Re: [asterisk-users] More testing - sorry guys

2018-03-28 Thread Markus Weiler 

I received it :-)


Am 28.03.2018 um 22:44 schrieb Matt Fredrickson:

Just a test.




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[asterisk-users] Half Off Topic Questions

2018-03-06 Thread Markus Weiler 
Hi Group,

we're just wondering, in German we call the different types of phone-numbers 
(Geographic,mobile,national,VoIP...)
Rufnummerngassen (phone number alleys ;-) )  
Is there an english word for this? 

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  markus_wei...@mailworks.org
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[asterisk-users] res_json

2018-01-10 Thread Markus Weiler 

Hi All,

this seems to be a really neat module, that could really help us.

https://github.com/drivefast/asterisk-res_json

Any opinions about if we should use it in a production system?

Maybe from "official" asterisk side?

thanks!!

Markus




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Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Markus Weiler 

Hi Derek,

I think Homer (http://sipcapture.org/) is the right answer :-)

HEP Agent will send the SIP trace to a remote Server (res_hep).


Markus


Am 18.02.2017 um 00:18 schrieb Tim Pozar:

You can tell it to just capture SIP traffic and not the RTP traffic.
Nice write up of using TCPdump and wireshark can be found here:

https://blog.flowroute.com/2014/04/10/how-to-capture-sip-packets/

BTW, I have found this works really well in trying to debug RTP traffic
as well.  Wireshark just does the right thing in putting audio back
together.  Very helpful in tracking down in and out of band DTMF
problems that we were having with various carriers.

Tim

On 2/17/17 3:07 PM, Derek Andrew wrote:

The SIP trace will be adequate but this is on a remote system with
limited disk space.

I would love to turn on debugging while making the troublesome calls,
then turn it off afterward.

Tcpdump is great, but starting it and stopping it and keeping all that
data would still be an issue.

d

On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar > wrote:

 Why not capture the packets with something like tcpdump and run it
 through Wireshark?

 Tim

 On 2/17/17 2:43 PM, Derek Andrew wrote:
 > I have some troublesome numbers that I would like to capture the SIP
 > dialogue when I am calling them. When I am about to dial the
 number, is
 > there any way to turn on SIP debugging in the dial plan before I make
 > the call? (and turn it off after the call is completed?)
 >
 >
 >
 >
 >

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Typed but not read.








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Re: [asterisk-users] Issue with handling of 480 DND

2017-01-06 Thread Markus Weiler 

Nobody any idea?

It would be really helpful,

Markus



Am 06.01.2017 um 12:07 schrieb Markus Weiler:

Hi List,

we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus

Stripped down example;

exten = 494X,n,Dial(SIP/4120089,15,w)
exten = 494X,n,Goto(98-${DIALSTATUS},1)
exten = 494X,n,Hangup()


.
exten = 98-BUSY,1,NoOp(Busy)
exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))

exten = 98-NOANSWER,1,NoOp(noanswer)
exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))


Íf the phone call times out, the call is sent to 98-NOANSWER and then
answered as expected.
If the User presses DND on his phone the call is sent to 98-BUSY which
is identical but then the call is hung up. This behaviour is
unexpected/unwanted.

We tried to figure out what the difference is and think it's how
Asterisk handles the "480 Do Not Disturb" from the phone
(xxx.xxx.xxx.xxx).
It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
is being forwarded".

Is this a bug or a feature? :-) How could we handle this correctly?

SIP and Asterisk debug log below. Any help would be appreciated!

Markus

SIP:

#
U 2017/01/06 11:38:29.515836 xxx.xxx.xxx.xxx:45731 ->
yyy.yyy.yyy.yy:5060
SIP/2.0 480 Do Not Disturb.
v: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport=5060.
f: "0160XXX" <sip:0160...@yyy.yyy.yyy.yy>;tag=as4ef364e1.
t: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
i: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 INVITE.
User-Agent: snom760/8.7.5.13.
m: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;reg-id=1.
l: 0.
.

#
U 2017/01/06 11:38:29.516045 yyy.yyy.yyy.yy:5060 ->
xxx.xxx.xxx.xxx:45731
ACK sip:4120089@192.168.178.70:45731;line=8lln9qsq SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport.
Max-Forwards: 70.
From: "0160XXX" <sip:0160...@yyy.yyy.yyy.yy>;tag=as4ef364e1.
To: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
Contact: <sip:0160...@yyy.yyy.yyy.yy:5060>.
Call-ID: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 13.12.1.
Content-Length: 0.
.

#
U 2017/01/06 11:38:29.516166 yyy.yyy.yyy.yy:5060 -> zzz.zzz.zzz.zzz:5060
SIP/2.0 181 Call is being forwarded.
Via: SIP/2.0/UDP
zzz.zzz.zzz.zzz;branch=z9hG4bK120a.dcdd311bf80a9536d4eec1ce380f66a7.0;received=zzz.zzz.zzz.zzz.
Via: SIP/2.0/UDP 93.189.169.102:5060;branch=z9hG4bK4ec14865.
Record-Route: <sip:zzz.zzz.zzz.zzz;lr=on;ftag=as47cd4bd4>.
From: "49160XXX" <sip:49160XXX@93.189.169.102>;tag=as47cd4bd4.
To: <sip:494xx...@zzz.zzz.zzz.zzz>;tag=as0823beee.
Call-ID: 15f8ee0e657ba8687db29cff0093c95b@93.189.169.102:5060.
CSeq: 102 INVITE.
Server: Asterisk PBX 13.12.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:494xx...@yyy.yyy.yyy.yy:5060>.
Content-Length: 0.
.



Asterisk Debug:



[Jan  6 11:38:29] VERBOSE[5383][C-000473c5] pbx.c: Executing
[494X@4120:7] Dial("SIP/SER_IB-0004cd6d", "SIP/4120089,15,w") in
new stack
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Asked to create a
SIP channel with formats: (alaw)
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Allocating new SIP
dialog for 0dac947057f4ddc613b1391948770...@yyy.yyy.yyy.yy:5060 - INVITE
(No RTP)
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: Using engine
'asterisk' for RTP instance '0x7fae14073678'
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Allocated
port 14098 for RTP instance '0x7fae14073678'
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: RTP instance
'0x7fae14073678' is setup and ready to go
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] acl.c: Not an IPv4 nor IPv6
address, cannot get port.
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: Splitting 'pbx6'
into...
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: ...host 'pbx6' and
port ''.
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Setup RTCP
on RTP instance '0x7fae14073678'
[Jan  6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
TOS bits 184
[Jan  6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
CoS mark 5
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] acl.c: For destination
'xxx.xxx.xxx.xxx', our source address is 'yyy.yyy.yyy.yy'.
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting
AST_TRANSPORT_UDP with address yyy.yyy.yyy.yy:5060
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan  6 11:38:29] DEBUG[5383][C-0004

[asterisk-users] Issue with handling of 480 DND

2017-01-06 Thread Markus Weiler 
Hi List,

we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus

Stripped down example;

exten = 494X,n,Dial(SIP/4120089,15,w)
exten = 494X,n,Goto(98-${DIALSTATUS},1)
exten = 494X,n,Hangup()


.
exten = 98-BUSY,1,NoOp(Busy)
exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))

exten = 98-NOANSWER,1,NoOp(noanswer)
exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))


Íf the phone call times out, the call is sent to 98-NOANSWER and then
answered as expected.
If the User presses DND on his phone the call is sent to 98-BUSY which
is identical but then the call is hung up. This behaviour is
unexpected/unwanted.

We tried to figure out what the difference is and think it's how
Asterisk handles the "480 Do Not Disturb" from the phone
(xxx.xxx.xxx.xxx).
It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
is being forwarded". 

Is this a bug or a feature? :-) How could we handle this correctly?

SIP and Asterisk debug log below. Any help would be appreciated!

Markus

SIP:

#
U 2017/01/06 11:38:29.515836 xxx.xxx.xxx.xxx:45731 ->
yyy.yyy.yyy.yy:5060
SIP/2.0 480 Do Not Disturb.
v: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport=5060.
f: "0160XXX" ;tag=as4ef364e1.
t: ;tag=0380h4r478.
i: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 INVITE.
User-Agent: snom760/8.7.5.13.
m: ;reg-id=1.
l: 0.
.

