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 ch
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 te
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
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 any
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 independ
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 wrote:
> Hi Ish,
>
> I assume you are using Fail2Ban to monitor the logs for dictionary a
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 '(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for '(:[0-9]{1,5})?' - No matching peer
found
Registration from
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 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 get a
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 s
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 e
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'
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