Hi,

is it possible to get the SIP IP address of the remote (calling) party,
in the dialplan or (preferrably) in an AGI script?

(This sounded like a rather basic question to me, but I could not find
an answer…)

TIA & regards
Jakob

hi all,

have asterisk set up in combination with fail2ban.
all works as expected only there is 1 extension that is trying to
register with a wrong password causing fail2ban to block the IP address,
normally that is ok behaviour but i have several extensions on that IP
address.

someone has once setup this extension as a test, but it seems to be
impossible to find where it was installed (probably a softphone) and its
a bit far away for me to go check it out.

since there is no way to stop the troubling extension i figured i might
as well let it connect by setting the wrongly used password as correct,
at least in that case the other extensions won’t be blocked and i can
try to call that extension myself.

anyway to see which “wrong password” is being used?

much obliged,

Randall

Hi List,

Just looking for advise on the current best approach to work around an
issue I am experiencing.

Currently I have a provider with whom I register, their domain has
multiple destinations defined in their SRV record for their domain.
Registrations appear to be working properly, and outbound calls appear
to be working as well. The problem arises when I receive inbound calls,
sometimes the calls are able to come through, and in other instances the
calls arrive but are not pushed through to the correct context because
the source ip address does not match that of the peer definition. I am
currently running 2 pbx’s, one is version 1.6.2.5 and the other is 1.8.5.0.

What are others out there doing that are facing the same issue. As far
as I am aware, the only way I know of to work around this issue is to
have multiple peer definitions for each address in the SRV list. Is this
still the case?

Thanks

I am using my first PAP2 device from linksys. Used many polycom phones…

I configured the PAP2 device with asterisk. I have the registration,
thought I was good to go.
Plugged in my Valcom 2924 public address analog connection, called the
extension
and I got busy… very strange I thought.

I then looked at the status page of the PAP2 and it says the following
Reg online and hook state OFF.

How do I get the hookstate to be ON so that I can call into the device?

I have power cycled it and it always shows hookstate OFF.

Thanks,

Jerry

hello list

i have configured the voicemail in my server asterisk 1.4 i can use it
without issue ,i have a question

i want to receive an email in my address email when there is no response
from 270 after 10 s

could you please verify the code below and tell me what is wrong

thanks and regards

extensions.conf
exten => 9999,1,VoiceMailMain(777@mb_tutorial)

exten => 270,1,Dial(SIP/270, 10)
exten => 270,n,VoiceMail(777@mb_tutorial)
exten => 270,n,PlayBack(vm-goodbye)
exten => 270,n,HangUp()

voicemail.conf
[mb_tutorial]
777 => 1111,270,salah.elharit200@gmail.com,,|attach=no|review=yes