#
U 2017/01/06 11:38:29.516045 yyy.yyy.yyy.yy:5060 ->
xxx.xxx.xxx.xxx:45731
ACK sip:4120089@192.168.178.70:45731;line=8lln9qsq SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport.
Max-Forwards: 70.
From: "0160XXX" ;tag=as4ef364e1.
To: ;tag=0380h4r478.
Contact: .
Call-ID: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 13.12.1.
Content-Length: 0.
.

#
U 2017/01/06 11:38:29.516166 yyy.yyy.yyy.yy:5060 -> zzz.zzz.zzz.zzz:5060
SIP/2.0 181 Call is being forwarded.
Via: SIP/2.0/UDP
zzz.zzz.zzz.zzz;branch=z9hG4bK120a.dcdd311bf80a9536d4eec1ce380f66a7.0;received=zzz.zzz.zzz.zzz.
Via: SIP/2.0/UDP 93.189.169.102:5060;branch=z9hG4bK4ec14865.
Record-Route: .
From: "49160XXX" ;tag=as47cd4bd4.
To: ;tag=as0823beee.
Call-ID: 15f8ee0e657ba8687db29cff0093c95b@93.189.169.102:5060.
CSeq: 102 INVITE.
Server: Asterisk PBX 13.12.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: .
Content-Length: 0.
.



Asterisk Debug:



[Jan  6 11:38:29] VERBOSE[5383][C-000473c5] pbx.c: Executing
[494X@4120:7] Dial("SIP/SER_IB-0004cd6d", "SIP/4120089,15,w") in
new stack
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Asked to create a
SIP channel with formats: (alaw)
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Allocating new SIP
dialog for 0dac947057f4ddc613b1391948770...@yyy.yyy.yyy.yy:5060 - INVITE
(No RTP)
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: Using engine
'asterisk' for RTP instance '0x7fae14073678'
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Allocated
port 14098 for RTP instance '0x7fae14073678'
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: RTP instance
'0x7fae14073678' is setup and ready to go
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] acl.c: Not an IPv4 nor IPv6
address, cannot get port.
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: Splitting 'pbx6'
into...
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: ...host 'pbx6' and
port ''.
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Setup RTCP
on RTP instance '0x7fae14073678'
[Jan  6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
TOS bits 184
[Jan  6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
CoS mark 5
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] acl.c: For destination
'xxx.xxx.xxx.xxx', our source address is 'yyy.yyy.yyy.yy'.
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting
AST_TRANSPORT_UDP with address yyy.yyy.yyy.yy:5060
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: SIP call-id
changed from '0dac947057f4ddc613b1391948770...@yyy.yyy.yyy.yy:5060' to
'7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060'
[Jan  6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: *** Our native
formats are (alaw)
[Jan  6 11:38:29]

Re: [asterisk-users] asterisk server stress test

2015-08-20 Thread Markus Weiler 

Am 20.08.2015 um 03:16 schrieb Pete Mundy:


Ah cr@p, sorry Steve, didn't mean to top-post there.


On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org 
mailto:markus_wei...@mailworks.org wrote:
We started the 500 calls and used milliwatt app on the first and 
record on the second host to check the quality. Alternatively just 
start 500+ calls and call yourself on top. So you can get a good 
idea how the quality is.


Markus

That's a fascinating concept!

Can you share any more about how you appraised the data and determined 
your results?


ie once you had the recordings on the second host what did you do do 
computationally score them? Do you look at the decoded (1khz?) 
waveform or do you appraise in another way?


Pete







Hi Pete,

we used different approaches.

Just to test the maximum channels a gateway can process the two Methods 
are enough, you can either listen to the Recordings or look at the waveform.
The easiest approach is to call a colleague and gradually increase the 
calls on the machine.


For systematic, continuous analysis Voipmonitor is a very useful tool.
We directed the traffic to a mirroring port on the Switch to which we 
connected a Server running Voipmon. (http://www.voipmonitor.org/)
Voipmon records the call and rates its quality. You can check the 
results either using the commercial Web Interface (test for free) or 
query the mysql DB.
Unfortunately Voipmon tends to crash on a regular basis (at least when 
we used it), but it's an awesome tool.
The underlying tool pcapsipdump is running a lot more stable, but you 
need to put a lot more work into it to get started.


hope i could help

Markus





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Re: [asterisk-users] asterisk server stress test

2015-08-19 Thread Markus Weiler 

Am 19.08.2015 um 19:07 schrieb Steve Edwards:

Please don't top post.

On Wed, 19 Aug 2015, James Cass wrote:


Steve, would you be willing to share that quick bash script?


There's no magic in the script, but here it is, embarrassing myself:

cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10

Here's what's wrong with this snippet:

1) I don't know why I chmod the 'template.' No idea whatsoever. 
Alcohol may have been involved.


2) I hate single character variable names. I love alcohol.

3) cp is ill advised. For a testing script, it was easy. For a 
production application, use mv.


In use, I would execute it specifying how many call files to create, 
like 50. Then, take a look at top, iftop, and vmstat. Lather, rinse, 
repeat to get to your goal.




We started the 500 calls and used milliwatt app on the first and record 
on the second host to check the quality. Alternatively just start 500+ 
calls and call yourself on top. So you can get a good idea how the 
quality is.


Call-Files are explained on 
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out


Markus

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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

2015-08-11 Thread Markus Weiler 

Hi Stefan,

we ran into a similar problem using Debian.

There we are able to check the current limits using:

pidof asterisk  - 23351
cat /proc/23351/limits

Output:
Limit Soft Limit   Hard Limit Units
Max open files1024 1024 files


I think that in the end

/etc/security/limits.conf

* hardnofile  50
* softnofile  50
root  hardnofile  50
root  softnofile  50


did the trick. We also tried

vi /etc/sysctl.conf
fs.file-max = 50

not sure what the solution in the end was. But I remember rebooting was 
important.



Markus



Am 11.08.2015 um 11:00 schrieb Stefan Viljoen:

Anybody else ran into this?

No, but I would ask myself why so many file descriptors are being used.
It sounds like you have a file descriptor leak (not being closed when
finished with).

Hi Tony

Thanks for replying.

I suspected something like that, though repeatedly running

lsof | wc -l

Always stays quite low - 100 000 open files, which is still 8 times less
than the system maximum as confirmed by running ulimit -n

I also note that this number will increase to about 125 000 but never go
higher than that, then, as calls hang up, decreate again - during times when
the CLI is spammed with 100s of broken pipe errors due to insuffiecient
file descriptors, this number never reaches beyond 125  000 out of the
available 800 000 open files.


You might also want to look at the output of lsof (or at least some of it)
to see what all these file descriptors are pointing to, and whether it is
indeed Asterisk that is consuming them.

If I grep by asterisk on the output of lsof the few thousand lines I have
looked at all seem to indicate legitimate uses - there are at least two
files for each conversation in progress (I assume for inward and outward
RTP) plus one for each file being mixmonitored (which also seems logical)
and also number-of-active-calls connections to res_timing_dahdi - which all
looks correct...


If it is Asterisk, it's quite possible, even probable, that such a leak
has been found and fixed, even in the 1.8 series. 1.8.11.0 is rather old -
the latest is 1.8.32.3, so it would be best to update to that version and
see if the problem persists.

Ok, I will have to consider that. The thing is the problem is not consistent
- I can (for example) run 60 calls, with no problems and no reported
failures in opening files, then calls will -decrease- to about 40 and then
later spike to 70, but around 50 calls I get the errors coming up thousands
of times in the CLI, then suddenly stop as the calls -increase- which
doesn't make sense. But this kind of behaviour does seem consistent with a
possible leak.

SOMETHING NEW

I have now ran

/usr/bin/prlimit --pid `pidof asterisk`

and I have noticed that even though I have 800 000 files specified, the
ACTUAL limit in place on Asterisk for numbers of files is only 1024?!

