[asterisk-users] Directory Application

2020-09-25 Thread John T. Bittner 
Hello all,

Anyone know an easy way to have the Directory 
Application 
lookup all the voicemail contexts in the system. Like a global option


John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

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[asterisk-users] Confbridge

2020-08-07 Thread John T. Bittner 
To all:

No matter what I try, I cannot get the system to wait for the admin to join. It 
just dumps users into the bridge directly.
I do not have a pin for users, does that matter?

What am I missing?

Another issue the absolute timeout is not working ? ... have recordings that 
last for over 24 hours... and this should not happen...
All calls should hangup after 4 ?

Any ideas ?

Any help is much appreciated.

Thanks

This is my dialplan.

exten => s,1,Wait(1)
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(absolute)=14400)
exten => s,n,NoOp(${CALLERID(name)})
exten => s,n,NoOp(${CALLERID(num)})
exten => s,n,NoOp()
exten => s,n,Playback(church) ; "Please hold while..."
exten => s,n,Set(CONFBRIDGE(user,announce_join_leave)=no)
exten => s,n,Set(CONFBRIDGE(user,startmuted)=yes)
exten => s,n,Set(CONFBRIDGE(user,template)=church)
exten => s,n,Set(CONFBRIDGE(user,marked)=no)
exten => s,n,Set(CONFBRIDGE(user,wait_marked)=yes)
exten => s,n,Set(CONFBRIDGE(user,end_marked)=yes)
exten => s,n,ConfBridge(xaccel)
exten => s,n,hangup

confbridge.conf

[general]
[church]
type=user
startmuted=yes
announce_join_leave=no
announce_user_count=no
wait_marked=yes
end_marked=yes
music_on_hold_when_empty=no
quiet=yes
;
[xaccel]
type=bridge
record_conference=yes
;
Then calling in I see this
Conference Bridge Name   Users  Marked Locked Muted
 == == == =
xaccel1  0 No No


John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

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Re: [asterisk-users] ICE error

2020-07-16 Thread John T. Bittner 
Hello all,

Running Asterisk 16.10.1

Does anyone know what this means?

rtp_recvfrom: PJ ICE Rx error status code: 70004 'Invalid value or argument 
(PJ_EINVAL)
How can I find what value it doesn't like ?

I switched to a few different stun servers and I still get the same error.

Calls still go through

Any help is much appreciated.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
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intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] includes with time and timezone.

2020-06-15 Thread John T. Bittner 
Hello,

I cannot find much on examples but I did find one in Russian that shows this to 
use + or - the time difference from GMT.
I have been testing and it does not work.

1st question do includes work with timezone

include =>  day,08:00-17:00,mon-fri,*,*,[+5]
Not sure on the formatting, is it correct ? ... I tried without the brackets... 
that also doesn't work.

If not supported in includes

What is the formatting for timezone in gotoiftime.

GotoIfTime(times,weekdays,mdays,months,[timezone]?[labeliftrue:[labeliffalse]])


Any helps is much appreciated.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] PJSIP

2020-05-29 Thread John T. Bittner 
Hello,

Anyone know how to set the "To:"  in an invite for PJSIP to custom settings. I 
got the "from" to be the way I need it.

From: 

I have tried a lot of changes to get to this but nothing works.

I am getting this
From: sip:109643...@xaccel.net;tag=42e4a9cb-59af-4d40-a21f-00261afbd3be
To: sip:34.221.174.202

I need to put "TEST" http://www.xaccel.net/>

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[asterisk-users] Modems

2020-02-11 Thread John T. Bittner 
Guys,

I have a customer that heavily uses modems, the problem they don't work 
reliably with some of the carriers I have used like Level3.
This is somewhat expected due to the limits in VoIP so I need a better solution.

If I set up an asterisk system on customer premise with an FXS card in it and 
have calls sent to another asterisk box with a PRI can I get this to be more 
reliable and better connect speeds.?
Any way to detect a modem call and turn off the echo canceller?

What if I use a lossless codec between the two systems, or would it be better 
to just run PCM to passthrough to the PRI.

Any ideas would be helpful.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] Looking Asterisk SIP Guru

2019-06-27 Thread John T. Bittner 
Joshua,

Thanks for looking into this, and sorry for not being more detailed.
Running asterisk 16.4.0

I was able to get in touch with an AIphone tech and it turns out that these 
issues are known bug on their side.

