Re: [asterisk-users] voicemail message not deleted

[Michael Keuter]

> Am 26.07.2021 um 07:28 schrieb Fourhundred Thecat <400the...@gmx.ch>:
> 
> Hello,
> 
> I have this in my voicemail.conf:
> 
>  attach=yes
> 
>  delete=yes
> 
> I do get an email when new voicemail is received, and I do get the
> voicemail message as attachment.
> 
> However, the original message is not deleted from the sevber.
> 
> How do I delete the message, after it has been sent per email as
> attachment? I don't want to store messages on the server indefinitely.
> 
> thanks,
> 
> -- 

I think you need to set "delete=yes" as option per mailbox account. 

100 => 1234,Test,,,delete=yes

The global setting is only an example.

Michael

http://www.mksolutions.info




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Re: [asterisk-users] BLF support in Asterisk and early/confirmed/terminated/proceeding NOTIFY states.

[Michael Keuter]

> Am 03.10.2020 um 03:26 schrieb Jobst Schmalenbach :
> 
> I have a setup with Yealink phones & Asterisk Server (all latest patches).
> 
> I am using BLF to display the states of other phones. While this works MOST 
> of the time (busy, being called) it does NOT work when a phone is NOT 
> regisstered at all, the yealink phones display a green dot EVEN if a phone is 
> turned off (try explain this to users, they are shaking their heads!!!)
> 
> I can see on the Asterisk server it shows correctly in the logs when a phone 
> is disconnected.
> It also advises the otehr phones correctly when a phone is busy, even if a 
> person starts dialing - the red dot shows up milliseconds later on ALL of the 
> other phones.
> 
> I have asked a question about this before Green Dot
> 
> So I went and asked Yealink about this. The reply was something like this: 
> “Currently the phone can only support the BLF LED display with 
> early/confirmed/terminated/proceeding NOTIFY states.”
> 
> My question now is can I implement this properly on an Asterisk server?
> I.e. when a phone gets disconnected that all other phones are advised “hey I 
> am not available” and actually show a RED dot.
> thanks
> 
> -- 

Hi Jobst,

there is a setting for the Yealink phones so that the BLF keys are "off" 
instead of "green" when idle:

BLF LED Mode = 1

in the provisioning files:

features.blf_led_mode = 1

Hope that helps.

Michael

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Re: [asterisk-users] Voice broken during calls (again...)

[Michael Keuter]

You could also use the 'mtr' command under Linux.

> Am 22.06.2020 um 17:41 schrieb Marek Greško :
> 
> Hello,
> 
> try pinging your sip peer ip address following way:
> 
> ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress}
> 
> Post several lines and the statistics.
> 
> Were you also thinking about MTU problems? Not very probable, but one
> never knows.
> 
> Marek
> 
> 
> 2020-06-22 17:18 GMT+02:00, Luca Bertoncello :
>> Am 22.06.2020 um 17:01 schrieb Telium Technical Support:
>>> I don't know if there was a prior email with more details, but
>>> 
>>> Latency is as important as speed.  Have you checked latency between your
>>> device and pop?  What about QoS at your location, and does your ITSP
>>> support/respect QoS?
>> 
>> That's a very good idea...
>> Could you suggest me how can I check it?
>> The Gateway is a Linux with Debian 9.
>> 
>>> Could problem be inside your network?  Have you tested/optimized internal?
>> 
>> Really difficult to believe... If I call another VoIP-phone in my
>> network (using the "internal number") the quality is excellent.
>> 
>> If I call my wife using the "external number", the quality is very bad...
>> 
>> Thanks
>> Luca Bertoncello
>> (lucab...@lucabert.de)


Michael

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Re: [asterisk-users] Voice "broken" during calls

[Michael Keuter]

> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello :
> 
> Am 13.06.2020 um 22:56 schrieb Antony Stone:
> 
> Hi again,
> 
>> 2b. Take your Thomson telephone to some other location with Internet access, 
>> let it register to your home Asterisk server, and them make a call to the 
>> same 
>> number yet again.  I'm sure you can get the Thomson to connect to Asterisk 
>> via 
>> some external network, since you say you can do this from your Android 
>> phone.  
>> Again, check the call quality.
> 
> I tried it on the network of a friend.
> Not possible to establish a connection at all...
> I *suppose* Deutsche Telekom just allow a logon on their servers from
> the IP of the user, who tries to log on (with other words: my VoIP login
> can just log on from my current IP)...