# prlimit --pid `pidof asterisk`
RESOURCE   DESCRIPTION SOFT  HARD UNITS
AS address space limitunlimited unlimited bytes
CORE   max core file size unlimited unlimited blocks
CPUCPU time   unlimited unlimited seconds
DATA   max data size  unlimited unlimited bytes
FSIZE  max file size  unlimited unlimited blocks
LOCKS  max number of file locks held  unlimited unlimited
MEMLOCKmax locked-in-memory address space 65536 65536 bytes
MSGQUEUE   max bytes in POSIX mqueues819200819200 bytes
NICE   max nice prio allowed to raise 0 0
NOFILE max number of open files1024  4096
NPROC  max number of processes30861 30861
RSSmax resident set size  unlimited unlimited pages
RTPRIO max real-time priority 0 0
RTTIME timeout for real-time tasksunlimited unlimited microsecs
SIGPENDING max number of pending signals  30861 30861
STACK  max stack size   8388608 unlimited bytes

Accordingly I have put this into a cronjob ran each minute:

prlimit --pid `pidof asterisk` --nofile=786000:786000

to try and force the running binary to keep a high file limit (sources say
to keep it less than the ACTUAL system file limit, in my case 800 000 files)
on the live Asterisk process.

I'll see if this maybe helps - the above runs via cron each minute.

So it appears for some reason somehow the live running asterisk process
loses track of how many open files it may have, or when it starts it
somehow does not start with the correct number of maximum open files, as set
in the system / kernel config?

Anyway, thank you for replying, I'll monitor this new Cronjob fixup I'm
trying and see if it helps.

No wonder it is complaining about running out of file handles if it ACTUALLY
was only

Re: [asterisk-users] asterisk email to fax

2015-06-25 Thread Markus Weiler 

Hi Eric,

This is not really easy. Especially the Mail2Tiff Conversion is tricky 
(lots of different MIME/File formats).
When you have the correct tiff file the rest ist easy. Try to narrow it 
down to an empty Mail Body using (one) PDF attachment.

We used a self written Java app to prepare the files.

Markus


Am 25.06.2015 um 18:09 schrieb Eric Cooper:

On Wed, Jun 24, 2015 at 08:15:06AM +0200, tux john wrote:

i would like to add email to fax functionality to the system. could someone
point me to the right direction to see how please?

I don't have a general solution, since I haven't needed to send faxes
recently.  But I did set up my dialplan so that I could test that it
works.

I used this:

 ;; For outgoing faxes
 FAX_NAME = My Company Name
 FAX_NUMBER = My Phone Number
 FAX_FILE = /tmp/fax.tiff

 [services]
 ...
 exten = sendfax,1,Verbose(3,Sending fax)
   same = n,Set(FAXOPT(headerinfo)=${FAX_NAME})
   same = n,Set(FAXOPT(localstationid)=${FAX_NUMBER})
   same = n,SendFax(${FAX_FILE})
   same = n,Verbose(3,Pages: ${FAXOPT(pages)}  Status: ${FAXOPT(status)})
   same = n,System(rm ${FAX_FILE})
   same = n,Hangup()


To dial the sendfax extension, I created a call file manually that
looked like this:
 Channel: DAHDI/g0/1-800-123-4567
 Context: services
 Extension: sendfax
In an automated system, you'd probably want to use the manager API,
but you could also generated call files from scripts.  Then I placed
the fax file in /tmp/fax.tiff and the call file in
/var/spool/asterisk/outgoing/ and Asterisk sent off the fax.  There
are some free fax echo numbers out there that are useful when
debugging.

If I wanted to drive this automatically from email, I would probably
just use procmail and some scripts.  Ghostscript produces the right
kind of image file if you use -sDEVICE=tiffg4.  You'd have to define
some syntax for specifying the outgoing number and cover page info in
the email body, and extract the MIME attachment for the document from
the email.

Hope this helps.

--
Eric Cooper e c c @ c m u . e d u




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Re: [asterisk-users] German sounds on Asterisk

2015-06-14 Thread Markus Weiler 

great,

would be the ideal time to comment the www.voip-info.org to contribute 
to the community :-)


Markus


Am 14.06.2015 um 09:42 schrieb Luca Bertoncello:

Markus Weiler markus_wei...@mailworks.org schrieb:

Hi


from voipinfo...

If an Asterisk command specifies a sound file in a*subdirectory*,
Asterisk looks in that subdirectory for the language subdirectory. For
example, theSayDigits
http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigitscommand may
play the sound file digits/6. Asterisk will, if the language code is
de, first look for /var/lib/asterisk/sounds/*digits/de/*6.gsm before
falling back to /var/lib/asterisk/sounds/digits/6.gsm.

Of course I read this page, but it does NOT work so...
Right now I got it, using a new structure:

/var/lib/asterisk/sounds/de/
/var/lib/asterisk/sounds/de/digits
/var/lib/asterisk/sounds/de/letters
/var/lib/asterisk/sounds/de/phonetics

and it works...

Regards
Luca Bertoncello
(lucab...@lucabert.de)




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Re: [asterisk-users] German sounds on Asterisk

2015-06-14 Thread Markus Weiler 

Hi,

from voipinfo...

If an Asterisk command specifies a sound file in a*subdirectory*, 
Asterisk looks in that subdirectory for the language subdirectory. For 
example, theSayDigits 
http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigitscommand may 
play the sound file digits/6. Asterisk will, if the language code is 
de, first look for /var/lib/asterisk/sounds/*digits/de/*6.gsm before 
falling back to /var/lib/asterisk/sounds/digits/6.gsm.


Markus

http://www.voip-info.org/wiki/view/Asterisk+multi-language

Am 14.06.2015 um 09:36 schrieb Luca Bertoncello:

Hi again

I'd like to configured my Asterisk to use german sounds for the
Say-commands...

I installed the sounds-files and I tried them with
Playback(de/demo-echodone) and it works.

Now I tried to add an extension to say the current time:

exten = 24,1,Verbose(2,Time asked by ${CALLERID(num)})
Exten = 24,n,Set(CHANNEL(language)=de)
Exten = 24,n,SayUnixTime()
Exten = 24,n,Hangup

But if I call the 24, it says the time in English...
On the CLI I see:

 -- Executing [24@default:2] Set(SIP/004935-0003, 
CHANNEL(language)=de) in new stack
 -- Executing [24@default:3] SayUnixTime(SIP/004935-0003, ) 
in new stack
 -- SIP/004935-0003 Playing 'digits/day-0.gsm' (language 'de')
 -- SIP/004935-0003 Playing 'digits/h-14.gsm' (language 'de')
...

So, it seems, it would use the German sounds, but it doesn't...
Has someone an explanation why it works so?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)



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Re: [asterisk-users] Strange and complete failure of Asterisk 1.8

2015-05-27 Thread Markus Weiler 

definitely DNS...
check your Register lines...


Markus



Am 27.05.2015 um 20:14 schrieb Duncan Turnbull:

DNS failure could do this

Asterisk used to get stuck in a symmetric DNS request wait state which meant 
everything ground to a halt as it waited for a reply while DNS timed out.

The recommended option was either ip only or a DNS proxy that failed fast this 
letting asterisk continue

Cheers Duncan




On 27/05/2015, at 11:55 pm, Stefan Viljoen viljo...@verishare.co.za wrote:

Hi all

We've had a very strange failure on an Asterisk 1.8 install that has been
running for about a year at a customer site.

The physical hardware is fine, all other services off the Centos 6.5 server
are running. Only Asterisk is not working...

The first symptom was that no calls can be made over the SIP phones used
with it, and no calls could be received over the SIP trunk connected to it.

I checked and noted that

sip show peers

in the CLI would either do nothing (e. g. just show asterisk*cli again,
with no response) or it would return only this:

asterisk*CLI sip show peers
Name/username  HostDyn
Forcerport ACL Port Status
asterisk*CLI

A module show like sip also does literally nothing, just

asterisk*cli module show like sip
asterisk*CLI

Soon after this, I lost the capacity to get any response if I do an asterisk
-r on the commandline - it would just hang indefintely.

Did a reboot, and then, I couldnt start asterisk at all - entering

# asterisk

would also just hang.

So, I recompiled asterisk from source and reinstalled the executable and all
the module files. Still the same.

