[asterisk-users] Problem with pjsip

[Yves]

Hello everyone.
I allow myself to submit a problem that I can not solve with my VOIP 
provider Orange in France


[2023-06-08 13:19:03] ERROR[185091]: 
res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error 
configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid 
character '@'
[2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 
aco_process_var: Error parsing from_user=75b55btqu...@orange-obs.fr at 
line 0 of

  == chan_pjsip.so => (PJSIP Channel Driver)

1) Error with "@" character which constitutes URI and authuser see 
excerpt from pjsip.conf.


[transport-udp]
type = transport
protocol=udp
bind=0.0.0.0:5060
local_net=172.16.1.0/255.255.255.0

[reg_orange-obs.fr]
type = registration
retry_interval = 120
max_retries = 10
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_orange-obs.fr
client_uri = sip:+3313445x...@orange-obs.fr
server_uri = sip:orange-obs.fr

[auth_reg_orange-obs.fr]
type=auth
password=3314C9BA9688C2AA
username = 75b55btqu...@orange-obs.fr

[Biv_Sortie]
type = aor
contact = sip:75b55btqu...@orange-obs.fr@orange-obs.fr
default_expiration = 3600

[Biv_Sortie]
type = identify
endpoint = Biv_Sortie
match = orange-obs.fr

[Biv_Sortie]
type=auth
username = Biv_Sortie
password=3314C9BA9688C2AA

[Biv_Sortie]
type=endpoint
context = Isdn_Inbound
dtmf_mode=rfc4733
disallow=all
allow = g722, alaw, g729
direct_media=no
trust_id_inbound = yes
send_rpid=yes
from_user = 75b55btqu...@orange-obs.fr
from_domain = orange-obs.fr
language = en
allow_subscribe = yes
auth = Biv_Exit
outbound_auth = Biv_Sortie
aors = Biv_Sortie

Question how can I solve this character problem "@"?

2) resolution of the orange-obs.fr DNS.  I am attaching an extract from 
the documentation that Orange issued in 2015


SIP/Internet is described in RFC3261 and following. THE
SIP/IMS is described by 3GPP standards. It's not the same
SIP.
In the Internet world, VoIP machines route
SIP messages to the IP addresses of the FQDNs of the SIP URIs
(VoIP domain). In the 3GPP world, SIP messages are
routed to an I/P-CSCF (depending on whether we are in interco or in
IPBX) which has a different FQDN from the VoIP domain.

BIV SIP

– P-CSCF FQDN: pcscfgm.orange-obs.fr, resolved by DNS
voice
– VoIP domain: orange-obs.fr, not resolved by voice DNS. ex :
INVITE sip:0142277...@orange-obs.fr SIP/2.0
2
 The VoIP/Internet machine will not be able to determine the address
recipient of SIP messages.

run the command “nslookup pcscfgm.orange-obs.fr” and
note the returned IP address 217.167.210.X
– add this address in the /etc/hosts file of the PBX:
217.167.210.X pcscfgm.orange-obs.fr orange-obs.fr

Note that it works with sip.conf . The current installation is 
operational with the information provided by /etc/hosts


below the debug in asterisk 19.6

[2023-06-08 13:37:17] DEBUG[185433]: res_config_odbc.c:115 
custom_prepare: Skip: 0; SQL: SELECT * FROM ps_auths WHERE id = ?
[2023-06-08 13:37:17] DEBUG[185433]: res_config_odbc.c:134 
custom_prepare: Parameter 1 ('id') = 'auth_reg_orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_odbc.c:808 
ast_odbc_release_obj: Releasing ODBC handle 0x55855d1977d0 into pool
[2023-06-08 13:37:17] DEBUG[185433]: config.c:3847 ast_parse_arg: 
extract uint from [32] in [0, 4294967295] gives [32](0)
[2023-06-08 13:37:17] DEBUG[185433]: 
res_pjsip_outbound_registration.c:699 handle_client_registration: 
Outbound REGISTER attempt 2 to 'sip:orange-obs.fr' with client 
'sip:+3313445x...@orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:475 
sip_resolve: Performing SIP DNS resolution of target 'orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:502 
sip_resolve: Transport type for target 'orange-obs.fr' is 'UDP transport'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:545 
sip_resolve: [0x55855d769c88] Created resolution tracking for target 
'orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174 
sip_resolve_add: [0x55855d769c88] Added target 'orange-obs.fr' with 
record type '35', transport 'UDP transport', and port '5060'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174 
sip_resolve_add: [0x55855d769c88] Added target '_sip._udp.orange-obs.fr' 
with record type '33', transport 'UDP transport', and port '5060'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174 
sip_resolve_add: [0x55855d769c88] Added target 'orange-obs.fr' with 
record type '1', transport 'UDP transport', and port '5060'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:616 
sip_resolve: [0x55855d769c88] Starting initial resolution using parallel 
queries for target 'orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185340]: dns.c:555 ast_search_dns_ex: DNS 
search failed for orange-obs.fr
[2023-06-08 13:37:17] DEBUG[185340]: dns_system_resolver.c:154 
dns_system_resolver_process_query: DNS search failed for query: 
'orange-obs.fr'
[2023-06-08 13:37:17] 

Re: [asterisk-users] Problem with AudioCodes MP-114 ATA

[Yves]

hi,

quite unlikely (besides of an defect) that the behaviour of your 
AudioCodes or Asterisk changed "from alone"... something must have changed.
What does the logs say (from asterisk... do you see register-events? and 
from you AudioCodes?)
The AudioCodes Devices can export and restore their config... do you 
have a backup?


regards,
yves

Am 10.01.2019 um 19:51 schrieb Tech Support:


All;

    I have an AudioCodes MP-114 four FXS ATA that recently stopped 
registering to my PBX. I’m pulling my hair out here trying to figure 
out the root cause without much success. Does anyone have a sample 
config file that I could use as a sample? Any insight at all would be 
greatly appreciated.