I will be more detailed next time

Thanks

John Bittner
Xaccel



-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Joshua C. Colp
Sent: Thursday, June 27, 2019 10:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Looking Asterisk SIP Guru

On Thu, Jun 27, 2019, at 11:28 AM, John T. Bittner wrote:
>  
> Hello,
> 
> 
> I am looking for a consultant that know asterisk in and out including 
> how to troubleshoot sip and rtp.
> 
> I have a device that this acting very strange and I need to prove it’s 
> the device code and not an issue with my setup.
> 
> 
> Very simple setup, all local no nat… Grandstream video phone and a 
> AIphone IX-MX7 door station.
> 
> 
> PJSIP … doorstation to grandstream 3370 works perfectly. Early video 
> works as well.
> 
> PJSIP … grandtream to doorstation I get a error from the doorstation I 
> get

You didn't provide the IP addresses of things involved, so anyone looking at 
the packet captures has to look in and decipher what is what which may be why 
noone on here has responded as of yet. The user agent of Asterisk is also 
changed so that confused things some for me to until I double checked the SDP 
and saw it's Asterisk.

Asterisk is sending a re-invite to 192.168.1.10 as an attempt to make both the 
audio and video streams bidirectional. The device at 192.168.1.10 is rejecting 
this with a 400 Bad Request. It should respond either with a 200 OK with an SDP 
answer of the state of the streams, or it should respond with a 488 Not 
Acceptable. Both of these would keep the call up and the appropriate stream 
would probably flow although I haven't tested this particular usage.

You also didn't specify an Asterisk version from what I can see, and stream 
behavior between 13 and 16 differs (as 16 understands streams) which could 
contribute to the behavior.

--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
www.digium.com & www.asterisk.org

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[asterisk-users] Looking Asterisk SIP Guru

2019-06-24 Thread John T. Bittner 
Hello,

I am looking for a consultant that know asterisk in and out including how to 
troubleshoot sip and rtp.
I have a device that this acting very strange and I need to prove it’s the 
device code and not an issue with my setup.

Very simple setup, all local no nat… Grandstream video phone and a AIphone 
IX-MX7 door station.

PJSIP … doorstation to grandstream 3370 works perfectly. Early video works as 
well.
PJSIP … grandtream to doorstation I get a error from the doorstation I get

SIP/2.0 400 Bad Request
To: ;tag=ec09c0b4zps4.0.0
From: "108";tag=bcaee3d1-d6d1-4354-aee1-885d4b89182d
Via: SIP/2.0/UDP 
192.168.1.154:5060;branch=z9hG4bKPj038d15bb-c53b-4677-a471-4fa44a21599b;rport
Call-ID: 08caa86d-8fc8-4ed1-bec2-6828acb9e017
CSeq: 17397 INVITE
Content-Length: 0
x-reinvitekind: mediadirectionchange

Tried a few things, I still don’t understand why I am getting this, I cannot 
find it coming from the asterisk system or the Grandstream in my traces.
So
Switch the Aiphone to use chan_sip on port 5099 just to test.

Again
SIP … doorstation to PJSIP grandstream 3370 works perfectly. Early video works 
as well.
PJSIP … granstream to SIP doorstation works somewhat, I get early video but no 
audio. If I answer the doorstation before the early video pops up, I get the 
window in the doorstation that allows me to put a call on hold.
When I do, and take back off hold, I get audio.
If I wait for early video on the doorstation and then answer it, the door 
station never comes up with the menus to put a call on hold. So no audio.

Anyone have any ideas or willing to do some consulting work please let me know 
asap.
FYI some captures are attached.

Thanks

John Bittner
CTO

380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.
0¿
ª0NEiJ@@LÀ¨
À¨ÄÄ}ª!SIP/2.0 400 Bad Request
To: ;tag=ec09c0b4zps4.0.0
From: "108";tag=bcaee3d1-d6d1-4354-aee1-885d4b89182d
Via: SIP/2.0/UDP 
192.168.1.154:5060;branch=z9hG4bKPj038d15bb-c53b-4677-a471-4fa44a21599b;rport
Call-ID: 08caa86d-8fc8-4ed1-bec2-6828acb9e017
CSeq: 17397 INVITE
Content-Length: 0
x-reinvitekind: mediadirectionchange





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Re: [asterisk-users] Hacking

2019-06-16 Thread John T. Bittner 
I took a look for that, Mysql running but blocked in the firewall.
I do have a web gui but its hides the passwords + has a single login for admin 
with complex password.
Even if they hacked the web site, they have no way of getting the passwords my 
configs are static in the asterisk folder.
SSH is blocked.

Logs do not show any http access, secure or any other fingerprints.

I am going to honeypot this box to see if I can capture there invites.