I happened to have an older asterisk executable from a few months before
laying around and sha256summed it - and there was a difference in the
checksum vs. the non-working asterisk binary - BUT it turned out that the
newly recompiled asterisk binary has the SAME SHA256 checksum as the
non-working asterisk binary.

System seems fine otherwise, nothing relevant in /var/log/messages or dmesg
indicating a hardware failure. /var/log/asterisk/messages also contains no
strange warnings or errors.

Anybody got any idea why I cannot resuscitate my Asterisk install, even
after recompiling it from scratch from source? Why would asterisk die like
this in the first place?

Thanks


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Re: [asterisk-users] Call Quality Measuring

2015-03-25 Thread Markus Weiler 

Hi Patrick,

try voipmon, there it's free and you can even track MOS.

Markus


Am 25.03.2015 um 14:21 schrieb Patrick Beaumont:

Hi everyone.

We regularly get customers complaining about call quality issues. Most of
the time it turns out to be their own broadband. Very occasionally server
load. Does anyone have any advice or links to advice on measuring call
quality?

I’ve been playing around with “sip show channelstats” but can’t other than
measuring the packet loss I don’t really know what I’m supposed to be
looking for in order to say “ah ha! that’s the problem!”. I also don’t
know what it’s limits are. Will the stats in “sip show channelstats” show
a customer using a torrent client and saturating their own broadband
connection?

Regards,
Patrick.




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[asterisk-users] Asterisk API

2015-03-08 Thread Markus Weiler 

Hi all,

currently we're looking to program a new asterisk application. Years ago 
we used AMI and Asterisk Java.
When we did this we pretty soon encountered performance issues when 
using a lot of channels.
We want to place calls, bridge channels, disconnect channels, monitor 
them, hangup.

What's the status with Asterisk REST API?
Any experiences on performance,stability,documentation, caveats? Any 
toolkits for a fast start, Frameworks in any language? Hints? Best 
practices?


Thanks for any insights!

Markus


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Re: [asterisk-users] 603 Declined Dialstatus Busy

2015-02-27 Thread Markus Weiler 

Hi Nick,

maybe this will help?

exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code : 
${HASH(SIP_CAUSE,${CDR(dstchannel)})})


(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause)

Markus

Am 27.02.2015 um 18:56 schrieb Nick Olsen:

Hello Everyone.
In my outbound contexts, I'm using ${DIALSTATUS} to fail over to 
other routes if the chosen route rejects the call.
Now, My current scenario is if I get BUSY back from the first 
provider, I send a busy back to my customer. If I get something like 
CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt 
the call.
This works great as expected. However, One of my SIP carriers likes to 
send back 603 DECLINED inplace of 503's. Asterisk ${DIALSTATUS} 
treats this as Busy. Can I change how asterisk interprets a 603 
Declined? So it treats is as CHANUNAVAIL?
The obvious quick fix is to change my Busy option to attempt another 
carrier before finally returning BUSY to the customer. But I was 
hoping to not have to do that. Any ideas?

Nick Olsen
Network Operations
(855) FLSPEED  x106





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Re: [asterisk-users] Attack on Sip server.

2014-06-27 Thread Markus Weiler 

very simple,
yet effective

http://www.palner.com/blog/171/asterisk-no-matching-peer-found-block/



Am 27.06.2014 16:58, schrieb Steven Howes:
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com 
mailto:anuragrana31...@gmail.com wrote:
There are lot of requests coming in and I am not able to stop it 
because I am unable to detect the IP address.

I used wireshark to capture the packets.


If you can capture the packet, surely you have the IP? If they intend 
to get the response then the IP header can't be forged.


Steve




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Re: [asterisk-users] Moving User Agent To Remote Location

2013-01-03 Thread Markus Weiler 

Am 03.01.2013 21:21, schrieb Nick Khamis:

Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008

do you mean 1_000_8 ?

Markus


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Re: [asterisk-users] Top Posting

2013-01-02 Thread Markus Weiler 

Hi,
one more hint... (trying to translate the commands to english)
in Thunderbird open - Extras - Filter.. -
Filter-Name:  enter Top Posting
Subject - Contains: enter Top Posting
Action: Delete

Markus



Am 02.01.2013 21:31, schrieb Steve Totaro:

On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote:

On 1/2/2013 12:20 PM, Steve Totaro wrote:

On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us
wrote:

On 1/2/2013 11:30 AM, Richard Kenner wrote:

If things were properly trimmed, the email would be short enough that
it
really doesn't matter that much if the new material is on the top or
bottom, but people who top-post and don't trim create really
hard-to-follow
emails.

Not really true often times when people do the right thing and post
debug and conf files often required to get meaningful help.

Yes, but if you put those at the end, where they belong, people reading
the email can follow the thread quite easily and can ignore those if
they don't need them.  Certainly only a tiny part of such, if any at
all,
should be included in a reply.


Ok folks, could not stop myself any longer.   This pissing and moaning is
foolish to say the least.  There was a post a while ago in the original
hijacked thread by Steve Edwards that gave a link to the rules of the
list
at:
http://www.asterisk.org/community/discuss/

GO READ THEM!

Directly before the list of Rules is:

Show consideration. It's important to read the rules before posting on a
mailing list.

Sage advice if you ask me, and yes I know nobody actually asked me.

It is not hard to follow the rules .  If the nice folks at Digium took
the
time to post rules we should at least TRY to follow them. If you do not
like
the rules you can always petition Digium to change them but, taking up
bandwidth on the list in this all to frequent pissing match is a futile
waste of time.

Grow up, follow the rules, have a good day.
JohnM


I became a list member way before any such rule and never had to click
through and agree to these update ToS.

I am grandfathered in.

Thanks,
Steve Totaro

So Steve, can I steal this and send it to the IRS? The ATF? Local Police
Department? G  Wouldn't that be nice!  Sorry couldn't  resist.

JohnM


What the hell are you implying?  The local police love me, I am in
good standing with the ATF, FBI, DoD, DoS, USAID, DoE, DoL, and NSA.

IRS wants some money in April but don't they always? LOL.

Thanks,
Steve T

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Re: [asterisk-users] Intruder

2012-11-16 Thread Markus Weiler 

Hi Felix,

ngrep -W byline port 5060|grep -B1 INVITE sip

Markus


Am 16.11.2012 17:50, schrieb Ruben Rögels:

Hi Felix,

you have several things to check:

netstat -a -n --udp --tcp

will show you connections and connection attempts on network layer level.
You have to look for incoming connections to port 5060 and if the call 
has been established for connections on your rtp ports. (see rtp.conf).
If you can see connections not supposed to be there: thats your 
intruder ;-)


I suggest you disable guest calls and you configure a default context 
in which dialed extensions can't be routed to charged destinations.


sip.conf:
allowguests=no
defaultcontext=default

extensions.conf:
[default]
exten = _X.,1,Answer()
exten = _X.,n,PlayBack(silence/1)
exten = _X.,n,PlayBack(ss-noservice)
exten = _X.,n,PlayBack(silence/1)
exten = _X.,n,MusicOnHold(default,10)
exten = _X.,n,PlayBack(silence/1)
exten = _X.,n,PlayBack(vm-goodbye)
exten = _X.,n,HangUp()

The  next step would be using fail2ban or something similiar to check 
the asterisk log for intruders.

fail2ban recognized them and dynamically sets appropriate firewall rules.

Good luck.

best regards,
Ruben



Am 16.11.2012 17:20, schrieb Felix Vazquez:


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.


Felix




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Re: [asterisk-users] Restarting MOH

2012-11-13 Thread Markus Weiler 

hi,

try to catch in in a cron job per minute.

asterisk -rx 'module unload res_musiconhold.so'

Markus




Am 13.11.2012 19:15, schrieb Markus:

Am 13.11.2012 19:01, schrieb Eric Wieling:

module unload res_musiconhold.so
and
module load res_musiconhold.so


Great, that works, but only if no caller is listening to MOH at that 
time. Since *all* my callers are listening to MOH and nothing else, 
that means for me it's the same like an Asterisk restart.


When I try to unload the module I get:

loader.c:542 ast_unload_resource: Soft unload failed, 
'res_musiconhold.so' has use count 2


Is there a way to force the unloading?

