hello every one
i want to have multiple sip calls with different codecs for each one. for
example call to 8100 has g729 codec while call to 7900 has ulaw codec.
i searched a lot and found that there is some variable like sip_codec
which can set codec for a special inbound or outbound call. i don't
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get
the 'no matching peer' error when we get a dictionary SIP attack.
Now the logs always show a 'wrong password' when there actually isn't a
matching peer.
We even have alwaysauthreject = yes in our sip.conf.
Has anyone
Ishfaq Malik wrote:
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
get the 'no matching peer' error when we get a dictionary SIP attack.
Now the logs always show a 'wrong password' when there actually isn't a
matching peer.
We even have alwaysauthreject = yes in our
Hi
Thanks for the quick response. I'll read all the change logs from now on, I
promise!
Ish
On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote:
Ishfaq Malik wrote:
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
get the 'no matching peer' error when we
Hi Ish,
I assume you are using Fail2Ban to monitor the logs for dictionary attacks - If
so, the following regex should work for 1.8:
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No matching peer
found
Hi Arthur
It was a fail2ban based query and fail2ban is still working fine.
I was just trying to find out if the change was intentional or not.
Regards
Ish
On 4 November 2013 15:52, Arthur J. Stanfield a...@dmcip.com wrote:
Hi Ish,
I assume you are using Fail2Ban to monitor the logs for
Hi all,
What should I do when my E1/SIP provider need a specific callerid(num)
setting for my outgoing calls?
My problem is that if I use Set(CALLERID(num)=XXYY) then I got XXYY on
the cdr src field, losing info of the real src of the call.
What is the best way out? (Preferably tech
Le 01/11/2013 18:02, A J Stiles a écrit :
What you want is not just a SIP client; it also has to integrate with
the phone's own GSM stack.
You probably will have the most success with Android, because you are
going to need well-documented Source Code to stand a chance of
getting
Le 01/11/2013 18:54, ad...@3a.hu a écrit :
Hi,
On 11/1/2013 5:02 PM, A J Stiles wrote:
You probably will have the most success with Android, because you
are going to need well-documented Source Code to stand a chance of
getting anywhere. You will need an Open Source SIP client and the
For outgoing calls you can write additional information into the userfield, or you can define
your own additional fields using an adaptive-odbc setup. For ISDN and POTS channels you can
typically set the callerid (just the number) for outgoing calls only to those numbers given to
you by your
Dear All.
When I calling a number from web, my softphone show me Answer and
Decline bottoms, and then I have to click Answer to call the number. it
seems it is two step to calling the number. If I type the number direct to
my client softphone, it calls directly the number without show me to
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