John
Xaccel



From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Sunday, June 16, 2019 6:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Hacking

John,

There are a lot of factors at play for instance are you using a gui that has a 
known vlun? Is there mysql running on the box with a simple password? Perhaps 
they didnt hack your PBX but they comprised a SIP phone  and once they had the 
credentials  they made calls? Do you have a provisioning system?

We have seen all of the above. Most of the compromises we are seeing these days 
is either via a Provisioning server or phones that are accessible on the 
internet with weak passwords




Regards,

Dovid
From: j...@xaccel.net
Sent: June 16, 2019 18:37
To: asterisk-users@lists.digium.com
Reply-to: 
asterisk-users@lists.digium.com
Subject: [asterisk-users] Hacking


Anyone know how someone can hack an asterisk box and register with every single 
account on the box.
This box only has 3 accounts, with very complex passwords. Have VoIP blacklist 
setup and fail2ban…

The hackers were able to make 2 calls to Cuba before my alerting system texted 
me.

I am running asterisk 16.3 with PJSIP.

This is my only box open to the outside world, a requirement for this one 
customer.
Looked into my logs… can't find anything out of the ordinary.


Any ideas ?



  Contact: 

==

  Contact:  
12120001001/sip:12120001001@5.79.64.23:9227
ee80678930 NonQual nan
  Contact:  848842405/sip: 
848842405@5.79.64.23:9227  
031ed703ba NonQual nan
  Contact:  848842405/sip: 
848842405@5.79.64.23:9227  
031ed703ba NonQual nan
  Contact:  
ghbhhm/sip:ghbhhm@5.79.64.23:9227  
959fc8fbf4 NonQual nan
  Contact:  
ghbhhm/sip:ghbhhm@5.79.64.23:9227  
959fc8fbf4 NonQual nan
  Contact:  
ghbhhm/sip:ghbhhm@5.79.64.23:9228  
d7bf838918 NonQual nan
  Contact:  
ghbhhm/sip:ghbhhm@5.79.64.23:9228  
d7bf838918 NonQual nan

Any helps is much appreciated.


John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.




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[asterisk-users] Hacking

2019-06-16 Thread John T. Bittner 
Anyone know how someone can hack an asterisk box and register with every single 
account on the box.
This box only has 3 accounts, with very complex passwords. Have VoIP blacklist 
setup and fail2ban...

The hackers were able to make 2 calls to Cuba before my alerting system texted 
me.

I am running asterisk 16.3 with PJSIP.

This is my only box open to the outside world, a requirement for this one 
customer.
Looked into my logs... can't find anything out of the ordinary.


Any ideas ?




  Contact: 

==

  Contact:  12120001001/sip:12120001001@5.79.64.23:9227ee80678930 NonQual   
  nan
  Contact:  848842405/sip: 848842405@5.79.64.23:9227  
031ed703ba NonQual nan
  Contact:  848842405/sip: 848842405@5.79.64.23:9227  
031ed703ba NonQual nan
  Contact:  ghbhhm/sip:ghbhhm@5.79.64.23:9227  959fc8fbf4 NonQual   
  nan
  Contact:  ghbhhm/sip:ghbhhm@5.79.64.23:9227  959fc8fbf4 NonQual   
  nan
  Contact:  ghbhhm/sip:ghbhhm@5.79.64.23:9228  d7bf838918 NonQual   
  nan
  Contact:  ghbhhm/sip:ghbhhm@5.79.64.23:9228  d7bf838918 NonQual   
  nan

Any helps is much appreciated.


John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] Fail2ban for asterisk 16 PJSIP

2019-06-07 Thread John T. Bittner 
Hopefully, this helps someone else.

This seems to be working for me.

# Fail2Ban configuration file
[INCLUDES]
#before = common.conf
[Definition]
failregex = NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*' 
.* - No matching endpoint found
NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*' 
.* - Failed to authenticate
NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*' 
.* - Error to authenticate
NOTICE.* .*: Request \'INVITE\' from '.*' failed for ':.*' .*

John Bittner
Xaccel

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of John T. Bittner
Sent: Thursday, June 6, 2019 3:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fail2ban for asterisk 16 PJSIP

Hello

Anyone have a working copy of Fail2ban asterisk filter asterisk.conf
for Asterisk 16 running PJSIP.