Any other suggestions?

Thank you!
Markus


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Re: [asterisk-users] Web based Click to Call Application

2012-11-10 Thread Markus Weiler 

Hi,

I suppose WebRTC is the best solution nowadays, it's extremely interesting.

I developed a C2C app in 2008, starting with call files and AMI, ended 
with asterisk-java and asterisk.NET to solve it.
Hint: Try to solve (al)most (all) of your problems using 
Dialplans/Variables. Basically it's just one originate action using 
local channels.


Markus


Am 09.11.2012 11:38, schrieb Binan AL Halabi:

Hi,
Here is a starting point (WebRTC):
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

Regards.

// Binan.

*Från:* akhilesh chand omakhileshch...@gmail.com
*Till:* Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

*Skickat:* fredag, 9 november 2012 11:32
*Ämne:* [asterisk-users] Web based Click to Call Application

Dear All,

I want to develop click to call(C2C) web based application.Is there 
any study material.

I will really appreciate your help, thank you.



Regards
Akhilesh

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[asterisk-users] Musiconhold Problem

2010-07-21 Thread Markus Weiler 
Hi,
we are facing the problem , that we cannot distinguish between a trunk 
an an extension.
On our trunk side, if the remote user puts us on hold the same 
Musiconhold is played as if we would call another extension on the sam 
Asterisk PBX.

Asterisk should play the music from the remote End not its own

see also https://issues.asterisk.org/view.php?id=16901

I Guess the Problem applies mainly to Germany because it's an ISDN Message.


are there any solutions??


cheer Markus


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Re: [asterisk-users] Random crashes on Bridgeaction

2010-01-03 Thread Markus Weiler 

Hi,

I still don't know if it's a bug or if it's already fixed esp. what 
exactly is the source...how could i find this out?

or where could i open the bug report?
thanks
Markus

Am 02.01.2010 19:16, schrieb Steve Totaro:

Did you open a bug report?

On Sat, Jan 2, 2010 at 12:37 PM, Markus Weiler 
markus_wei...@mailworks.org mailto:markus_wei...@mailworks.org wrote:


Could anybody give me a hint how to investigate that problem?
cheers Markus

Am 31.12.2009 18:17, schrieb Markus Weiler:
 Sorry wrong topic...

 Hi,

 I'm issuing a Bridgeaction through the manager interface.
 One Person is called, when answered second one is called first
gets MoH. After the second person
 answers both channels are bridged together.
 Randomly (approx. 1/5.000 calls (sometimes twice a day,
sometimes once a week)) asterisk crashes.
 I suspected res_musiconhold and updated to the latest Version
(repository) but nothing changed.
 here are some backtraces( number 'd).
 I can offer various core dumps, dialplan etc:
 Help would be greatly appreciated as I don't get any further on
this problem and I have no Idea what to do.

 Verision: Asterisk 1.6.0.19


 03.dec

 #0  0xb7d24572 in free () from /lib/tls/i686/cmov/libc.so.6
 #1  0xb7d20ac4 in _IO_free_backup_area () from
/lib/tls/i686/cmov/libc.so.6
 #2  0xb7d213c9 in __underflow () from /lib/tls/i686/cmov/libc.so.6
 #3  0xb7d1d868 in ?? () from /lib/tls/i686/cmov/libc.so.6
 #4  0xb7d1fcf8 in _IO_sgetn () from /lib/tls/i686/cmov/libc.so.6
 #5  0xb7d12fe0 in fread () from /lib/tls/i686/cmov/libc.so.6
 #6  0xb5e6fced in wav_read (s=0x87c7c08, whennext=0xb54fca14) at
 format_wav.c:363
 #7  0x080dcebb in read_frame (s=0x87c7c08, whennext=0xb54fca14) at
 file.c:697
 #8  0x080dcf48 in ast_readframe (s=0x87c7c08) at file.c:718
 #9  0xb760088a in spawn_mp3 (class=0xb54f7a34) at
res_musiconhold.c:501
 #10 0xb7600909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
 #11 0x0809abc0 in ast_read_generator_actions (chan=0xb63b7a90,
 f=0xb63b6c40) at channel.c:2514
 #12 0x0809c9e3 in __ast_read (chan=0xb63b7a90, dropaudio=0) at
 channel.c:3001
 #13 0x0809cd50 in ast_read (chan=0xb63b7a90) at channel.c:3037
 #14 0x08097743 in ast_safe_sleep_conditional (chan=0xb63b7a90,
ms=9967,
 cond=0, data=0x0) at channel.c:1297
 #15 0x080977a2 in ast_safe_sleep (chan=0xb63b7a90, ms=1) at
 channel.c:1309
 #16 0xb760214c in moh_alloc (chan=0xb63b7a90, params=0xb54ff0b8) at
 res_musiconhold.c:905
 #17 0x08107446 in pbx_exec (c=0xb63b7a90, app=0xb7a09960,
 data=0xb54ff0b8) at pbx.c:951
 #18 0x0810ee3f in pbx_extension_helper (c=0xb63b7a90, con=0x0,
 context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142,
priority=3,
 label=0x0,
callerid=0xb63e7b70 49711XXX, action=E_SPAWN,
 found=0xb5501208, combined_find_spawn=1) at pbx.c:3138
 #19 0x08110b76 in ast_spawn_extension (c=0xb63b7a90,
context=0xb63b7cd8
 Click2Call4_0, exten=0xb63b7d28 142, priority=3,
callerid=0xb63e7b70 49711XXX, found=0xb5501208,
 combined_find_spawn=1) at pbx.c:3605
 #20 0x081112ed in __ast_pbx_run (c=0xb63b7a90, args=0x0) at
pbx.c:3692
 #21 0x08112714 in pbx_thread (data=0xb63b7a90) at pbx.c:3965
 #22 0x0816a6ed in dummy_start (data=0xb63e7dd0) at utils.c:861
 #23 0xb7c9c4ff in start_thread () from
/lib/tls/i686/cmov/libpthread.so.0
 #24 0xb7d9749e in clone () from /lib/tls/i686/cmov/libc.so.6


 08. Dec

 #0  0xb8000424 in __kernel_vsyscall ()
 #1  0xb7d2d6d0 in raise () from /lib/tls/i686/cmov/libc.so.6
 #2  0xb7d2f098 in abort () from /lib/tls/i686/cmov/libc.so.6
 #3  0xb7d6b24d in ?? () from /lib/tls/i686/cmov/libc.so.6
 #4  0xb7d71604 in ?? () from /lib/tls/i686/cmov/libc.so.6
 #5  0xb7d6d57d in _IO_file_seekoff () from
/lib/tls/i686/cmov/libc.so.6
 #6  0xb7d63760 in ?? () from /lib/tls/i686/cmov/libc.so.6
 #7  0xb7d6ad37 in ftello64 () from /lib/tls/i686/cmov/libc.so.6
 #8  0xb715cc19 in wav_read (s=0x8f08d78, whennext=0xb6adba14) at
 format_wav.c:352
 #9  0x080dcebb in read_frame (s=0x8f08d78, whennext=0xb6adba14) at
 file.c:697
 #10 0x080dcf48 in ast_readframe (s=0x8f08d78) at file.c:718
 #11 0xb76508b3 in spawn_mp3 (class=0xb6ad6a34) at
res_musiconhold.c:504
 #12 0xb7650909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
 #13 0x0809abc0 in ast_read_generator_actions (chan=0xb650a2d8,
 f=0xb650b810) at channel.c:2514
 #14 0x0809c9e3 in __ast_read (chan=0xb650a2d8, dropaudio=0) at
 channel.c:3001
 #15 0x0809cd50 in ast_read (chan=0xb650a2d8) at channel.c:3037
 #16 0x08097743 in ast_safe_sleep_conditional (chan=0xb650a2d8,
ms=1887,
 cond=0, data=0x0) at channel.c:1297
 #17 0x080977a2 in ast_safe_sleep

Re: [asterisk-users] Random crashes on Bridgeaction

2010-01-02 Thread Markus Weiler 
Could anybody give me a hint how to investigate that problem?
cheers Markus

Am 31.12.2009 18:17, schrieb Markus Weiler:
 Sorry wrong topic...