Hi Luca,

the standard Deutsche Telekom SIP-account (former ISDN Mehrgeräteanschluß PTMP 
with 3-10 numbers) is always tied to your DSL account.

There is a special "DeutschlandLAN SIP-Trunk Pure" where it does not depend on 
your DSL account (as it is standard with most other VoIP providers).

> This would explain why I didn't got my mobile phone connecting to the
> Telekom's server and establish a call...
> 
> I also tried to stop Asterisk and all other network services on my
> Linux-Box Firewall/Gateway, including the traffic shaper (in the case,
> this was the problem), then connect my Thomson phone to the Telekom's
> server and call my father in law.
> Always the same problem...
> 
> So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290).
> I'll connect it to my network and my Asterisk and will try to call my
> father in law for a test.
> 
> I really do *not* expect any change in the situation... I think, the
> problem should be somewhere by Deutsche Telekom...
> 
> What is your opinion?
> 
> Btw: I did all tests with my father in law, since he had time for me
> today, but the problem exists an almost all calls, incoming or outgoing,
> no matter from/to which network provider...
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)

Michael

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Re: [asterisk-users] Voice "broken" during calls

[Michael Keuter]

So the call used Alaw as Codec.

> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello :
> 
> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
> 
> Hi
> 
>> Try "sip show peer " for a phone.
> 
> So:
> 
> mobile phone:
> bpi*CLI> sip show peer 0049177xxx
> 
> 
> 
> 
>  * Name   : 0049177xxx
> 
> 
>  Description  :
> 
> 
>  Secret   : 
> 
> 
>  MD5Secret: 
> 
> 
>  Remote Secret: 
> 
> 
>  Context  : default
> 
> 
>  Record On feature : automon
> 
> 
>  Record Off feature : automon
> 
> 
>  Subscr.Cont. : 
> 
> 
>  Language : de
> 
> 
>  Tonezone : 
>  AMA flags: Unknown
>  Transfer mode: open
>  CallingPres  : Presentation Allowed, Not Screened
>  Callgroup: 1
>  Pickupgroup  : 1
>  Named Callgr :
>  Nam. Pickupgr:
>  MOH Suggest  :
>  Mailbox  :
>  VM Extension : asterisk
>  LastMsgsSent : 0/0
>  Call limit   : 2147483647
>  Max forwards : 0
>  Dynamic  : Yes
>  Callerid : "0049177xxx" <>
>  MaxCallBR: 384 kbps
>  Expire   : -1
>  Insecure : no
>  Force rport  : Yes
>  Symmetric RTP: Yes
>  ACL  : No
>  DirectMedACL : No
>  T.38 support : Yes
>  T.38 EC mode : FEC
>  T.38 MaxDtgrm: 4294967295
>  DirectMedia  : No
>  PromiscRedir : No
>  User=Phone   : No
>  Video Support: No
>  Text Support : No
>  Ign SDP ver  : No
>  Trust RPID   : No
>  Send RPID: Yes
>  Path support : No
>  Path : N/A
>  TrustIDOutbnd: Legacy
>  Subscriptions: Yes
>  Overlap dial : No
>  DTMFmode : rfc2833
>  Timer T1 : 500
>  Timer B  : 32000
>  ToHost   :
>  Addr->IP : (null)
>  Defaddr->IP  : (null)
>  Prim.Transp. : UDP
>  Allowed.Trsp : UDP
>  Def. Username:
>  SIP Options  : (none)
>  Codecs   :
> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk)
>  Auto-Framing : No
>  Status   : UNKNOWN
>  Useragent:
>  Reg. Contact :
>  Qualify Freq : 6 ms
>  Keepalive: 0 ms
>  Sess-Timers  : Refuse
>  Sess-Refresh : uac
>  Sess-Expires : 1800 secs
>  Min-Sess : 90 secs
>  RTP Engine   : asterisk
>  Parkinglot   :
>  Use Reason   : No
>  Encryption   : No
> 
> VoIP-phone (Thomson ST2022):
> bpi*CLI> sip show peer 0049351xxx
> 
> 
> 
> 
>  * Name   : 0049351xxx
> 
> 
>  Description  :
> 
> 
>  Secret   : 
> 
> 
>  MD5Secret: 
> 
> 
>  Remote Secret: 
> 
> 
>  Context  : default
> 
> 
>  Record On feature : automon
> 
> 
>  Record Off feature : automon
>  Subscr.Cont. : 
>  Language : de
>  Tonezone : 
>  AMA flags: Unknown
>  Transfer mode: open
>  CallingPres  : Presentation Allowed, Not Screened
>  Callgroup: 1
>  Pickupgroup  : 1
>  Named Callgr :
>  Nam. Pickupgr:
>  MOH Suggest  :
>  Mailbox  :
>  VM Extension : asterisk
>  LastMsgsSent : 0/0
>  Call limit   : 2147483647
>  Max forwards : 0
>  Dynamic  : Yes
>  Callerid : "0049351xxx" <>
>  MaxCallBR: 384 kbps
>  Expire   : 3111
>  Insecure : no
>  Force rport  : Yes
>  Symmetric RTP: Yes
>  ACL  : Yes
>  DirectMedACL : No
>  T.38 support : Yes
>  T.38 EC mode : FEC
>  T.38 MaxDtgrm: 4294967295
>  DirectMedia  : No
>  PromiscRedir : No
>  User=Phone   : No
>  Video Support: No
>  Text Support : No
>  Ign SDP ver  : No
>  Trust RPID   : No
>  Send RPID: Yes
>  Path support : No
>  Path : N/A
>  TrustIDOutbnd: Legacy
>  Subscriptions: Yes
>  Overlap dial : No
>  DTMFmode : rfc2833
>  Timer T1 : 500
>  Timer B  : 32000
>  ToHost   :
>  Addr->IP : 192.168.200.10:25572
>  Defaddr->IP  : (null)
>  Prim.Transp. : UDP
>  Allowed.Trsp : UDP
>  Def. Username: 0049351xxx
>  SIP Options  : (none)
>  Codecs   : (alaw|ulaw|ilbc|g729|g723|gsm)
>  Auto-Framing : No
>  Status   : OK (17 ms)
>  Useragent: THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23
>  Reg. Contact : sip:0049351xxx@192.168.200.10:25572;user=phone
>  Qualify Freq : 6 ms
>  Keepalive: 0 ms
>  Sess-Timers  : Refuse
>  Sess-Refresh : uac
>  Sess-Expires : 1800 secs
>  Min-Sess : 90 secs
>  RTP Engine   : asterisk
>  Parkinglot   :
>  Use Reason   : No
>  Encryption   : No
> 
> 
>> Then "sip show cha