I have tried 10 different filters but none of them show any matches when 
testing with
fail2ban-regex

I see date template hits but no matches

My log
[2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
50670137772977-30593645157868@192.168.1.8<mailto:50670137772977-30593645157868@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:37:52] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"as100" ' failed for 
'188.214.128.172:5076' (callid: 03e7f9d2dcdf4252506c440137e822b7) - No matching 
endpoint found
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 
211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>)
 - Failed to authenticate
[2019-06-06 15:39:17] NOTICE[18081]

[asterisk-users] Fail2ban for asterisk 16 PJSIP

2019-06-06 Thread John T. Bittner 
Hello

Anyone have a working copy of Fail2ban asterisk filter asterisk.conf
for Asterisk 16 running PJSIP.

I have tried 10 different filters but none of them show any matches when 
testing with
fail2ban-regex

I see date template hits but no matches

My log
[2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 50670137772977-30593645157868@192.168.1.8) - Failed to authenticate
[2019-06-06 15:37:52] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"as100" ' failed for 
'188.214.128.172:5076' (callid: 03e7f9d2dcdf4252506c440137e822b7) - No matching 
endpoint found
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate
[2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate
[2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate
[2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' 
(callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate
[2019-06-06 15:39:17] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"as100" ' failed for 
'188.214.128.172:5071' (callid: 8e12f1560bfe2c3ed5be895108727c46) - No matching 
endpoint found

Any help is much appreciated.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

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Re: [asterisk-users] Account code PJSIP

2019-05-02 Thread John T. Bittner 
Hopefully, this may help someone in the future.

If I set this before I dial out... it works.

I have always in the past set this on hangup... that does not work anymore.

John
Xaccel


From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of John T. Bittner
Sent: Wednesday, May 1, 2019 12:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Account code PJSIP

Does anyone know how to set accountcode into the asterisk CDR when using PJSIP?

I have tried Set(CHANNEL(accountcode)=XX) and a few other older ways... 
nothing works.

If I add accountcode into the pjsip endpoint config it works... but I need to 
set it via dialplan.

Any help is much appreciated.

Testing on asterisk 16.3.0

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net<http://www.xaccel.net/>

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.




Teach Canit xAntispam if this mail is spam:
Spam<http://mx1.xantispam.net/canit/b.php?c=s=0206R70lG=6ea45c5ca3d8=xaccel-net>
Not 
spam<http://mx1.xantispam.net/canit/b.php?c=n=0206R70lG=6ea45c5ca3d8=xaccel-net>
Forget previous 
vote<http://mx1.xantispam.net/canit/b.php?c=f=0206R70lG=6ea45c5ca3d8=xaccel-net>
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[asterisk-users] Account code PJSIP

2019-04-30 Thread John T. Bittner 
Does anyone know how to set accountcode into the asterisk CDR when using PJSIP?

I have tried Set(CHANNEL(accountcode)=XX) and a few other older ways... 
nothing works.

If I add accountcode into the pjsip endpoint config it works... but I need to 
set it via dialplan.

Any help is much appreciated.

Testing on asterisk 16.3.0

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] PJSIP DNS ISSUE

2019-02-21 Thread John T. Bittner 
I tried a bunch of stuff , all my stuff was ip based. I was on asterisk 15, the 
fix was to upgrade to 16.

After that every staying working after the internet went down.

Thanks for the help

John

Sent from my iPhone

On Feb 20, 2019, at 11:33 AM, Ryan, Travis 
mailto:ry...@oscarwinski.com>> wrote:

Can’t you just reference everything in IPs? If not, then hardcode the IPs in 
your /etc/hosts file. I think that’s a bad idea, but that’s one way to ensure 
you always have the Ip of a domain name.

From: asterisk-users 
mailto:asterisk-users-boun...@lists.digium.com>>
 On Behalf Of John T. Bittner
Sent: Wednesday, February 20, 2019 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] PJSIP DNS ISSUE

Anyone know how to disable DNS in asterisk so PJSIP still works when the 
internet goes down.


I tried a few things but nothing is working. I even installed BIND on the 
asterisk box …that didn’t even work. Once I pull the plug on the internet, I 
cant dial anything.



John Bittner
CTO

380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net<http://www.xaccel.net/>

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.




Teach Canit xAntispam if this mail is spam:
Spam<http://mx1.xantispam.net/canit/b.php?c=s=02XD4xKDC=44dcd078a67a=xaccel-net>
Not 
spam<http://mx1.xantispam.net/canit/b.php?c=n=02XD4xKDC=44dcd078a67a=xaccel-net>
Forget previous 
vote<http://mx1.xantispam.net/canit/b.php?c=f=02XD4xKDC=44dcd078a67a=xaccel-net>
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[asterisk-users] PJSIP DNS ISSUE

2019-02-20 Thread John T. Bittner 
Anyone know how to disable DNS in asterisk so PJSIP still works when the 
internet goes down.