Thanks Much;

John




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Re: [asterisk-users] Use AGi Commands without script in Dialplan

[Yves]

Am 08.10.2018 um 13:02 schrieb Antony Stone:

On Monday 08 October 2018 at 12:44:43, Yves wrote:


I am looking for an easy way to execute any AGI Command directly from the
dialplan without the need to call an external script.

The whole point of AGI is that it calls an external script in order to replace
commands in the dialplan.

Executing an AGI command without an external script doesn't make sense.


Antony.


Hi Antony,

thanks for your answer, even if it is a bit disappointing for me. I 
understand the point... but...

why aren´t then all AGI-Commands also available as Dialplan Functions?
I can only find a small amount of functions for the dialplan that could 
be seen as an equivalent

or near-equivalent of an AGI Command...

thank you,
Yves


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Re: [asterisk-users] Use AGi Commands without script in Dialplan

[Yves]

Am 09.10.2018 um 13:56 schrieb Joshua Colp:

On Mon, Oct 8, 2018, at 7:44 AM, Yves wrote:

Hello, everybody,
often it is necessary to issue a single AGI command...
How can I realize this within a normal dialplan processing without
having to go the circumstantial way through an AGI script every time?
Why is it not possible to use the AGI commands like other functions
within the dialplan?
Although there are many dialplan functions that can be used as a
substitute for one or the other AGi command, or whose results are the
same, but not always...

Example:
AGI_Command "Set Autohangup"...

There is no way (at least of what I know) to set this AutoHangup feature
for a "normal" Call within the dialplan... and again, this is just an
example. I am looking
for an easy way to execute any AGI Command directly from the dialplan
without the need to call an external script.

In particular for this it can done in dialplan using the TIMEOUT dialplan 
function[1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_TIMEOUT


Hi,

thank you, great to have a replacement for this particular function.

Yves


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[asterisk-users] Use AGi Commands without script in Dialplan

[Yves]

Hello, everybody,
often it is necessary to issue a single AGI command...
How can I realize this within a normal dialplan processing without 
having to go the circumstantial way through an AGI script every time?
Why is it not possible to use the AGI commands like other functions 
within the dialplan?
Although there are many dialplan functions that can be used as a 
substitute for one or the other AGi command, or whose results are the 
same, but not always...


Example:
AGI_Command "Set Autohangup"...

There is no way (at least of what I know) to set this AutoHangup feature 
for a "normal" Call within the dialplan... and again, this is just an 
example. I am looking
for an easy way to execute any AGI Command directly from the dialplan 
without the need to call an external script.


Thank you,
Yves



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Re: [asterisk-users] Trying to add MoH to conference bridge

[Yves]

could you switch asterisk to verbose >=3 and show the output from the cli?
which version of asterisk do you use?

yves

Am 23.05.2018 um 23:23 schrieb Mike Diehl:

Hi all,

I've got an AGI script that launches the conference bridge with a line 
like:


"$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)"

The $conf variable contains the room number.

I'm trying to configure it so that when only one person is in the 
conference, they hear moh.


My /etc/asterisk/confbridge.conf looks like:

===
[general]

[default_bridge]
type=bridge

[default_user]
type=user
quiet=no
announce_join_leave=yes
music_on_hold_class=default
music_on_hold_when_empty=yes

[default_menu]
type=menu
0=playback_and_continue(/none)
1=increase_listening_volume
2=toggle_mute
3=increase_talking_volume
4=reset_listening_volume
5=admin_toggle_mute_participants
6=reset_talking_volume
7=decrease_listening_volume
8=admin_toggle_conference_lock
9=decrease_talking_volume
*=admin_kick_last
\#=participant_count
===

However, my user isn't hearing anything.  MoH does work otherwise.

What am I missing?

Thanks in advance,

Mike.




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Re: [asterisk-users] Looking for better fax handling

[Yves]

of course you can query asterisk asterisk and look, if your fax is still 
running...:


asterisk -rx "fax show sessions" lists you all acive fax sessions...

yves

Am 22.05.2018 um 12:19 schrieb D'Arcy Cain:

On 2018-05-22 02:17 AM, Yves wrote:

you could

- use "global variables"
- use the asterisk built in database

Both of those seem difficult as the process is split between Asterisk
and an external script.


- mv the file to temporary folder _before_ faxing (would be the most
easy solution as you already
know how to mv a file via asterisk...)

True.  This or John Kiniston's idea of lock files could work.  I guess I
would need to have some process to move it back if it is still there
after an hour or so in case something went wrong.  The same sort of
thing would be needed for John's solution as well.

It sure would be nice if I could query Asterisk to see if the fax
process was still running.

Thanks.




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Re: [asterisk-users] Looking for better fax handling

[Yves]

you could

- use "global variables"
- use the asterisk built in database
- mv the file to temporary folder _before_ faxing (would be the most 
easy solution as you already

know how to mv a file via asterisk...)

regards,
yves

Am 21.05.2018 um 19:49 schrieb D'Arcy Cain:

I am having troubles with sending faxes.  I hope someone can help me
work out a better method.