Re: [asterisk-users] Voice "broken" during calls

[Michael Keuter]

> Am 13.06.2020 um 13:36 schrieb Luca Bertoncello :
> 
> Am 13.06.2020 09:30, schrieb Luca Bertoncello:
> 
> Hi again (again)
> 
> I noticed right now another strange detail...
> I made a call using my mobile phone (connected to the Asterisk). The quality 
> was top...
> Maybe is the problem in a codec used from our phones at homes?
> Could someone suggest me how to check the codec used by my mobile phone and 
> the codec used by the phones at home?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)

Try "sip show peer " for a phone.
Then "sip show channels" during an existing call.
And "sip show channel " for more info.

Michael

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Re: [asterisk-users] MWI Delayed on Polycom VVX phones

[Michael Keuter]

> Am 15.01.2019 um 15:23 schrieb Doug Lytle :
> 
> Hi all,
> 
> When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has 
> resulted in a MWI clearing delay of around 5 minutes.
> 
> After listening to a voicemail and deleting it, the Polycom VVX 601's MWI 
> light is left on for around five minutes, before clearing.
> 
> Installing Asterisk 13.24.1 did not fix this.
> 
> Moving back to 13.23.1 allows the MWI to clear immediately.  I see a note in 
> the change logs for 13.24.0
> 
> [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead 
> of mailboxes=##@default
> 
> Any suggestions on what to look at to diagnose?
> 
> Doug

Hi Doug,

applying this patch helped in my case (with AstLinux 1.3.x + Asterisk 13.24.1):

https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17

I am using "mailboxes=##@default" and had the issue as well (before).