I tried a few things but nothing is working. I even installed BIND on the 
asterisk box ...that didn't even work. Once I pull the plug on the internet, I 
cant dial anything.



John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] opus

2019-01-21 Thread John T. Bittner 

Does anyone know where do get opus for asterisk 16 that runs on GLIBC_2.12?

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] Cisco ATA 191

2018-12-05 Thread John T. Bittner 
Hello,

Anyone have a copy of the SEP or SIP config file for this type of device. ( 
Cisco ATA 191 )  I looked all over the internet and can't find anything.

I did find the sip firmware below but very little docs on this model.
ATA191.12-0-1-29.loads.

Any help is much appreciated.

Thanks


John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
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unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] BLF screen alerts

2018-11-05 Thread John T. Bittner 
Anyone know how to turn off screen notification on phones with BLF buttons.

I am using PJSIP.

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] Convert SIP to PJSIP

2018-09-26 Thread John T. Bittner 
Hello all,

I am having some trouble converting this setup from SIP to PJSIP. Any help is 
much appreciated.

I used the converter script and get most of it but don't see a registration 
entry.
How do you convert this entry into PJSIP.
This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321

[17185553321]
type = peer
host = sip.ringcentral.com
transport=udp
defaultuser=332940285773   ; Authentication username for outbound proxies
username = 332940285773
fromuser=17185553321   ; Many SIP providers require this
fromdomain=sip.ringcentral.com
secret = ARi4uYb2Mz
canreinvite = no
disallow = all
allow = ulaw
nat = yes
dtmfmode = auto
rfc2833compensate = yes
trustrpid = yes
usereqphone = yes  ; This provider requires ";user=phone" on URI
callcounter = yes  ; Enable call counter for parallel outbound calls
busylevel = 2  ; Signal busy at 2 or more calls (feel free to 
adjust)
outboundproxy=sip12.ringcentral.com:5090

This is what it was converted too: But nothing for the registration ?

[17185553321]
type = aor
contact = sip:332940285...@sip.ringcentral.com

[17185553321]
type = identify
endpoint = 17185553321
match = sip.ringcentral.com

[17185553321]
type = auth
username = 17185553321
password = ARi4uYb2Mz

[17185553321]
type = endpoint
dtmf_mode = none
disallow = all
allow = ulaw
rtp_symmetric = yes
rewrite_contact = yes
outbound_proxy = sip12.ringcentral.com:5090
direct_media = no
from_user = 17185553321
from_domain = sip.ringcentral.com
device_state_busy_at = 2
auth = 17185553321
outbound_auth = 17185553321
aors = 17185553321

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

-- 
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Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Convert SIP to PJSIP

2018-09-24 Thread John T. Bittner 
Hello all,

I am having some trouble getting this to work under pjsip. Any help is much 
appreciated.

I used the converter script and I see it register but can't receive or send to 
ringcentral.

Anyone get this working with PJSIP?

Works with chan_sip...

This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321

[17185553321]
type = peer
host = sip.ringcentral.com
transport=udp
defaultuser=332940285773   ; Authentication username for outbound proxies
username = 332940285773
fromuser=17185553321   ; Many SIP providers require this
fromdomain=sip.ringcentral.com
secret = ARi4uYb2Mz
canreinvite = no
disallow = all
allow = ulaw
nat = yes
dtmfmode = auto
rfc2833compensate = yes
trustrpid = yes
usereqphone = yes  ; This provider requires ";user=phone" on URI
callcounter = yes  ; Enable call counter for parallel outbound calls
busylevel = 2  ; Signal busy at 2 or more calls (feel free to 
adjust)
outboundproxy=sip12.ringcentral.com:5090

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

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[asterisk-users] Early or Pre VIdeo

2018-02-24 Thread John T. Bittner 
Does anyone know if asterisk 15 supports Video before answering a call.

I am running PJSIP and have tested a bunch of settings, progress, answer call 
before calling the other phone and  even set the devices to send direct media.
Source video device is a door phone that is supposed to support early video.

Video works ok once answered.

Any help on this would be very helpful and appreciated.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
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intended recipient(s) and may contain confidential
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unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] CDR with conference asterisk 12

2015-03-01 Thread John T. Bittner 
Hello,

Anyone see this issue, I have a conference bridge setup for a church with a 
Barix unit that streams audio into the bridge.
The bridge is started by calling in to a number that executes a call file and 
the system calls the Barix unit starting the broadcast.
Users then call in and can listen to the sermons live. The system works 
flawless with 1 issue I can't get accurate cdr's.
Every time a new caller enters the bridge, a dupe cdr is created for users that 
are already in the bridge. So instead of say 30 calls showing up in the 
reports, they show 300.

