externip/localnet

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Hi All, Is it possible to specify more than 1 localnet? I know this is an odd
question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For
this to work I need to use the externip and localnet directive. If I do this
it rewrites the SDP with the external IP address of the main site on dialog
with the VPN'd sites. This means that I can either…

Asterisk Users 5 years ago 1 Answer

Registration attempts

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I am getting several hundred registration attempts on my aserterisk per
minute. I have fail2ban installed but it's not stopping the attempts. Any
suggestions. Whatever they are using is changing the userid on each
attempt. Latest IP: 209.172.57.219 Thanks,
Dave

Asterisk Users 5 years ago 3 Answers

quick 1.8 question on console/dsp

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In 1.4 I used alsa.conf and Dial(Console/Dsp) In 1.8 this is not working (as I had it) . I know there is a new
chan_console
I'd like to try both. What is the correct Dial() for ALSA direct?
What is the correct Dial() for chan_console? I "thought" if chan_alsa was loaded it would default to old behaviour
if chan_console was not loaded. Thanks, jerry

Asterisk Users 5 years ago 0 Answers

3rd party app store

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I recently came across this email that I wrote in May 2008 ......  http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html It's such a shame that Digium manhandled the project away from the community only to then bury it and not allow it to proceed. I really wonder when I look at the Apple iphone development community as to where the 3rd party Asterisk development community could have been if Digium didn't kill this project.
(for those of you not involved in Asterisk back in 2208 here is the audio of that conference call.
http://recordings.talkshoe.com/TC-22622/TS-109845.mp3?dl=1  )
Regards,
Dean Collins

Asterisk Users 5 years ago 17 Answers

Rotary phone on Asterisk

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I'm trying to use a couple of old Western Electric type 500 phones (desk
model, rotary dial). These phones work fine, as tested with telco lines
(they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino
channel bank which is connected to Asterisk via a Sangnoma card (T1 with
echo cancellation). Other phones (touch tone) work fine, as does any phone
with a pulse/tone switch, even when these electronic phones are in "pulse"
mode. I'm thinking that Asterisk is a…

Asterisk Users 5 years ago 3 Answers

do carriers detect unusual / unauthorized VoIP calling patterns?

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All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our
security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently
last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in remotely to a
company e-mail server and one found that his provider, Verizon, had blocked the server static IP. My question: do carriers build some type of "internal blacklist" if they detect unusual VoIP calling patterns?…

Asterisk Users 5 years ago 1 Answer

behavior changed between 1.4.25.1 and 1.4.36 with .call files

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Hi All, We have a production system running 1.4.25.1 and yesterday we upgraded it to
1.4.36. Basically we use this system to generate scheduled calls via .call
files. Sample .call file used: Channel: local/111111111@context-out
WaitTime: 30
CallerId: 333333333
Extension: 222222222
Context: context-out
Priority: 1 With the sample call file in 1.4.25.1 the behavior was: 1 - The asterisk box calls 111111111 with callerid 333333333
2 - When 111111111 answer the call asterisk calls 222222222 with
callerid 111111111 With the sample call file in 1.4.36 the behavior is: 1 - The asterisk…

Asterisk Users 5 years ago 0 Answers

need help with IVR dialplan

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Hi list
i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server
the call flow
IP phone ---INVITE 1001----> opensips -----> ASterisk ----INVITE
5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP
phone (5000) my case is:
1/ IP phone(5000) --->Opensips
2/ IVR number : 1001
3/ IP Phone calls 1001 to opensips --> asterisk, ASterisk will play IVR
4/ IP phone press 1, asterisk will Dial(SIP/to_opensips/5001,20)
5/ there are some cases when asterisk send call back to opensips
5.1/…

Asterisk Users 5 years ago 0 Answers

ISDN BRI call disconnection issue

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Hi, I have a Netmod ISDN BRI router and from the router I have connected the
analog port in Asterisk via FXO card. Two analog lines I have connected to
asterisk machine. When both the lines are established, after 31 minutes the
call is automatically disconnected. While checking the log it shows as busy tone is detected because of this
existing call is disconnected. Did anybody faced this kind of issue. Also some assistance would be much
appreciated.

Asterisk Users 5 years ago 0 Answers