Michael

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Re: [asterisk-users] Best Asterisk Platform

[Michael Keuter]

Am 23.12.2015 um 14:31 schrieb er ic :

> What is the best asterisk platform to use? What are you guys using?
> 
> I am looking for something to host either in our data center or at the 
> customer prem where I have the control over the unit and not through a 
> contractor.
> 
> I dont mind paying a license fee for a front end interface but still would 
> rather not have to pay.
> 
> Thanks,
> --Eric

Hi Eric,

it depends on what you need :-).
For smaller installations (<100), and if you don't need a fancy GUI,
I can recommend AstLinux (Open Source), a complete communication plattform in 
one box (or VM):

http://www.astlinux.org

Michael

http://www.mksolutions.info





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Re: [asterisk-users] how can queue agents choose which call to answer?

[Michael Keuter]

Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee:

 Hi everybody,
 
 I'm looking for a solution for the following scenario:
 
 • Asterisk queue
 • At peak hours, there will be more callers then queue members/agents, so 
 some callers will spend some time on hold
 • Agents should be able to choose which of the on hold calls to answer 
 instead of answering the next one in queue
 
 We already have a web interface where agents can see the callers on hold, so 
 the best solution would be if they could just click a callers number to get 
 his call. But I have not found a way to tell Asterisk to do something to a 
 call on hold in a queue.
 
 Priority queues are not really an option, as the agents will be deciding on 
 the fly which caller is more important.
 
 I am not really sure if queues are the correct solution for this problem. 
 However, we have existing statistics built for queue logs, so it would be 
 really nice if the solution was queue-based.
 
 Thanks for any thoughts,
 
 -- 
 
 marie


Hello Marie,

maybe FOP2  [1] is an option for you. There you can visually pick up a call 
from a queue.
It's not open source though.

[1] http://www.fop2.com

Michael

http://www.mksolutions.info





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Re: [asterisk-users] How to implement priority queuing within a single queue ?

[Michael Keuter]

Am 25.01.2013 um 17:22 schrieb Olivier:

 Hi,
 
 Let say that in a call center, callers are recognized and categorized in 4 
 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, 
 and so on)  before entering a Queue.
 How can you make sure a priority 2 caller is answered before priority 3 
 callers, for instance ?
 
 I can think of several solutions but none really pleases me :
 
 1. Have 4 different queues, set penalty value and let each caller enter one 
 queue depending on its own priority.
 I don't like this solution because I foresee editing stats for 4 queues 
 instead of one is harder.

Just set the Queue_PRIO for that specific caller-type before you send them all 
into the same queue:

exten = s,n,Set(QUEUE_PRIO=10)
exten = s,n,Queue(test,tC,,,180)

 
 2. Iterate over each call waiting in the queue and insert new call with 
 Queue's position argument accordingly valued.
 I don't like this one because I'm afraid coding this won't be so easy.
 
 What would you suggest ?
 
 Regards
 
 
 
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Re: [asterisk-users] How to implement priority queuing within a single queue ? [SOLVED]

[Michael Keuter]

Am 25.01.2013 um 17:39 schrieb Olivier:

 
 
 2013/1/25 Michael Keuter li...@mksolutions.info
 
 Am 25.01.2013 um 17:22 schrieb Olivier:
 
  Hi,
 
  Let say that in a call center, callers are recognized and categorized in 4 
  priority levels (priority 1 for Very Very Important Personalities, 2 for 
  VIP, and so on)  before entering a Queue.
  How can you make sure a priority 2 caller is answered before priority 3 
  callers, for instance ?
 
  I can think of several solutions but none really pleases me :
 
  1. Have 4 different queues, set penalty value and let each caller enter one 
  queue depending on its own priority.
  I don't like this solution because I foresee editing stats for 4 queues 
  instead of one is harder.
 
 Just set the Queue_PRIO for that specific caller-type before you send them 
 all into the same queue:
 
 exten = s,n,Set(QUEUE_PRIO=10)
 exten = s,n,Queue(test,tC,,,180)
 
 That's exactly what I was looking for.
 The strange thing is I couldn't find it mentioned in Queue app doc, if I'm 
 not mistaken (but that's another story).
 