Any idea why this would happen, I have upgraded asterisk with no change.
Dial plan is very basic.

I see this in CDR debugs of a simple test. 3 phones calling in and you will see 
2 dupes for the 2 tests.

Any help is much appreciated

Thanks

John Bittner
Xaccel

0x7fbbe8001628 - Created CDR for channel SIP/199.73.108.66-0003
0x7fbbe8001628 - Transitioning CDR for SIP/199.73.108.66-0003 from state 
NONE to Single
    -- Executing [2018913284@did:1] Goto(SIP/199.73.108.66-0003, 
extensions,2551,1) in new stack
    -- Goto (extensions,2551,1)
    -- Executing [2551@extensions:1] Answer(SIP/199.73.108.66-0003, ) 
in new stack
0x7fbbe8001628 - Set answered time to 1425186131.851613
    0x7fbb9c019480 -- Probation passed - setting RTP source address to 
199.73.108.66:14962
    -- Executing [2551@extensions:2] Playback(SIP/199.73.108.66-0003, 
churchwelcome) in new stack
    -- SIP/199.73.108.66-0003 Playing 'churchwelcome.slin' (language 'en')
    -- Executing [2551@extensions:3] Set(SIP/199.73.108.66-0003, 
CONFBRIDGE(user,announce_join_leave)=no) in new stack
    -- Executing [2551@extensions:4] Set(SIP/199.73.108.66-0003, 
CONFBRIDGE(user,startmuted)=yes) in new stack
    -- Executing [2551@extensions:5] Set(SIP/199.73.108.66-0003, 
CONFBRIDGE(user,template)=church) in new stack
    -- Executing [2551@extensions:6] ConfBridge(SIP/199.73.108.66-0003, 
) in new stack
    -- Channel SIP/199.73.108.66-0003 joined 'softmix' base-bridge 
6dfaada2-2d01-44e4-b025-914ca18eee25
Bridge Enter message for channel SIP/199.73.108.66-0003: 1425186138.00529497
0x7fbbe8001628 - Updating Party A SIP/199.73.108.66-0003 snapshot
0x7fbbe8001628 - Processing bridge enter for SIP/199.73.108.66-0003
0x7fbbe8001628 - Transitioning CDR for SIP/199.73.108.66-0003 from state 
Single to Bridged
0x7fbbe80025f8 - Created CDR for channel Local/2498@extensions-;1
0x7fbbe80025f8 - Transitioning CDR for Local/2498@extensions-;1 from 
state NONE to Single
0x7fbbe80025f8 - Set answered time to 1425186138.529575
0x7fbbe80025f8 - Transitioning CDR for Local/2498@extensions-;1 from 
state Single to Bridged
0x7fbbe80025f8 - Party A Local/2498@extensions-;1 has new Party B 
SIP/199.73.108.66-0003
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
0x7fbbe8003aa8 - Created CDR for channel SIP/199.73.108.66-0004
0x7fbbe8003aa8 - Transitioning CDR for SIP/199.73.108.66-0004 from state 
NONE to Single
    -- Executing [2018913284@did:1] Goto(SIP/199.73.108.66-0004, 
extensions,2551,1) in new stack
    -- Goto (extensions,2551,1)
    -- Executing [2551@extensions:1] Answer(SIP/199.73.108.66-0004, ) 
in new stack
0x7fbbe8003aa8 - Set answered time to 1425186165.284291
    0x7fbb9c02ba60 -- Probation passed - setting RTP source address to 
199.73.108.66:18760
    -- Executing [2551@extensions:2] Playback(SIP/199.73.108.66-0004, 
churchwelcome) in new stack
    -- SIP/199.73.108.66-0004 Playing 'churchwelcome.slin' (language 'en')
    -- Executing [2551@extensions:3] Set(SIP/199.73.108.66-0004, 
CONFBRIDGE(user,announce_join_leave)=no) in new stack
    -- Executing [2551@extensions:4] Set(SIP/199.73.108.66-0004, 
CONFBRIDGE(user,startmuted)=yes) in new stack
    -- Executing [2551@extensions:5] Set(SIP/199.73.108.66-0004, 
CONFBRIDGE(user,template)=church) in new stack
    -- Executing [2551@extensions:6] ConfBridge(SIP/199.73.108.66-0004, 
) in new stack
    -- Channel SIP/199.73.108.66-0004 joined 'softmix' base-bridge 
6dfaada2-2d01-44e4-b025-914ca18eee25
Bridge Enter message for channel SIP/199.73.108.66-0004: 1425186171.00965862
0x7fbbe8003aa8 - Updating Party A SIP/199.73.108.66-0004 snapshot
0x7fbbe8003aa8 - Processing bridge enter for SIP/199.73.108.66-0004
0x7fbbe8003aa8 - Transitioning CDR for SIP/199.73.108.66-0004 from state 
Single to Bridged
0x7fbbe8005048 - Created CDR for channel Local/2498@extensions-;1
0x7fbbe8005048 - Transitioning CDR for Local/2498@extensions-;1 from 
state NONE to Single
0x7fbbe8005048 - Set answered time to 1425186171.965933
0x7fbbe8005048 - Transitioning CDR for Local/2498@extensions-;1 from 
state Single to Bridged
0x7fbbe8005048 - Party A Local/2498@extensions-;1 has new Party B 
SIP/199.73.108.66-0004
0x7fbbe8001628 - Party A

