Re: [asterisk-users] how to decrypt encrypted SIP user's secret

[Duncan]

On 29/06/16 16:37, Nathan Anderson wrote:


You must mean that engineer before you used "md5secret" instead of 
"secret" for each user in sip.conf?


If so, why can't you just copy the md5secret line from the old server 
to the new server for each user?


-- Nathan

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ikka 
Tirtawidjaja

*Sent:* Tuesday, June 28, 2016 6:20 PM
*To:* asterisk-users
*Subject:* [asterisk-users] how to decrypt encrypted SIP user's secret

Dear all,

My office have an old asterisk PBX system (asterisk 11.4), and it 
encrypt all the SIP User's secret.


But the voip engineer before me didn't save / documented those password.
Now the server's hardware is begin to broke, it hangs a lot, and have 
a lot of call problem.


We already have a new asterisk PBX to replace it, but we have 
difficulty to retrieve the encrypted password.


Are you talking about sip registrations, as above they can be found in 
the sip config files, assuming your PBX is still at least working and 
you can look at the config files.



about a hundred of our customer use an old IPPhone that doesn't have a 
reset button to hard reset the admin password (back to factory 
default). The previous engineer also change the IPPhone's admin 
password without any documentation. So, we can not move / change those 
IPPhone to the new PBX.




Is there a way for us to retrieve / decrypt those SIP secret ?


Or anyone has any experience how to reset this IP Phone (Dayou Ddip-100)

Most phones don't have a reset button, rather a keypress combination to 
reset them. I can't read the manual unfortunately, can you get hold of 
the manufacturers?



Any suggestion are appreciate, because I'm really desperate.

You can figure out the SIP password from the files, and if you shift the 
IP address to the new server you should be fine.



Thanks in advance,


Regards,

Ikka

Jakarta - Indonesia





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Re: [asterisk-users] how to decrypt encrypted SIP user's secret

[Nathan Anderson]

You must mean that engineer before you used "md5secret" instead of "secret" for 
each user in sip.conf?

If so, why can't you just copy the md5secret line from the old server to the 
new server for each user?

-- Nathan

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ikka Tirtawidjaja
Sent: Tuesday, June 28, 2016 6:20 PM
To: asterisk-users
Subject: [asterisk-users] how to decrypt encrypted SIP user's secret

Dear all,

My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all 
the SIP User's secret.
But the voip engineer before me didn't save / documented those password.
Now the server's hardware is begin to broke, it hangs a lot, and have a lot of 
call problem.

We already have a new asterisk PBX to replace it, but we have difficulty to 
retrieve the encrypted password.

about a hundred of our customer use an old IPPhone that doesn't have a reset 
button to hard reset the admin password (back to factory default). The previous 
engineer also change the IPPhone's admin password without any documentation. 
So, we can not move / change those IPPhone to the new PBX.

Is there a way for us to retrieve / decrypt those SIP secret ?

Or anyone has any experience how to reset this IP Phone (Dayou Ddip-100)

Any suggestion are appreciate, because I'm really desperate.

Thanks in advance,


Regards,

Ikka
Jakarta - Indonesia
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[asterisk-users] Asterisk hep.conf

[Annus Fictus]

hello,

I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server.

My hep.conf Asterisk configuration is:

[general]
enabled = yes
capture_address=107.170.151.154:9060
;capture_password = foo
capture_id = 2464

SIP Signaling work correctly but no RTCP STATS arrive to Homer Server. 
On the Asterisk Console, many messages like this:


NOTICE[3739] res_hep.c: Unable to send packet: Address Family mismatch 
between source/destination


Any hint?

Regards

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[asterisk-users] how to decrypt encrypted SIP user's secret

[Ikka Tirtawidjaja]

Dear all,

My office have an old asterisk PBX system (asterisk 11.4), and it encrypt
all the SIP User's secret.
But the voip engineer before me didn't save / documented those password.
Now the server's hardware is begin to broke, it hangs a lot, and have a lot
of call problem.

We already have a new asterisk PBX to replace it, but we have difficulty to
retrieve the encrypted password.

about a hundred of our customer use an old IPPhone that doesn't have a
reset button to hard reset the admin password (back to factory default).
The previous engineer also change the IPPhone's admin password without any
documentation. So, we can not move / change those IPPhone to the new PBX.