 Thank you very much. 

I found it here a while ago:
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
https://wiki.asterisk.org/wiki/display/AST/Various+application+variables

Michael

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Re: [asterisk-users] Is there a need to secure RTP ports?

[Michael Keuter]

Am 23.01.2013 um 18:33 schrieb Carlos Alvarez:

 On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk wrote:
 I have an Asterisk server with one SIP trunk to a SIP provider. As my server 
 registers with the SIP provider, I don't have any SIP ports open at my end to 
 the Internet. However, I have the RTP ports open (as SIP has some trouble 
 with my NAT). My question is - what are the vulnerabilities in this scenario 
 at my end? I suppose some man-in-the-middle or eavesdropping  attack is 
 always a possibility - but that aside, is there anything that will attack RTP 
 ports on Asterisk when there are no SIP ports open? I was looking into 
 installing fail2ban - until I realised that there is no SIP port exposed for 
 an attacker to poke at.
 
 I've been working in IP telephony for about ten years.  I've never once heard 
 of any attack on the RTP ports.  While you can never say anything is 
 impossible there's simply nothing listening on those ports.  It's probably 
 possible to have a DOS attack where someone starts sending RTP to all of your 
 ports and they would interfere with a call, but they couldn't do more than 
 that.  That could work if your router has full cone NAT and a lot of other 
 things fall into place.  Still kind of out there as a real threat.
 
 
 -- 
 Carlos Alvarez
 TelEvolve
 602-889-3003

2 years ago someone demonstrated on the 27C3 in Berlin some interstings things 
you can do with RTP:
http://media.ccc.de/browse/congress/2010/27c3-4193-en-having_fun_with_rtp.html

(use the original file)

Michael

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Re: [asterisk-users] Asterisk 11 - Security event logging over syslog

[Michael Keuter]

Am 24.11.2012 um 11:25 schrieb Michael Keuter:

 
 Am 24.11.2012 um 01:33 schrieb Matthew Jordan:
 
 On 11/22/2012 04:00 PM, Michael Keuter wrote:
 Hi all,
 
 I am just testing with Asterisk 11.01.
 The SIP security event logging works fine for me for console and file 
 logging, but the security events are not logged over syslog.
 
 logger.conf:
 ...
 syslog.local0 = notice,warning,error,security
 
 Is this on purpose, a fault on my side, or is this a bug? 
 
 
 No, that should work.  What's the output of 'logger show channels'?
 
 -- 
 Matthew Jordan
 
 logger show channels 
 Channel Type StatusConfiguration
 ---  ---
 syslog.local0   Syslog   Enabled- NOTICE WARNING 
 ERROR SECURITY 
 /var/log/asterisk/security_log  File Enabled- SECURITY 
Console  Enabled- NOTICE WARNING ERROR 
 SECURITY 
 
 Everything else except security works fine over syslog.
 
 Michael
 
 http://www.mksolutions.info


I created an issue on the bugtracker for this:
https://issues.asterisk.org/jira/browse/ASTERISK-20744

Michael

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Re: [asterisk-users] Asterisk 11 - Security event logging over syslog

[Michael Keuter]

Am 24.11.2012 um 01:33 schrieb Matthew Jordan:

 On 11/22/2012 04:00 PM, Michael Keuter wrote:
 Hi all,
 
 I am just testing with Asterisk 11.01.
 The SIP security event logging works fine for me for console and file 
 logging, but the security events are not logged over syslog.
 
 logger.conf:
 ...
 syslog.local0 = notice,warning,error,security
 
 Is this on purpose, a fault on my side, or is this a bug? 
 
 
 No, that should work.  What's the output of 'logger show channels'?
 
 -- 
 Matthew Jordan

logger show channels 
Channel Type StatusConfiguration
---  ---
syslog.local0   Syslog   Enabled- NOTICE WARNING ERROR 
SECURITY 
/var/log/asterisk/security_log  File Enabled- SECURITY 
Console  Enabled- NOTICE WARNING ERROR 
SECURITY 

Everything else except security works fine over syslog.

