perhaps instead of asking how to
implement the solution that you have already decided on you should
instead tell us what problem you are trying to solve.  Are you really
trying to make your voicemail available to anyone who calls you or are
you limiting it to just the registered phone?  How are you accessing VM?

> Please clarify what you mean by the following:
> 
> "If someone presses '*' while listening to your answer message then
> they are in your mailbox."
> 
> Do you mean while someone is listening to another user's 'unavailable
> message' or 'busy message', if they press '*' at that time they will
> enter the other person's mailbox?

That's exactly what I mean.  That's why you need to password protect
it.  The above recipe allows you to bypass the password prompt but only
when called from the registered phone and only when "*98" is dialed
from that phone.

I suspect that you need to read the documentation a lot more.  VoIP/SIP
is complicated.  I certainly don't understand everything but I was able
to craft the above extension by reading up on extensions as well as
system variables.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Asterisk News]

On 07/30/2016 11:19 PM, Nabeel wrote:

I tried your extension definition as suggested:

exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail)
same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s)
same => n,Hangup

But there was no change in the prompts asked, ie. the voice first asked
for 'mailbox', and then 'password' as before. The prompts are not removed.

Please clarify what you mean by the following:

"If someone presses '*' while listening to your answer message then they
are in your mailbox."

Do you mean while someone is listening to another user's 'unavailable
message' or 'busy message', if they press '*' at that time they will
enter the other person's mailbox?

Nabeel


Depending on the version of Asterisk you are running, and assuming that 
your phones are SIP devices, you might want to try this (lots of debug 
at the top which could be removed):



exten => *98,1,NoOp(--- VOICEMAIL MENU ---)
same => n,NoOp(CALLERID(num): ${CALLERID(num)})
same => n,NoOp(ip : ${SIPPEER(${CALLERID(num)},ip)})
same => n,NoOp(mailbox : ${SIPPEER(${CALLERID(num)},mailbox)})
same => n,set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same => n,NoOp(DATETIME : ${DATETIME})
same => n,Answer
same => n,Playback(silence/1)
;same => n,ExecIf($["${CALLERID(num)}" = 
"206"]?Set(VMBOX=102):Set(VMBOX=100))

same => n,Set(VMBOX=${SIPPEER(${CALLERID(num)},mailbox)})
same => n,VoicemailMain(${VMBOX},s)
same => n,Hangup

If you wish to override the default mailbox for a particular user, you 
can also uncomment the line above which in this example connects 
extension 206 to the voicemail box for extension 100.


;same => n,ExecIf($["${CALLERID(num)}" = 
"206"]?Set(VMBOX=102):Set(VMBOX=100))



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users