Basically we have a special address that our users can send to.  It
winds up on our Asterisk server which runs a Python script that parses
the message for attachments and the phone number from the recipient
address.  The attachments are converted to TIFF and stored in a folder
with various information encoded into the file name such as phone number
and retry information.  That all works fine.

A separate process picks up the files and sends them using an AMI script
like this:

Action: Originate
Channel: SIP/provider/%(destination)s
Context: LocalSets
CallerID: Vybe Consulting Inc Fax Service <6475551212>
Exten: sendfax
Priority: 1
Timeout: 3
Variable: faxfile=%(faxfile)s
Variable: uid=%(uid)s
Variable: destination=%(destination)s
Variable: sender_name=Vybe Consulting Inc Fax Service
Variable: sender_num=6475551212

It then renames the file encoding the next retry time and incrementing
the number of retries.

The same script checks for files in a success folder and sends the users
a confirmation message that the fax was sent.  The files are moved into
the success folder by Asterisk using this dialplan:

sendfax,1,Verbose(0,FAX ${faxfile} to ${destination})
   same => n,Set(FAXOPT(headerinfo)=${sender_name})
   same => n,Set(FAXOPT(localstationid)=${sender_num})
   same => n,SendFax(${faxfile},d)
   same => n,Set(STATUS=Status: ${FAXOPT(status)})
   same => n,Set(STATUS=${STATUS}\nRemote ID: ${FAXOPT(remotestationid)})
   same => n,Set(STATUS=${STATUS}\nMaxrate: ${FAXOPT(maxrate)})
   same => n,Set(STATUS=${STATUS}\nMinrate: ${FAXOPT(minrate)})
   same => n,Set(STATUS=${STATUS}\nECM: ${FAXOPT(ecm)})
   same => n,Set(STATUS=${STATUS}\nnumber of pages: ${FAXOPT(pages)})
   same => n,Set(STATUS=${STATUS}\nRate: ${FAXOPT(rate)})
   same => n,Set(STATUS=${STATUS}\nResolution: ${FAXOPT(resolution)})
   same => n,GotoIf($["${FAXOPT(status)}" = "SUCCESS"]?faxok)
   same => n,Set(STATUS=${STATUS}\nError: ${FAXOPT(error)})
   same => n(faxok),Verbose(0,FAX ${destination} Status (S): ${STATUS})
   same => n,Set(FAXNAME=${CUT(faxfile,/,6)})
   same => n,Set(FILE(/fax_status/${FAXNAME})=${STATUS})
   same => n,GotoIf($["${FAXOPT(status)}" != "SUCCESS"]?faxfail)
   same => n,System(/bin/mv '${faxfile}' '/fax_success/${FAXNAME}')

   same => n,Set(CDR(userfield)=${destination})
   same => n,Verbose(0,FAX to ${destination} charged to ${uid})
   same => n(faxfail),Verbose(0,FAX ${destination} Status (F): ${STATUS})
   same => n,Hangup()

My problem is that if the faxes get too big it starts sending it again
before the previous one has finished.  I can't raise the retry limit too
far because sometimes the receiver is busy and we have to retry in a
reasonable time.

Is there a way to get a token from the AMI script that I can use to
determine later if Asterisk is still busy with the fax before I try
sending it again?

Or, am I approaching this all wrong?  Is there a better method of doing
this?

I am running Asterisk 13.19.0 on NetBSD/amd64 7.1.0.




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Re: [asterisk-users] pcapsipdump or general sip debug question - the solution

[Yves]

Hi,

i know about this feature and use it a lot...
my question was, how to get pcapsipdebug to generate only one file...

BUT... meanwhile I found out how to accomplish this easy task.

1.) open first pcap file in wireshark
2.) open second pcap file in wireshark using the menu "file -> merge"
3.) go to "telephony -> sip flows"
4.) select the two "legs" of the call
5.) klick button "flow sequence" et voilà... one ladder diagram exactly 
the way I needed it


thanks anyways,
yves

Am 17.01.2017 um 12:34 schrieb Jean Aunis:

Hello,

There is a built-in tool in Wireshark for this : menu Telephony => 
Voip Calls, the select your call and click on "Flow Sequence".


Best regards

Jean Aunis


Le 17/01/2017 à 12:27, Yves a écrit :

Hi,

I am using pcapsipdump for debugging sip calls.

when I have to debug a call, pcapsipdump generates two files per 
call... one for the sip dialog between the client (softphone) and the 
server (asterisk) and one
for the sip dialog between the server (asterisk) and the sip 
registrar... is there a way to get this into one file ? the objective 
is to see both sides of the call in
a single ladder diagram or just to have more comfort in analyzing the 
full flow within wireshark.


If this is not possible, is there a free tool for sip (together with 
rtp) debugging that is able to catch the full sip flow between both 
ends of one call in a single file

(per call) with pcap compatibility (including the rtp packets)?

thank you
yves








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[asterisk-users] pcapsipdump or general sip debug question

[Yves]

Hi,

I am using pcapsipdump for debugging sip calls.

when I have to debug a call, pcapsipdump generates two files per call... 
one for the sip dialog between the client (softphone) and the server 
(asterisk) and one
for the sip dialog between the server (asterisk) and the sip 
registrar... is there a way to get this into one file ? the objective is 
to see both sides of the call in
a single ladder diagram or just to have more comfort in analyzing the 
full flow within wireshark.