 Hi,

 I'm issuing a Bridgeaction through the manager interface.
 One Person is called, when answered second one is called first gets MoH. 
 After the second person
 answers both channels are bridged together.
 Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a 
 week)) asterisk crashes.
 I suspected res_musiconhold and updated to the latest Version (repository) 
 but nothing changed.
 here are some backtraces( number 'd).
 I can offer various core dumps, dialplan etc:
 Help would be greatly appreciated as I don't get any further on this problem 
 and I have no Idea what to do.

 Verision: Asterisk 1.6.0.19


 03.dec

 #0  0xb7d24572 in free () from /lib/tls/i686/cmov/libc.so.6
 #1  0xb7d20ac4 in _IO_free_backup_area () from /lib/tls/i686/cmov/libc.so.6
 #2  0xb7d213c9 in __underflow () from /lib/tls/i686/cmov/libc.so.6
 #3  0xb7d1d868 in ?? () from /lib/tls/i686/cmov/libc.so.6
 #4  0xb7d1fcf8 in _IO_sgetn () from /lib/tls/i686/cmov/libc.so.6
 #5  0xb7d12fe0 in fread () from /lib/tls/i686/cmov/libc.so.6
 #6  0xb5e6fced in wav_read (s=0x87c7c08, whennext=0xb54fca14) at
 format_wav.c:363
 #7  0x080dcebb in read_frame (s=0x87c7c08, whennext=0xb54fca14) at
 file.c:697
 #8  0x080dcf48 in ast_readframe (s=0x87c7c08) at file.c:718
 #9  0xb760088a in spawn_mp3 (class=0xb54f7a34) at res_musiconhold.c:501
 #10 0xb7600909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
 #11 0x0809abc0 in ast_read_generator_actions (chan=0xb63b7a90,
 f=0xb63b6c40) at channel.c:2514
 #12 0x0809c9e3 in __ast_read (chan=0xb63b7a90, dropaudio=0) at
 channel.c:3001
 #13 0x0809cd50 in ast_read (chan=0xb63b7a90) at channel.c:3037
 #14 0x08097743 in ast_safe_sleep_conditional (chan=0xb63b7a90, ms=9967,
 cond=0, data=0x0) at channel.c:1297
 #15 0x080977a2 in ast_safe_sleep (chan=0xb63b7a90, ms=1) at
 channel.c:1309
 #16 0xb760214c in moh_alloc (chan=0xb63b7a90, params=0xb54ff0b8) at
 res_musiconhold.c:905
 #17 0x08107446 in pbx_exec (c=0xb63b7a90, app=0xb7a09960,
 data=0xb54ff0b8) at pbx.c:951
 #18 0x0810ee3f in pbx_extension_helper (c=0xb63b7a90, con=0x0,
 context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3,
 label=0x0,
callerid=0xb63e7b70 49711XXX, action=E_SPAWN,
 found=0xb5501208, combined_find_spawn=1) at pbx.c:3138
 #19 0x08110b76 in ast_spawn_extension (c=0xb63b7a90, context=0xb63b7cd8
 Click2Call4_0, exten=0xb63b7d28 142, priority=3,
callerid=0xb63e7b70 49711XXX, found=0xb5501208,
 combined_find_spawn=1) at pbx.c:3605
 #20 0x081112ed in __ast_pbx_run (c=0xb63b7a90, args=0x0) at pbx.c:3692
 #21 0x08112714 in pbx_thread (data=0xb63b7a90) at pbx.c:3965
 #22 0x0816a6ed in dummy_start (data=0xb63e7dd0) at utils.c:861
 #23 0xb7c9c4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
 #24 0xb7d9749e in clone () from /lib/tls/i686/cmov/libc.so.6


 08. Dec

 #0  0xb8000424 in __kernel_vsyscall ()
 #1  0xb7d2d6d0 in raise () from /lib/tls/i686/cmov/libc.so.6
 #2  0xb7d2f098 in abort () from /lib/tls/i686/cmov/libc.so.6
 #3  0xb7d6b24d in ?? () from /lib/tls/i686/cmov/libc.so.6
 #4  0xb7d71604 in ?? () from /lib/tls/i686/cmov/libc.so.6
 #5  0xb7d6d57d in _IO_file_seekoff () from /lib/tls/i686/cmov/libc.so.6
 #6  0xb7d63760 in ?? () from /lib/tls/i686/cmov/libc.so.6
 #7  0xb7d6ad37 in ftello64 () from /lib/tls/i686/cmov/libc.so.6
 #8  0xb715cc19 in wav_read (s=0x8f08d78, whennext=0xb6adba14) at
 format_wav.c:352
 #9  0x080dcebb in read_frame (s=0x8f08d78, whennext=0xb6adba14) at
 file.c:697
 #10 0x080dcf48 in ast_readframe (s=0x8f08d78) at file.c:718
 #11 0xb76508b3 in spawn_mp3 (class=0xb6ad6a34) at res_musiconhold.c:504
 #12 0xb7650909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
 #13 0x0809abc0 in ast_read_generator_actions (chan=0xb650a2d8,
 f=0xb650b810) at channel.c:2514
 #14 0x0809c9e3 in __ast_read (chan=0xb650a2d8, dropaudio=0) at
 channel.c:3001
 #15 0x0809cd50 in ast_read (chan=0xb650a2d8) at channel.c:3037
 #16 0x08097743 in ast_safe_sleep_conditional (chan=0xb650a2d8, ms=1887,
 cond=0, data=0x0) at channel.c:1297
 #17 0x080977a2 in ast_safe_sleep (chan=0xb650a2d8, ms=1) at
 channel.c:1309
 #18 0xb765214c in moh_alloc (chan=0xb650a2d8, params=0xb6ade0b8) at
 res_musiconhold.c:905
 #19 0x08107446 in pbx_exec (c=0xb650a2d8, app=0xb7a16d28,
 data=0xb6ade0b8) at pbx.c:951
 #20 0x0810ee3f in pbx_extension_helper (c=0xb650a2d8, con=0x0,
 context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3,
 label=0x0,
callerid=0xb6509a58 49711XXX, action=E_SPAWN,
 found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3138
 #21 0x08110b76 in ast_spawn_extension (c=0xb650a2d8, context=0xb650a520
 Click2Call4_0, exten=0xb650a570 142, priority=3,
callerid=0xb6509a58 49711XXX, found=0xb6ae0208,
 combined_find_spawn=1) at pbx.c:3605
 #22 0x081112ed in __ast_pbx_run (c=0xb650a2d8, args=0x0) at pbx.c

Re: [asterisk-users] identifying channel for softhangup

2009-12-31 Thread Markus Weiler 

Hi,

I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH. After 
the second person
answers both channels are bridged together.
Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a week)) 
asterisk crashes.
I suspected res_musiconhold and updated to the latest Version (repository) but 
nothing changed.
here are some backtraces( number 'd).
I can offer various core dumps, dialplan etc:
Help would be greatly appreciated as I don't get any further on this problem 
and I have no Idea what to do.