Re: [asterisk-users] One mailbox for multiple extensions with individual greetings

2014-05-10 Thread John T. Bittner 
Why don't you use the voicemail copy feature?
Create 3 mailboxes 1234, 6789 and 2000 for the shared.

VoiceMail(1234@default2000@default,su)
VoiceMail(6789@default2000@default,su)

Set both 1234 and 6789 to email the voicemail to a fake email address and 
delete after email.
A copy of the message for each will be dropped into 2000 and deleted from the 
original box.


John


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Saturday, May 10, 2014 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] One mailbox for multiple extensions with individual 
greetings

Hi,

Is there a way in Asterisk 11 to use a single voicemailbox for multiple 
extensions while still hearing each extension's individual greeting?

Use case: someone has 2 numbers and wants all voicemail messages for both 
numbers to end up in one mailbox. So when dialing 1234 and NOANSWER you would 
hear the person at extension 1234 is unavailable and the message would be 
stored in mailbox mymailbox and when dialing 6789 and NOANSWER you would hear 
the person at extension 6789 is unavailable 
and that message would also be stored in mailbox mymailbox. The user then 
dials an extension to reach mymailbox and hears all messages for both the 1234 
and 6789 numbers.

I think I can solve it by symlinking
/var/spool/asterisk/voicemail/default/6789/INBOX to 
/var/spool/asterisk/voicemail/default/1234/INBOX (and the other directories 
too) but it would be nice if this could be done within the dialplan.

If that's not possible, would adding an extension option to app_voicemail.c 
solve this by decoupling the extension from the mailbox?

Thanks for any pointers.

Cheers,
Patrick

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Re: [asterisk-users] Asterisk Real-time Static Voicemail

2013-11-11 Thread John T. Bittner 
I am not running ODBC storage for Voicemail.

Just running Real-time time static for configurations.

John

===
;Do you have compiled asterisk by yourself? In the Voicemail Build Option, what 
option have you selected? I think you need to select ODBC Storage and then 
;configure ODBC on the system to connect to your database.;;
;;;
;Leandro



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[asterisk-users] Asterisk Realtime Static Voicemail

2013-11-10 Thread John T. Bittner 
Guys,

I need you help on this one.

Don't know when this broke but we have a custom gui that runs on top of 
Asterisk running a real-time static for configurations.
Nothing has changed with the database other than upgrades of Asterisk 10.

Customer complained that there password was not changing when they called into 
voicemail and changed it.
Database is running standard ast_config with the following fields.

++--+--+-+-++
| Field  | Type | Null | Key | Default | Extra  |
++--+--+-+-++
| id | int(11)  | NO   | PRI | NULL| auto_increment |
| cat_metric | int(11)  | NO   | | 0   ||
| var_metric | int(11)  | NO   | | 0   ||
| commented  | int(11)  | NO   | | 0   ||
| filename   | varchar(128) | NO   | | ||
| category   | varchar(128) | NO   | | default ||
| var_name   | varchar(128) | NO   | | ||
| var_val| varchar(255) | NO   | | ||
++--+--+-+-++
8 rows in set (0.00 sec)

Did some tests and asterisk does change the password but in the 
/etc/asterisk/voicemail.conf file.
Rename the file to see if it will then try the database. It recreates the file 
and changes the password.
The issue is when it reads the password it looks at ast_config so it never 
really changes.
Ran debug and no errors, I don't even see it trying to update mysql

Any idea what this could be.  The file below is an exact match of what's in 
ast_config.