If this is not possible, is there a free tool for sip (together with 
rtp) debugging that is able to catch the full sip flow between both ends 
of one call in a single file

(per call) with pcap compatibility (including the rtp packets)?

thank you
yves


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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

sorry... typo
the problematic phone has the 192.168.0.13
the asterisk has 192.168.1.211

when i connect a snom phone on the cable that was in the soundstation 
6000 before and configure the

phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...

it would be helpful if someone, that has a running soundstation ip 6000 
could send the configuration... :-/


regards,
yves


Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:

On Wed, Dec 21, 2016 at 7:50 AM, Yves <yves...@gmx.de> wrote:

Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... remember...
when I use tcp the phone tries to register, but does not even try with
udp...

thank you,
yves


   I am a bit confused: is your problematic phone's IP 192.168.0.13
(what the error log is reporting below) or 192.168.1.13?


Am 21.12.2016 um 13:34 schrieb Mark Wiater:

Yves,

Didn't you say that

AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:

. It is sure for 100% that there is no firewall or something else mangeling
in between... another Hardphone works as expected using the same
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?


50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
255.255.255.0

The line above suggests to me that your phone and your asterisk server are
on a different network, there has to be something that routes between those
two networks. Often what routes, can firewall.

000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
Temporarily not available



Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it 
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... 
remember... when I use tcp the phone tries to register, but does not 
even try with udp...


thank you,
yves


Am 21.12.2016 um 13:34 schrieb Mark Wiater:

Yves,

Didn't you say that


AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:
. It is sure for 100% that there is no firewall or something else 
mangeling
in between... another Hardphone works as expected using the same 
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?



50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet 
mask 255.255.255.0


The line above suggests to me that your phone and your asterisk server 
are on a different network, there has to be something that routes 
between those two networks. Often what routes, can firewall.


000122.941|sip |4|03|Registration failed User: 165, Error Code:480 
Temporarily not available





Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

Hi,

I do not have a switch to mirror the traffic... I am only remotely 
connected to the office, where all is set up.
I have full control over asterisk and the phone and I tcpdumped the 
traffic coming from the phone.
The weird thing is... if I configure the SIP-Server Setting to use TCP 
on Port 80, I see REGISTER requests.
If I configure to use UDP only on Port 5060, I do not see nothing at 
all... not a single Request coming
from the phone... and, yes... It is sure for 100% that there is no 
firewall or something else mangeling
in between... another Hardphone works as expected using the same 
Netzworkcable on the same Networkplug

with UDP on Port 5060...
Meanwhile I tried all available firmware-Versions, with and without 
provisioning. I am wondering about
downloads, the phone is trying to receive from downloads.polycom.com 
that constantly fail (yes, these

files do not exists there, the phone can communicate with the internet...)
On the other hand, I don´t think that this has something to do with the 
problem, as the phone tries to

REGISTER when I use TCP / 80

Olivier, would you mind and mail me your config-files and some 
screenshots from the phone-webconfig?

Which software-versions are you using?

thank you,
yves

if someone wants to take a look at the phone-logs:

boot-log
02.335|so   |*|01|-- Initial log entry --
02.335|so   |*|01|+++ Note that Updater log times are in GMT +++
02.335|boot |*|01|Initial log entry. Current logging level 3
02.335|copy |*|01|Initial log entry. Current logging level 3
02.335|utilm|*|01|Initial log entry. Current logging level 4
02.335|hw   |*|01|Initial log entry. Current logging level 4
02.335|ethf |*|01|Initial log entry. Current logging level 4
02.335|dns  |*|01|Initial log entry. Current logging level 3
02.335|curl |*|01|Initial log entry. Current logging level 3
02.335|sec  |*|01|Initial log entry. Current logging level 4
02.641|wdog |*|01|Initial log entry. Current logging level 4
02.641|lldp |*|01|Initial log entry. Current logging level 3
02.641|cdp  |*|01|Initial log entry. Current logging level 3
02.641|key  |*|01|Initial log entry. Current logging level 4
02.642|so   |3|01|Platform: Model=SoundStation IP 6000, 
Assembly=3111-15600-001 Rev=W Region=

02.642|so   |3|01|Platform: Board=3111-15600-001 B 0
02.642|so   |3|01|Platform: MAC=0004f2070cd3
02.643|so   |3|01|Platform: BootBlock=3.0.4.0001 (15600-001) 
11-Jul-12 08:53

02.644|so   |*|01|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56
02.644|so   |3|01|Application, main: Label=Updater, Version=Azurite 
5.0.5.2324 09-Dec-13 15:31

02.644|so   |3|01|Application, main: P/N=-Y-YYY
02.644|log  |*|01|Install file upload callback for 'Updater'

02.644|app1 |*|01|Initial log entry. Current logging level 3
02.645|cfg  |*|01|Initial log entry. Current logging level 2
02.651|app1 |3|01|Application, load: Type=SIP, Version=4.0.4.2906 
18-Apr-13 01:11

02.652|boot |*|01|Using TFFS for flash load
02.652|boot |*|01|Code length: 0x0097A585
02.652|boot |*|01|Code checksum:   0x4B86ABFB
03.631|so   |3|01|Link status is Net up Speed 100 full Duplex.
17.497|app1 |4|01|Loaded application sip.ld from local system 
successfully.