Michael

http://www.mksolutions.info






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[asterisk-users] Asterisk 11 - Security event logging over syslog

[Michael Keuter]

Hi all,

I am just testing with Asterisk 11.01.
The SIP security event logging works fine for me for console and file logging, 
but the security events are not logged over syslog.

logger.conf:
...
syslog.local0 = notice,warning,error,security

Is this on purpose, a fault on my side, or is this a bug? 

Michael

http://www.mksolutions.info






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Re: [asterisk-users] Intruder

[Michael Keuter]

Am 16.11.2012 um 18:08 schrieb Michael L. Young:

 - Original Message - 
 
 From: Felix Vazquez felix.vazq...@theboshgroup.com
 To: asterisk-users@lists.digium.com
 Sent: Friday, November 16, 2012 11:20:46 AM
 Subject: [asterisk-users] Intruder
 
 I am in the asterisk CLI and can see an unidentified caller trying
 the make calls out of the asterisk system. How do I stop them? How
 do I identify them and how can I see how the go in?
 
 This is an example of what I would see:
 
 NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call from '' to
 extension '90111235551212' rejected because extension not found.
 
 I would recommend you read README-SERIOUSLY.bestpractices.txt, top level of 
 source code.
 
 Another thing you can do is turn on security logging if you are using 
 Asterisk 10/11.  Take a look at logger.conf.  It may provide you with some 
 extra information on who is trying to make the call.
 
 Take a look at this page:
 https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
 
 I would recommend using fail2ban as well.
 
 Michael
 (elguero)


Hi Michael,

the security logging in Asterisk 11 was a nice tip. 
I tried it, but unfortunately it doesn't work over syslog for me, only console 
and file logging.
Do you know if that is on purpose?

In AstLinux we have our own kind of Fail2ban solutions which parses the syslog.

Michael

http://www.mksolutions.info






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Re: [asterisk-users] Asterisk 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP

[Michael Keuter]

Am 21.06.2012 um 18:05 schrieb Richard Mudgett:

 My previous message was incomplete.
 
 
 On thing to note is I had to forbid hfcmulti in modprobe.d in the
 second box to comply with a warning from dahdi. Without that, I could
 see this line in the output of lsmod:
 mISDN-core  hfcmulti
 
 
 1. What is the root cause that makes a board change its sync source ?
 How can I check this ?
 
 I would think layer 1 going down.  Many European telcos for BRI PTMP lines
 drop layer 2 and then layer 1 to conserve power.
 
 Is the switching of clock sources causing a problem?
 
 2.  How can I get rid of these alarms ?
 
 See the chan_dahdi.conf.sample file about the following options.
 
 You could use the layer1_presence option to make Asterisk ignore those
 alarms.
 
 You could use the layer2_persistence option to keep layer 2 up.  To use
 this option however, requires using libpri SVN 1.4 branch code as current
 released versions do not support the option.  Using the layer2_persistence
 option restores behavior that was removed for better Q.921 conformance for
 PTMP after libpri v1.4.10.2 and is why you are seeing a behavior difference
 between versions.

Hi Richard,

any plans when libpri 1.4.13 will be released?

Michael

http://www.mksolutions.info






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Re: [asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

[Michael Keuter]

I recently bought 2 Beronet BN4S0 cards on eBay, each for under 100 €.
they have 4 S0 ports.

Sent from my iPad

Michael

Am 29.05.2012 um 14:18 schrieb Kevin P. Fleming kpflem...@digium.com:

 On 05/29/2012 01:48 AM, Michelle Konzack wrote:
 No, it does not fit, since PCI 2.0 is 5V and has only one notch.
 
 PCI 2.1, 2.2. and 2.3 do have two notches, because they are 3.3V.
 
 In clear, you can not insert old 5V PCI 2.0 cards into a 3.3V PCI-X slot
 
 Ahh, your real issue is voltage then, not the PCI specification that the card 
 is compliant with. Cards can be compliant with any of the PCI versions you 
 mentioned and still be 5V only, 3.3V only, or 5V/3.3V compatible.
 
 All modern ISDN BRI cards usable with Asterisk are both 5V and 3.3V 
 compatible, but as you say, they aren't available in your price range.
 