Verision: Asterisk 1.6.0.19


03.dec

#0  0xb7d24572 in free () from /lib/tls/i686/cmov/libc.so.6
#1  0xb7d20ac4 in _IO_free_backup_area () from /lib/tls/i686/cmov/libc.so.6
#2  0xb7d213c9 in __underflow () from /lib/tls/i686/cmov/libc.so.6
#3  0xb7d1d868 in ?? () from /lib/tls/i686/cmov/libc.so.6
#4  0xb7d1fcf8 in _IO_sgetn () from /lib/tls/i686/cmov/libc.so.6
#5  0xb7d12fe0 in fread () from /lib/tls/i686/cmov/libc.so.6
#6  0xb5e6fced in wav_read (s=0x87c7c08, whennext=0xb54fca14) at
format_wav.c:363
#7  0x080dcebb in read_frame (s=0x87c7c08, whennext=0xb54fca14) at
file.c:697
#8  0x080dcf48 in ast_readframe (s=0x87c7c08) at file.c:718
#9  0xb760088a in spawn_mp3 (class=0xb54f7a34) at res_musiconhold.c:501
#10 0xb7600909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
#11 0x0809abc0 in ast_read_generator_actions (chan=0xb63b7a90,
f=0xb63b6c40) at channel.c:2514
#12 0x0809c9e3 in __ast_read (chan=0xb63b7a90, dropaudio=0) at
channel.c:3001
#13 0x0809cd50 in ast_read (chan=0xb63b7a90) at channel.c:3037
#14 0x08097743 in ast_safe_sleep_conditional (chan=0xb63b7a90, ms=9967,
cond=0, data=0x0) at channel.c:1297
#15 0x080977a2 in ast_safe_sleep (chan=0xb63b7a90, ms=1) at
channel.c:1309
#16 0xb760214c in moh_alloc (chan=0xb63b7a90, params=0xb54ff0b8) at
res_musiconhold.c:905
#17 0x08107446 in pbx_exec (c=0xb63b7a90, app=0xb7a09960,
data=0xb54ff0b8) at pbx.c:951
#18 0x0810ee3f in pbx_extension_helper (c=0xb63b7a90, con=0x0,
context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3,
label=0x0,
 callerid=0xb63e7b70 49711XXX, action=E_SPAWN,
found=0xb5501208, combined_find_spawn=1) at pbx.c:3138
#19 0x08110b76 in ast_spawn_extension (c=0xb63b7a90, context=0xb63b7cd8
Click2Call4_0, exten=0xb63b7d28 142, priority=3,
 callerid=0xb63e7b70 49711XXX, found=0xb5501208,
combined_find_spawn=1) at pbx.c:3605
#20 0x081112ed in __ast_pbx_run (c=0xb63b7a90, args=0x0) at pbx.c:3692
#21 0x08112714 in pbx_thread (data=0xb63b7a90) at pbx.c:3965
#22 0x0816a6ed in dummy_start (data=0xb63e7dd0) at utils.c:861
#23 0xb7c9c4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
#24 0xb7d9749e in clone () from /lib/tls/i686/cmov/libc.so.6


08. Dec

#0  0xb8000424 in __kernel_vsyscall ()
#1  0xb7d2d6d0 in raise () from /lib/tls/i686/cmov/libc.so.6
#2  0xb7d2f098 in abort () from /lib/tls/i686/cmov/libc.so.6
#3  0xb7d6b24d in ?? () from /lib/tls/i686/cmov/libc.so.6
#4  0xb7d71604 in ?? () from /lib/tls/i686/cmov/libc.so.6
#5  0xb7d6d57d in _IO_file_seekoff () from /lib/tls/i686/cmov/libc.so.6
#6  0xb7d63760 in ?? () from /lib/tls/i686/cmov/libc.so.6
#7  0xb7d6ad37 in ftello64 () from /lib/tls/i686/cmov/libc.so.6
#8  0xb715cc19 in wav_read (s=0x8f08d78, whennext=0xb6adba14) at
format_wav.c:352
#9  0x080dcebb in read_frame (s=0x8f08d78, whennext=0xb6adba14) at
file.c:697
#10 0x080dcf48 in ast_readframe (s=0x8f08d78) at file.c:718
#11 0xb76508b3 in spawn_mp3 (class=0xb6ad6a34) at res_musiconhold.c:504
#12 0xb7650909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
#13 0x0809abc0 in ast_read_generator_actions (chan=0xb650a2d8,
f=0xb650b810) at channel.c:2514
#14 0x0809c9e3 in __ast_read (chan=0xb650a2d8, dropaudio=0) at
channel.c:3001
#15 0x0809cd50 in ast_read (chan=0xb650a2d8) at channel.c:3037
#16 0x08097743 in ast_safe_sleep_conditional (chan=0xb650a2d8, ms=1887,
cond=0, data=0x0) at channel.c:1297
#17 0x080977a2 in ast_safe_sleep (chan=0xb650a2d8, ms=1) at
channel.c:1309
#18 0xb765214c in moh_alloc (chan=0xb650a2d8, params=0xb6ade0b8) at
res_musiconhold.c:905
#19 0x08107446 in pbx_exec (c=0xb650a2d8, app=0xb7a16d28,
data=0xb6ade0b8) at pbx.c:951
#20 0x0810ee3f in pbx_extension_helper (c=0xb650a2d8, con=0x0,
context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3,
label=0x0,
 callerid=0xb6509a58 49711XXX, action=E_SPAWN,
found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3138
#21 0x08110b76 in ast_spawn_extension (c=0xb650a2d8, context=0xb650a520
Click2Call4_0, exten=0xb650a570 142, priority=3,
 callerid=0xb6509a58 49711XXX, found=0xb6ae0208,
combined_find_spawn=1) at pbx.c:3605
#22 0x081112ed in __ast_pbx_run (c=0xb650a2d8, args=0x0) at pbx.c:3692
#23 0x08112714 in pbx_thread (data=0xb650a2d8) at pbx.c:3965
#24 0x0816a6ed in dummy_start (data=0xb6502e98) at utils.c:861
#25 0xb7ceb4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
#26 0xb7de649e in clone () from

[asterisk-users] Random crashes on Bridgeaction

2009-12-31 Thread Markus Weiler 
Sorry wrong topic...

Hi,

I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH. After 
the second person
answers both channels are bridged together.
Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a week)) 
asterisk crashes.
I suspected res_musiconhold and updated to the latest Version (repository) but 
nothing changed.
here are some backtraces( number 'd).
I can offer various core dumps, dialplan etc:
Help would be greatly appreciated as I don't get any further on this problem 
and I have no Idea what to do.

Verision: Asterisk 1.6.0.19


03.dec

#0  0xb7d24572 in free () from /lib/tls/i686/cmov/libc.so.6
#1  0xb7d20ac4 in _IO_free_backup_area () from /lib/tls/i686/cmov/libc.so.6
#2  0xb7d213c9 in __underflow () from /lib/tls/i686/cmov/libc.so.6
#3  0xb7d1d868 in ?? () from /lib/tls/i686/cmov/libc.so.6
#4  0xb7d1fcf8 in _IO_sgetn () from /lib/tls/i686/cmov/libc.so.6
#5  0xb7d12fe0 in fread () from /lib/tls/i686/cmov/libc.so.6
#6  0xb5e6fced in wav_read (s=0x87c7c08, whennext=0xb54fca14) at
format_wav.c:363
#7  0x080dcebb in read_frame (s=0x87c7c08, whennext=0xb54fca14) at
file.c:697
#8  0x080dcf48 in ast_readframe (s=0x87c7c08) at file.c:718
#9  0xb760088a in spawn_mp3 (class=0xb54f7a34) at res_musiconhold.c:501
#10 0xb7600909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
#11 0x0809abc0 in ast_read_generator_actions (chan=0xb63b7a90,
f=0xb63b6c40) at channel.c:2514
#12 0x0809c9e3 in __ast_read (chan=0xb63b7a90, dropaudio=0) at
channel.c:3001
#13 0x0809cd50 in ast_read (chan=0xb63b7a90) at channel.c:3037
#14 0x08097743 in ast_safe_sleep_conditional (chan=0xb63b7a90, ms=9967,
cond=0, data=0x0) at channel.c:1297
#15 0x080977a2 in ast_safe_sleep (chan=0xb63b7a90, ms=1) at
channel.c:1309
#16 0xb760214c in moh_alloc (chan=0xb63b7a90, params=0xb54ff0b8) at
res_musiconhold.c:905
#17 0x08107446 in pbx_exec (c=0xb63b7a90, app=0xb7a09960,
data=0xb54ff0b8) at pbx.c:951
#18 0x0810ee3f in pbx_extension_helper (c=0xb63b7a90, con=0x0,
context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3,
label=0x0,
  callerid=0xb63e7b70 49711XXX, action=E_SPAWN,
found=0xb5501208, combined_find_spawn=1) at pbx.c:3138
#19 0x08110b76 in ast_spawn_extension (c=0xb63b7a90, context=0xb63b7cd8
Click2Call4_0, exten=0xb63b7d28 142, priority=3,
  callerid=0xb63e7b70 49711XXX, found=0xb5501208,
combined_find_spawn=1) at pbx.c:3605
#20 0x081112ed in __ast_pbx_run (c=0xb63b7a90, args=0x0) at pbx.c:3692
#21 0x08112714 in pbx_thread (data=0xb63b7a90) at pbx.c:3965
#22 0x0816a6ed in dummy_start (data=0xb63e7dd0) at utils.c:861
#23 0xb7c9c4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
#24 0xb7d9749e in clone () from /lib/tls/i686/cmov/libc.so.6