App-log
001139.870|app1 |*|03|Manual Reboot
001139.870|so   |5|03|soAudioChannel compiledOffsetsApply error: 
unrecognized verAudio 11 for headset

001140.026|so   |*|03|SoNcasC::procMsg: Client service shutdown complete
001144.025|wdog |*|03|Watchdog Expired: tSup
04.975|log  |*|03|-- Initial log entry --
04.975|so   |*|03|Platform: Model=SoundStation IP 6000, 
Assembly=3111-15600-001 Rev=W Region=

04.975|so   |*|03|Platform: Interfaceeth0 MAC=0004f2070cd3
04.977|so   |*|03|Platform: BootBlock=3.0.4.0001 (15600-001) 
11-Jul-12 08:53

04.977|so   |*|03|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56
04.977|so   |*|03|Platform: Updater=5.0.5.2324 09-Dec-13 15:31
04.977|so   |*|03|Application, main: Label=SIP, Version=Mink 
4.0.4.2906 18-Apr-13 01:11

04.977|so   |*|03|Application, main: P/N=3150-11530-404
04.977|rdisk|*|03|RAM disk created, size: 8,388,608 bytes
04.978|ocsp |*|03|O.C.S.P. Enabled = 0
04.978|tls  |*|03|Initial log entry. Current logging level 4
04.998|pmt  |*|03|Initial log entry. Current logging level 4
04.998|wdog |*|03|Initial log entry. Current logging level 4
04.998|ethf |*|03|Initial log entry. Current logging level 4
04.998|hw   |*|03|Initial log entry. Current logging level 4
04.998|ares |*|03|Initial log entry. Current logging level 4
04.998|dns  |*|03|Initial log entry. Current logging level 4
04.998|cfg  |*|03|Initial log entry. Current logging level 4
04.998|dot1x|*|03|Initial log entry. Current logging level 4
05.000|cfg  |*|03|RT|Network eth0 link went up
05.000|cfg  |*|03|RT

[asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

Hi,

I am pulling my hair for days now...

I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register 
with my Asterisk.


There are no SIP Packets arriving at my asterisk at all... and it has 
nothing to do with a firewall or similar...


Simple Question:
Does anybody have a running SoundStation IP 6000 registerd with asterisk?
If so... would you please be so kind to tell me whats wrong with my setup?

AsteriskServer: 192.168.1.211
SIP-user: 165

(the SIP-Settings on asterisk-side are OK, tested with a normal 
Softphone... registering and placing calls is no problem...)


The phone-log only says: "Registration failed User: 165, Error Code:480 
Temporarily not available"


I tried with newest firmware, resetting to factory 100 times, using a 
provisionig file (which the SoundStation correctly downloads)
but it is always the same... the SoundStation does not contact the 
asterisk for registering...


Phoneversion:
Telefoninformationen
Telefonmodell   SoundStation IP 6000
Teilenummer 3111-15600-001 Rev:W
MAC-Adresse 00:04:F2:07:0C:D3
IP-Adresse  192.168.0.13
UC-Softwareversion  4.0.11.0583
BootROM-Softwareversion 5.0.5.2324


I can ping the phone from the asterisk, the phone can reach the asterisk 
server (as it downloads the tftp files, if used with
a provisioning profile), so the route and everything is correct... I 
even connected another Hardphone on the same cable
that stuck in the Polycom... no problem... the other phone can register 
and works, so there is really no cable or firewall

related problem here... it must be a setting!

thank you



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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Yves]

ok,

thank you... then I´ll take it as it is

cheers,
yves

Am 18.12.2016 um 13:15 schrieb Larry Moore:

Hi,

I haven't found anything definitive however I expect the TSI that is 
sent during initial fax call establishment is stored by the receiving 
terminal, see pages 28 & 29 of the English version of the document at 
https://www.itu.int/rec/T-REC-T.30-200509-I/en , I expect the header, 
which will include the TSI, is all part of the image (Tagline in 
HylaFAX) and not stored separately on the receiving terminal.


Cheers,

Larry.

On 18/12/2016 6:20 PM, Yves wrote:

Hi,

thanks for your answer. Unfortunately this is, what I already know. I 
was wondering, why it is possible to set ID and Header for an 
outgoing fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while 
it seems to not be possible to get the Header from a received fax 
(only the id), although it is present in the faxdocument.
The ID is also present in the faxdocument and there does a 
Faxopt(remotestationid) exist... so I thought, this info must be 
transferred not only binary within the "image", but
also within the "meta-data" / protocol-data of the fax (within the 
TSI) otherwise asterisk must do some kind of ocr to get the ID, 
what it definitely does not...


btw... when using sendfax, asterisk inserts the date, the id, the 
header and the pagenum on top of each faxpage... someone knows how to 
modify some settings like font, position, and so on?


thanks,
yves


Am 18.12.2016 um 00:02 schrieb Larry Moore:
The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I 
haven't checked for this.


From my days working with fax machines, the header could be inserted 
in the line the TSI is on or in the image being transmitted, if you 
receive a fax that has been sent to you with the latter set, then 
the 'headerinfo' will not be of any use. Perhaps someone with more 
knowledge may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


To specify the position of Header Position printed on a sent
fax ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and 
the headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo 
besides the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves
















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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Yves]

Hi,

thanks for your answer. Unfortunately this is, what I already know. I 
was wondering, why it is possible to set ID and Header for an outgoing 
fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while it 
seems to not be possible to get the Header from a received fax (only the 
id), although it is present in the faxdocument.
The ID is also present in the faxdocument and there does a 
Faxopt(remotestationid) exist... so I thought, this info must be 
transferred not only binary within the "image", but
also within the "meta-data" / protocol-data of the fax (within the 
TSI) otherwise asterisk must do some kind of ocr to get the ID, what 
it definitely does not...


btw... when using sendfax, asterisk inserts the date, the id, the header 
and the pagenum on top of each faxpage... someone knows how to modify 
some settings like font, position, and so on?


thanks,
yves


Am 18.12.2016 um 00:02 schrieb Larry Moore:
The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I haven't 
checked for this.