/etc/asterisk/voicemail.conf
;! Automatically generated configuration file
;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
;! Generator: AppVoicemail
;! Creation Date: Mon Nov 11 01:12:51 2013
;!
[default]
9105 = 1234,Genee Jacobs,,,tz=|attach=|saycid=|hidefromdir=
201 = ,Anne Long,,,tz=|attach=|saycid=|hidefromdir=|delete=
[zonemessages]
pacific = US/Pacific|'vm-received' Q 'digits/at' IMp
eastern = America/New_York|'vm-received' Q 'digits/at' IMp
central = America/Chicago|'vm-received' Q 'digits/at' IMp
central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
gmt = Europe/London|'vm-received' q 'digits/at' H N 'hours'
cet = Europe/Zurich|'vm-received' q 'digits/at' H N 'hours'
hkg = Asia/Hong_Kong|'vm-received' q 'digits/at' H N 'hours'
[general]
format = wav49|gsm|wav
serveremail = nwvoicem...@randrealty.com
attach = yes
emaildateformat = %A, %B %d, %Y at %r
maxlogins = 3
sendvoicemail = yes
operator = yes
pagerdateformat = %A, %B %d, %Y at %r
externnotify = /usr/local/sigman/scripts/voicemailapp


John Bittner
CTO
[cid:image003.png@01CEDE7D.6B15EB80]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.nethttp://www.xaccel.net/

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] Hack

2013-10-18 Thread John T. Bittner 
Hi Steve,

Not using real-time.

John


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Friday, October 18, 2013 4:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hack

On 18 Oct 2013, at 04:06, John T. Bittner 
j...@xaccel.netmailto:j...@xaccel.net wrote:
Today I was hacked but caught it very quickly. This is the weird part, they 
hacked an IP Auth based account by simply knowing the account name.

How is this possible? I am running Asterisk 11.5.0. Now it's my fault I used a 
dictionary based account name but how did they bypass the set ip I had under 
the account for this host.

Did the IP show under sip show peer xxx? If it's realtime it's possible to set 
it and need to prune it / sip reload.

Steve



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[asterisk-users] Hack

2013-10-17 Thread John T. Bittner 
Today I was hacked but caught it very quickly. This is the weird part, they 
hacked an IP Auth based account by simply knowing the account name.

How is this possible? I am running Asterisk 11.5.0. Now it's my fault I used a 
dictionary based account name but how did they bypass the set ip I had under 
the account for this host.

This also happened with fail2ban running and I pay for Humbug . Nothing caught 
it. Its just chance that I happen to be in the CLI and noticed it.
In a span of 30 minutes they had made over $200 worth of calls all to the same 
number .

Anyone have any idea on this and any ideas on preventing this.



John Bittner
CTO
[cid:image003.png@01CECB8D.765B3840]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.nethttp://www.xaccel.net/

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] Endpoint call forwarding

2013-07-02 Thread John T. Bittner 
Anyone having issues with endpoint call forwarding on asterisk 11?

Was working perfect with 10. Issues are not phone related have tried cisco, 
polycom and Xlite, all fail.

Backtrack to 10 and it works ok again.

Any help is appreciated.

Thanks


John Bittner
CTO
[cid:image003.png@01CE76D7.8AB33690]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.nethttp://www.xaccel.net/

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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[asterisk-users] DTMF

2013-06-21 Thread John T. Bittner 
Anyone see this before?

I have a main Asterisk box 11.4 connected to Windstream via SIP trunks in my 
colo.
So as a did comes in they are routed to appropriate customers, in this case 
another asterisk 11.4 box.

All is working well with the exception of DTMF. Losing the last digits so say 
someone hits 123... on the customer box I only get 12
This is the weird part, it only happens on 1 DID. If I point another DID from 
the same carrier to the customer it works. Both the main asterisk box and 
customer are running real IP's (no nat)

Did a test and routed the DID to a IVR on my main system and I see the correct 
DTMF, so somewhere in the process of local bridging DTMF is getting messed up. 
Both carrier and customer were set to RFC2833.

Upgraded Asterisk 11.4 to 11.5 on both systems same issue.

Changed customer to DTMF info and it fixed it.

Could line level of a DID cause this?  I researched this a lot and find 
conflicting answers.

How does asterisk local bridge calls, does it affect the line levels. My 
understanding of RFC2833 is that line level issues should not affect it.



John Bittner
CTO
[cid:image003.png@01CE6E8B.C39E8B20]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.nethttp://www.xaccel.net/

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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