Is there a way for us to retrieve / decrypt those SIP secret ?

Or anyone has any experience how to reset this IP Phone (Dayou Ddip-100)

Any suggestion are appreciate, because I'm really desperate.

Thanks in advance,


Regards,

Ikka
Jakarta - Indonesia
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[asterisk-users] AttendedTransfer Event not Fired

[Valter Nogueira]

I am using Asterisk 13.9.1 and want to catch AttendedTransfer, but it is
not fired at all.

Thank you,

Valter
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[asterisk-users] Audio cutting in and out - asterisk 13.1 cert6 / confbridge

[Robert McGilvray]

Hello,

We use Asterisk extensively for conferencing - for the last 8 years or so this 
has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 
350 concurrent users. Right around that number DAHDI hit's a hard coded memory 
limit and kicks allocation errors in the log.

[Jun 22 10:04:13] WARNING[9095] app_meetme.c: Unable to open DAHDI pseudo 
channel: Cannot allocate memory

In order to support our growing user count we recently upgraded to 13.1-cert6 
with pjsip and replaced meetme with confbridge. During all of our UAT and load 
testing everything seemed to be fine, there were no perceived audio quality 
issues or any logs that would indicate an issue. Unfortunately now that we're 
in production I'm getting consistent complaints that the audio from 
participants is cutting in and out. It only seems to occur while under load 
with > 350 users but that is anecdotal at best. This is not a simple networking 
issue, we've pretty much ruled that out with various performance testing. That 
was not the case initially and we had incrementing UDP packet receive errors 
which we've eliminated with a bit of tuning.

There are numerous architectural differences between the two installations and 
so far I have not been successful in determining the root cause. I'm reaching 
out to the community and the developers for insight and feedback hoping there 
is prior experience with this issue and how to resolve it. As you can see below 
the most significant difference is probably the use of VMware on the new 
install. I've tuned the ESXi host and guest per VMware's recommendations for 
latency and jitter (full cpu/mem reservations) with no improvement. With all of 
the reading I've done I suspect my issue may come down to a timing source and 
VMware not providing a reliable clock. It seems they allow a backlog of 
interrupts and if it hasn't caught up in 60s they are simply dropped.

Before I rip apart the environment and rebuild on physical I'd like to try and 
confirm that hypothesis. In the past this was a simple matter of running 
dahdi_test which would report the accuracy. I'm not sure how to interpret the 
results of "timing test" in the Asterisk CLI. If I increase the number of ticks 
per second the results are erratic while under load. I'm using the timerfd 
module in Asterisk with a 1000HZ tick kernel and high res timers enabled. I've 
tried both hpet and tsc as system clock sources, both exhibit the same breaks 
in audio. It sounds like someone presses the mute button in the middle of a 
sentence.

Any insight is appreciated!

Here are the specs on the new install:

Physical HWCisco UCS Blade (UCSB-B200-M3)
vMware   ESXI 5.5
VM Guest   4 vCPU w/ 32G of RAM tuned for latency/jitter 
(sensitivity=high) and full cpu/memory reservations.
VM OS  Redhat EL7 kernel 3.10.0-327.13.1.el7.x86_64 with 
tickless disabled e.g nohz=off and 1000HZ.
Asterisk13.1-cert6 using the timerfd module.

Regards
Robert McGilvray
SS&C GlobeOp
Associate Director, IT Network Security

GlobeOp Financial Services | 1565 Front Street | Yorktown Hts NY 10598

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[asterisk-users] SIP client able to handle Access-URL: header

[Bertrand LUPART - Linkeo.com]

Hello,


I'm playing with the optional URL parameter of the Dial() command, which "will 
also be sent to the called party upon successful connection, if the channel 
technology supports the sending of URLs in this way."[1]

Basically, the following asterisk dialplan directive :

- - 8< - - 8< - - 8< - -
same => n,Dial(SIP/1234,30,tr,http://www.example.com/)
- - >8 - - >8 - - >8 - -

Will generate the following SIP header in the INVITE request :

- - 8< - - 8< - - 8< - -
Access-URL: ;mode=active
- - >8 - - >8 - - >8 - -


Does anyone know about a SIP client able to handle this header ? I'd expect it 
to automatically open this webpage.


Thank you,


[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

-- 
Bertrand LUPART


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