From my days working with fax machines, the header could be inserted 
in the line the TSI is on or in the image being transmitted, if you 
receive a fax that has been sent to you with the latter set, then the 
'headerinfo' will not be of any use. Perhaps someone with more 
knowledge may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


To specify the position of Header Position printed on a sent
fax ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and 
the headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo besides 
the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves








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[asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Yves]

Hi,

I am using asterisk 11.8 in combination with spandsp to send and receive 
T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and the 
headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo besides 
the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves


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[asterisk-users] WhatsApp feature on Asterisk

[Yves biganiro]

Can anyone put light on whatsapp features   and how it can be operated .
What are the technology that need to be installed ,

Regards

Yves
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Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Yves biganiro]

Asterisk 1.8.23.0-1_centos5.go

DAHDI Version: 2.6.1 Echo Canceller: HWEC

On Wed, Jul 20, 2016 at 5:32 PM, A J Stiles <asterisk_l...@earthshod.co.uk>
wrote:

> On Wednesday 20 Jul 2016, Yves biganiro wrote:
> > Hi all
> >
> > Hi,I'm facing a strange  issue where by SANGOMA not detected  by
> goautodial
> > system ,
>
> Is this some kind of one-stop, pre-prepared distribution with Linux,
> Asterisk,
> DAHDI, a web server and some custom scripts, that all installs from one
> place?
>
> We really need to know your Asterisk and DAHDI versions.
>
> Type in a root terminal,
>
> # asterisk -V
>
> and note the version number displayed  (it will be on the first line).
> Then
> enter
>
> *CLI> dahdi show version
>
> and note the DAHDI version displayed.
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Senior IT Consultant - independent

Tel +250727612605


##A tech entrepreneur and web developer, Passionate about technology with
working experience in web development.##
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Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Yves biganiro]

I have  forcefully installed everything but it says that the card is not
found.

On Wed, Jul 20, 2016 at 5:05 PM, Yves biganiro <yves.bigan...@gmail.com>
wrote:

> Hi all
>
> Hi,I'm facing a strange  issue where by SANGOMA not detected  by
> goautodial system ,  Thats the problem :
>  Configuring ISDN BRI cards [A500/B700]
> 
>
> No Sangoma ISDN BRI cards detected
>
> Press any key to continue:
> 
> Configuring GSM cards [W400]
> 
>
> No Sangoma GSM cards detected
>
>
>
> regards
>
>
>
>
>


-- 

Yves Biganiro

Senior IT Consultant - independent

Tel +250727612605


##A tech entrepreneur and web developer, Passionate about technology with
working experience in web development.##
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[asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Yves biganiro]

Hi all

Hi,I'm facing a strange  issue where by SANGOMA not detected  by goautodial
system ,  Thats the problem :
 Configuring ISDN BRI cards [A500/B700]


No Sangoma ISDN BRI cards detected

Press any key to continue:

Configuring GSM cards [W400]


No Sangoma GSM cards detected



regards
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[asterisk-users] open source pbx free

[Yves biganiro]

Anyone have any experience running an open source pbx and call center
solution?Need to start a call center of 10 users  and i need help

I have already  installer a server with Ubuntu Server 14.04  , E1 installed


Please advice me how to process  from here

Regards

Yves
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Re: [asterisk-users] my dahdi dont'n start

[Yves]

Hello,
I was faced with this problem, it is enough to place
subdirectory under ./tools installation dahdi when compiling and
run make install-config it should work.

we must have :
mkdir -p / etc / dahdi
mkdir -p /etc/modprobe.d
install -m644 xpp / genconf_parameters / etc / dahdi / genconf_parameters
install -m644 init.conf.sample /etc/dahdi/init.conf
install -m644 blacklist.sample /etc/modprobe.d/dahdi-blacklist.conf
install -m644 modprobe.conf.sample /etc/modprobe.d/dahdi.conf
make -f ./Makefile.legacy top_srcdir =. srcdir =. config
make [1]: Entering directory 
'/usr/src/dahdi-linux-complete-2.11.1+2.11.1/tools'

install -D dahdi.init /etc/init.d/dahdi
/usr/sbin/update-rc.d dahdi defaults 15 30
DAHDI has-been configured.

Le 28/04/2016 16:37, A J Stiles a écrit :

On Thursday 28 Apr 2016, Mamadou NGOM wrote:

Hello,
  it doesn't work my dahdi yet .for information, i use debian 8 .
I put the file dahdi.bash   in /etc/init.d and I gave it the permission 755
but i have  the same error: bash: /etc/init.d/dahdi: No such file or
directory

You need to name the file just "dahdi", not "dahdi.bash"; because the command
"service dahdi start" is looking for a file just called "dahdi".  If you run
# mv /etc/init.d/dahdi.bash /etc/init.d/dahdi
then
# service dahdi start
should work.

You probably also need to run
# update-rc.d dahdi defaults
to ensure it starts up everytime the computer is booted up.




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Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

[Yves]

Le 06/04/2016 18:12, Markos Vakondios a écrit :

Good evening,

My English is limited but if I can help.

We install Asterisk Version 13.1 on VmWare with Debian 8.2, no
problem since June 2015, currently I have tested on Unbutu 14.04 but 
problem with network-manager (problem of stability with Asterisk 1.8.32 
and difficulty with routing network-manager).