 -- 
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] looking for solid state like PC suitable for Asterisk

[Michael Keuter]

Am 10.05.2012 um 15:03 schrieb Bart Coninckx:

 There seems to even be a 1.6 Ghz Intel Atom device.
 One site I'm looking to use this for has about 40 SIP phones and three BRIs. 
 It's always a guessing game whether  devices like this are up for that.
 If they do have some processing power, I might even consider combining them 
 as a highly available Asterisk cluster (using DRBD and Pacemaker).
 
 Anyone 2 cents about that?
 
 BC


Hi, we have in AstLinux special configuration tips for these DualCore Atom 
boards:

http://doc.astlinux.org/userdoc:board_jetway_nf96fl-525

Michael

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Re: [asterisk-users] Pickup calls coming from queues

[Michael Keuter]

Am 23.01.2012 um 23:25 schrieb Alec Davis:

 
 How can I test this solution on a 1.8.8.1 system ?
 If I'm not mistaken, diff 
 https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
 
 I've just checked out 1.8.8.1 and download my patch from
 https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean,
 using the following on a debian lenny box:
 
 svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 asterisk-1.8.8.1
 cd asterisk-1.8.8.1
 wget --no-check-certificate
 https://reviewboard.asterisk.org/r/1619/diff/raw/
 mv index.html r1619.diff.txt
 patch -p0  r1619.diff.txt
 
 Alec

Hi Alec,

that patch works fine for me in 1.8.9.0-rc3 too. Thanks a lot.
Is there any chance that this would be included into the 1.8 branch?

Michael

http://www.mksolutions.info





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Re: [asterisk-users] Pickup calls coming from queues

[Michael Keuter]

Am 25.01.2012 um 20:24 schrieb Alec Davis:

 Great to hear it's working for others.
 
 Regarding inclusion into 1.8 branch, as it's a new feature, it would only
 ever go into trunk, unless there is an outcry from the community.

Outcry! :-)

 To assist others implementing this, and from a different viewpoint, would
 you mind documenting how you implemented it.

Sure, I did it slightly different than you, because in your way the only Hint 
worked fine for me, but Pickup not (I guess because of my 
'notifycid=ignore-context' in sip.conf.

I put all into my [test] context, but the hint in my standard hint context 
[blf] and used PICKUPMARK:

;; Queue Pickup with Hint
:;exten = 8501,hint,Queue:itg_queue ;Provide a hint for the queue = 
in [blf]
exten = _*98501,1,Pickup(itg@blf)  ;Pickup the queue
exten = 8501,1,Set(__PICKUPMARK=8501)
exten = 8501,n,Queue(itg_queue,crhH,,,127) ;Ring the queue

[blf]
exten = 8501,hint,Queue:itg_queue

 I'm sure I've over complicated my examples.
 
 Alec
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
 Michael Keuter
 Sent: Thursday, 26 January 2012 2:23 a.m.
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Pickup calls coming from queues
 
 
 Am 23.01.2012 um 23:25 schrieb Alec Davis:
 
 
 How can I test this solution on a 1.8.8.1 system ?
 If I'm not mistaken, diff
 https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
 
 I've just checked out 1.8.8.1 and download my patch from 
 https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied 
 clean, using the following on a debian lenny box:
 
 svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 
 asterisk-1.8.8.1 cd asterisk-1.8.8.1 wget --no-check-certificate 
 https://reviewboard.asterisk.org/r/1619/diff/raw/
 mv index.html r1619.diff.txt
 patch -p0  r1619.diff.txt
 
 Alec
 
 Hi Alec,
 
 that patch works fine for me in 1.8.9.0-rc3 too. Thanks a lot.
 Is there any chance that this would be included into the 1.8 branch?
 
 Michael


Michael

http://www.mksolutions.info





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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Michael Keuter]

Am 05.01.2012 um 04:55 schrieb Matt Darnell:

 On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
 da...@klaverstyn.com.au wrote:
 I'm using  the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and 
 rx_fax on multiple installations with no problems.
 
 David,
 
 Are you running 10.0 or 1.8?
 
 Glad to know that the PAP2T has a solid T.38 implementation!
 
 -Matt

There seem to be at least 2 versions of the PAP2T. The one I have (in Germany) 
does NOT support T.38.

Michael

http://www.mksolutions.info





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