08. Dec

#0  0xb8000424 in __kernel_vsyscall ()
#1  0xb7d2d6d0 in raise () from /lib/tls/i686/cmov/libc.so.6
#2  0xb7d2f098 in abort () from /lib/tls/i686/cmov/libc.so.6
#3  0xb7d6b24d in ?? () from /lib/tls/i686/cmov/libc.so.6
#4  0xb7d71604 in ?? () from /lib/tls/i686/cmov/libc.so.6
#5  0xb7d6d57d in _IO_file_seekoff () from /lib/tls/i686/cmov/libc.so.6
#6  0xb7d63760 in ?? () from /lib/tls/i686/cmov/libc.so.6
#7  0xb7d6ad37 in ftello64 () from /lib/tls/i686/cmov/libc.so.6
#8  0xb715cc19 in wav_read (s=0x8f08d78, whennext=0xb6adba14) at
format_wav.c:352
#9  0x080dcebb in read_frame (s=0x8f08d78, whennext=0xb6adba14) at
file.c:697
#10 0x080dcf48 in ast_readframe (s=0x8f08d78) at file.c:718
#11 0xb76508b3 in spawn_mp3 (class=0xb6ad6a34) at res_musiconhold.c:504
#12 0xb7650909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511
#13 0x0809abc0 in ast_read_generator_actions (chan=0xb650a2d8,
f=0xb650b810) at channel.c:2514
#14 0x0809c9e3 in __ast_read (chan=0xb650a2d8, dropaudio=0) at
channel.c:3001
#15 0x0809cd50 in ast_read (chan=0xb650a2d8) at channel.c:3037
#16 0x08097743 in ast_safe_sleep_conditional (chan=0xb650a2d8, ms=1887,
cond=0, data=0x0) at channel.c:1297
#17 0x080977a2 in ast_safe_sleep (chan=0xb650a2d8, ms=1) at
channel.c:1309
#18 0xb765214c in moh_alloc (chan=0xb650a2d8, params=0xb6ade0b8) at
res_musiconhold.c:905
#19 0x08107446 in pbx_exec (c=0xb650a2d8, app=0xb7a16d28,
data=0xb6ade0b8) at pbx.c:951
#20 0x0810ee3f in pbx_extension_helper (c=0xb650a2d8, con=0x0,
context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3,
label=0x0,
  callerid=0xb6509a58 49711XXX, action=E_SPAWN,
found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3138
#21 0x08110b76 in ast_spawn_extension (c=0xb650a2d8, context=0xb650a520
Click2Call4_0, exten=0xb650a570 142, priority=3,
  callerid=0xb6509a58 49711XXX, found=0xb6ae0208,
combined_find_spawn=1) at pbx.c:3605
#22 0x081112ed in __ast_pbx_run (c=0xb650a2d8, args=0x0) at pbx.c:3692
#23 0x08112714 in pbx_thread (data=0xb650a2d8) at pbx.c:3965
#24 0x0816a6ed in dummy_start (data=0xb6502e98) at utils.c:861
#25 0xb7ceb4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
#26 0xb7de649e in

[asterisk-users] Dial option limit call duration

2009-06-10 Thread Markus Weiler 
Hi,

we're using the limit option like this:
Dial L(6:3)


[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] -- 
Limit Data for this call:
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
timelimit  = 6
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
play_warning   = 3
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
play_to_caller = yes
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
play_to_callee = yes
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
warning_freq   = 0
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
start_sound=
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
warning_sound  = timeleft
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] 
end_sound  =
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41]   == Using 
SIP RTP CoS mark 5

but the warning is played to late it used to be exactly
you have 30 seconds left but now its

[Jun 10 16:15:20] VERBOSE[12201] logger.c: [Jun 10 16:15:20] -- 
SIP/OB-0847a3c8 Playing 'vm-youhave.gsm' (language 'de')
[Jun 10 16:15:21] WARNING[12201] file.c: File digits/8-and does not 
exist in any format
[Jun 10 16:15:21] WARNING[12201] file.c: Unable to open digits/8-and 
(format 0x4 (ulaw)): No such file or directory
[Jun 10 16:15:21] VERBOSE[12201] logger.c: [Jun 10 16:15:21] -- 
SIP/OB-0847a3c8 Playing 'digits/20.gsm' (language 'de')
[Jun 10 16:15:23] VERBOSE[12201] logger.c: [Jun 10 16:15:23] -- 
SIP/OB-0847a3c8 Playing 'queue-seconds.gsm' (language 'de')

please ignore the File digits/8-and does not exist in any format. Taht 
is not the problem.

Why is the message two seconds too late. 28 sec instead of 30 secs??

thanks

Markus




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Re: [asterisk-users] Can YOU find a trailing parenthesis?

2009-05-19 Thread Markus Weiler 




Hi,

In VI:

In 'vi', moving the cursor over any bracket,
brace, etc, and then pressing '%' moves the cursor to the 'matching'
bracket/brace character. 

That can be very useful when programming, to find missing/extra
brackets and braces. It even seems to find matching pairs of #ifdef /
#endif pragmas in C/C++ programs.

in /etc/vimrc

set showmatch


although not really an asterisk-java issue :-)

Markus



sean darcy wrote:

  Philipp Kempgen wrote:
  
  
sean darcy schrieb:


  On 1.6.1, I must be losing my eyesight:
  


  exten = _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" )
  

exten = _6000XXXNXXX,n,Set(Time_in_secs=${STRFTIME(${EPOCH},,%s)})
^


  CLI:
 -- Starting simple switch on 'DAHDI/1-1'
[2009-05-17 14:54:49] WARNING[13433]: pbx.c:2846 func_args: Can't find 
trailing parenthesis?
 -- Executing [60001234...@internal:1] Set("DAHDI/1-1", 
"Time-in-secs= "1242586489" ") in new stack
  

Philipp Kempgen

  
  
Doh.

Any time you get to New York, I'll buy you a beer.

Thanks.

sean


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[asterisk-users] Free Fax for asterisk

2009-05-13 Thread Markus Weiler 
Hi,

I installed Digiums Free Fax for Asterisk and found out, that it 
automatically retries failed faxes, is there a way to stop that?

Thanks

Markus


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Re: [asterisk-users] Free Fax for asterisk

2009-05-13 Thread Markus Weiler 
Hi,
it was was my fault, there is no retry ... sorry to bother you.

@David:
I wasn`t very conviced about Spandsp, after trying several versions it 
worked, but not well.
We are sending faxes via SIP. When sending faxes from our 1.6 Asterisk 
to our 1.4 Asterisk 50%+ Faxes failed.
T.38 worked once then stopped I never found the right 
Asterisk-App_fax-Spandsp-dialplan setup again.
I event patched the app_fax, which contained errors, just to get it working.
Although I have to admit I installed it on our other Asterisk 
1.4.17-BRIstuffed (Spandsp 0.0.4-test6) and it works just fine.

I just tried Digiums solution to test if it´s better and it is, all the 
testfaxes went through.
T.38 worked instantly.
Configuration was pretty easy and well documented.
I think $50 per channel is not too much money either, just the 
support...well there is none.

hope i could help

Markus



David Klaverstyn wrote:
 Hi All,

 Sorry to hijack this post but I am confused.  What is the advantage of using 
 this Digium Fax For Asterisk product when you can use Asterisks' 1.6.x 
 module app_fax or Asterisks' 1.4.x agx-ast-addons with the app_txfax and 
 app_rxfax modules?

 Regards
 David.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler
 Sent: Wednesday, 13 May 2009 5:44 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Free Fax for asterisk

 Hi,

 I installed Digiums Free Fax for Asterisk and found out, that it 
 automatically retries failed faxes, is there a way to stop that?

 Thanks

 Markus


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