I also installed Asterisk on KVM (Debian 8.2) no problem (but not test 
with dahdi) without particular problem.


here is my little opinion

Hello everyone

Proxmox and KVM on Ubuntu

On Wednesday, 6 April 2016, Ryan, Travis > wrote:


What is the best virtual server tech (and most stable, etc) to use
for a asterisk virtual hosting environment?

I have a client that wants to do virtual hosting of Asterisk (only
SIP or IAX, no PRI, etc) and I’m wondering if Xen or something
else would be best? We’d like to stay away from the costs of
VMWare if possible.

Thanks!

Travis

//





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Re: [asterisk-users] The To header was truncated in call... Whats this means?

[Yves]

I have seen these messages only on asterisk boxes that are open to 
public and I think this may have something to
do with sip-attacks... I´d recommend some wiresharking or at least sip 
debugging...


yves

Am 07.01.2016 um 21:23 schrieb Vitor Mazuco:

Hi everybody,

My Asterisk, all time appear this log

[Jan  7 15:37:04] ERROR[1174] chan_sip.c: The To header was truncated
in call '6c66e5b6058ae257003c0f7e778da0fe@191.x'. This call
setup will fail.
[Jan  7 15:37:18] ERROR[1174] chan_sip.c: The To header was truncated
in call '18e0a12e434364254b0cc2e52d20755b@191.x. This call setup
will fail.
...

Whats this massege means?

Thanks.




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[asterisk-users] placing calls with linphone.org SIP account

[Yves]

Happy new year!

maybe off-topic... but maybe someone _knows_ a solution.

I have a free SIP Account at linphone.org... calling other linphone.org 
users via SIP and receiving SIP calls from other users
registered at linphone.org is no problem... just "dial" the username... 
BUT... as far as I understand the documentation, linphone.org
offers internet wide SIP Calls... so... how can I call other users 
registered at other SIP-Providers?
I tried all well-known SIP URI Syntaxes but none worked... does anyone 
reliably know, if it is possible at all and if so, what is the

dialstring looking like? (I am trying with zoiper softphone)

Unfortunately there is no support-email-address for linphone.org users...

thanks,
yves

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[asterisk-users] No QueueCallerJoin Event...

[Yves]

Hi,

I am using asterisk 13.5.0 and although my AMI-user has read=all and 
write=all permissions, I don´t get any QueueCallerJoin Events fired, 
when a new caller calls into a Queue...
Strange enough, a QueueCallerAbandoned Event is fired, when the caller 
hangs up without beeing connected to an agent...

Is it a bug or am I missing something?

regards,
Yves


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Re: [asterisk-users] asterisk-java is dead?

[Yves]

No, its not dead and mails to the asterisk-java-list become replied.

regards,
yves

Am 18.09.2015 um 02:35 schrieb symack:

Hello Everyone,

I am trying to make use of asterisk-java live and had some questions
for the mailing list however, it does not seem like it's an active
mailing list? Is the project dead?

Thanks,

Nick






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[asterisk-users] call failed... but why? What means SIP_ALREADYGONE?

[Yves A.]

Hi,

I have watched a phenomen, that I can not explain... maybe one of you 
can see the reason why the call failed, and if the cause
is the Snom Hardphone, or the asterisk, or the SIP-Provider... the debug 
log given below is all I have...

What does Setting SIP_ALREADYGONE on dialog.. mean?

thanks for watching,
yves

SIP Phone 110 (callerid 061444018110) tried to call the external Phone 
Number 0616677823 and gets an hangup after 2 seconds. Another try 
immediately
after the failed call goes fine. The failed call did not arrive at the 
destination.


[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Begin: 
parsing SIP Supported: timer, 100rel, replaces, from-change
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found 
SIP option: -timer-
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched 
SIP option: timer
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found 
SIP option: -100rel-
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched 
SIP option: 100rel
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found 
SIP option: -replaces-
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched 
SIP option: replaces
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found 
SIP option: -from-change-
[Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched 
SIP option: from-change
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Trying to put 
'SIP/2.0 401' onto UDP socket destined for 192.168.0.165:3072
[Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL 
RealTime: Connection okay.
[Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = 
'00616677823' AND h

ost = 'dynamic'
[Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL 
RealTime: Connection okay.
[Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823'
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Stopping 
retransmission on '9a6bdc548d19-goay25ioz0nd' of Response 1: Match Found
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Using engine 
'asterisk' for RTP instance '0x7f2a74158788'
[Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Allocated 
port 19528 for RTP instance '0x7f2a74158788'
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: RTP instance 
'0x7f2a74158788' is setup and ready to go
[Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Setup RTCP 
on RTP instance '0x7f2a74158788'
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Setting NAT on RTP 
to On
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
session-level SDP v=0... UNSUPPORTED OR FAILED.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
session-level SDP o=root 871055034 871055034 IN IP4 192.168.0.165... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
session-level SDP s=call... UNSUPPORTED OR FAILED.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
session-level SDP c=IN IP4 192.168.0.165... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
9 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
0 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
8 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
99 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
108 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
18 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 
101 based on m type on 0x7f2a80b1a620
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=rtpmap:99 G726-32/8000... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=rtpmap:108 AAL2-G726-32/8000... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing 
media-level (audio) SDP a=fmtp:18 annexb=no... OK.
[Feb 12 10:00:11] DEBUG[1567][C-380e

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

Hi,

I know this Bug,,, at least when you´re talking about x-lite 3... quite 
annoying, but if you know it...
so no... its not the phone... tested with zoiper and 3cx ... both 
work...but the problem occurs ONLY,

as soon as I register at more than one registrar...

yves

Am 22.11.2014 um 19:19 schrieb Ron Wheeler:

You might check your phones as well.
We had this problem early on with a softphone and it was a setting in 
the phone that was set to hang up after 30 seconds of inactivity in 
case of network disruption. For some reason it was detecting network 
disruption in every call even when the calls were proceeding normally.

Unchecking this box solved the problem.

It may not be related to your problem but if it is the cause, you will 
spend a lot of time trying to fix this in Asterisk. :-D At least I did!


On the bright side, it does force people to get point in a hurry!

Ron

On 22/11/2014 12:50 PM, Eric Wieling wrote:

Try setting directmedia=no in sip.conf.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.

Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but 
only when


Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

but as soon as I configure another sip registration on another server,
outgoing
calls  drop after 32 seconds.

Are both your servers behind the same NAT router?


thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net

and the other one is

sip.ovh.fr

how can i determine and how could that affect... I mean... why do they
interfere at all?

thanks,
yves

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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

Hi,

the useragents nothing to do with the problem... i tried numeric, alpha 
and alphanumeric... no difference.

they work all as long as I only use ONE registrar...
as soon as I register at more than one registrar... the line drops after 
32 seconds really strange.


yves

Am 22.11.2014 um 19:01 schrieb Rafael Visser:


Hi Yves..
This may be silly... but what is the useragent of your sip configuration?
In the case that useragent has some special characters like (., 
please remove it and tell us if there is any change!!.

Regards.
rv


2014-11-22 14:50 GMT-03:00 Eric Wieling ewiel...@nyigc.com 
mailto:ewiel...@nyigc.com:


Try setting directmedia=no in sip.conf.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but
only when

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
 but as soon as I configure another sip registration on another
server,
 outgoing
 calls  drop after 32 seconds.
 Are both your servers behind the same NAT router?

thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net http://siptrunk.ovh.net

and the other one is

sip.ovh.fr http://sip.ovh.fr

how can i determine and how could that affect... I mean... why do they
interfere at all?

thanks,
yves

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[asterisk-users] how to set timerb in sip.conf

[Yves A.]

Hi list,

I have tried to set the value for timerb in sip.conf, general section 
and in user-context...
tried on asterisk 1.4 up to version 13... no success. The value for 
timerb remains unchanged.
(reload, restart, reboot all does not help...) sip show settings 
always show 32000ms for

timerB.
How can I configure the timerb value?

thx,
yves



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[asterisk-users] Softphone signals busy although it isn´t

[Yves A.]

Hi,

I have written an click2dial application that rings an agent soft phone 
and connects the agent

with a customer.
very often I can see, that the agent softphones signal a busy back to 
server, although the phone

is definitely hung up and the previous calls where handled normally.
I testet 3cx Version 6 and X-Lite V1 up to V3... all show the same 
misbehaviour.
I did a SIP Trace and can see, the phone replies with a busy on the 
invite... but I don´t know why.
Has anybody experienced similar things and knows a reason or a 
workaround for this?

I am working on a asterisk 11.7 using sip-realtime peers with mysql.

thanks for reading,
yves


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[asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

hi,

I have a really strange problem which is driving me crazy for days now.

If I register my asterisk (tried all versions from 1.6 up to 13.x) with 
one sip registrar,

everything works... calls go out and call come in... no 32 seconds limit.

but as soon as I configure another sip registration on another server, 
outgoing

calls  drop after 32 seconds.

as far as I know, there is no firewall in between...

I tried to work around this by increasing the settings for timerb... 
but I

realized that asterisk does not care at all, what I set this value to...
sip show settings always gives me 32000ms, and it does not make any
difference if I configure timerb in the general context or in the phone 
context...


any ideas?

thanks,
yves

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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

but as soon as I configure another sip registration on another server,
outgoing
calls  drop after 32 seconds.

Are both your servers behind the same NAT router?


thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net

and the other one is

sip.ovh.fr

how can i determine and how could that affect... I mean... why do they 
interfere at all?


thanks,
yves

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[asterisk-users] Best strategy to find and solve voice quality problems

[Yves A.]

Hi,

in my company we use an asterisk installation with around 50 soft- and 
hardphones of all kind.
From time 2 time the users (almost only Softphone users) report some 
voice qualities... mostly echoes.
These problems do not occur on all PCs at the same time and since setup 
of our PBX almost any PC user

has gotten these issues.
When I come there to check, everything is fine again... and I can´s see 
anymore, what could have caused
the problem... may it be a high network load, or a high cpu usage or 
whatever...
I activated call recording to hear the quality after such 
missing-quality reports but every call I listened
to showed no issues in the recording so I assume the problem is on the 
client side. Because it is not
always the same user or the same PC I think it cannot be a misbehaviour 
like wrong headset usage or

a problem of a single PC.

What is the best strategy to find and solve these kind of problems? Are 
there any (free would be cool) tools
that can monitor the pc-state (concerning at least network and cpu- / 
process usage) over a long period

and display the results in an appropriate way?

Is there a way under Windows XP / 7 to ensure Bandwidth for VoIP like 
QoS (google only showed me such

settings for Lync or Windows Server machines...?

Is there a way under Windows XP / 7 to ensure CPU-Bandwidth for 
Applications (like VoIP Clients)?


Thanks for any hint,
yves

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Re: [asterisk-users] how to send dtmf after pause ?

[Yves A.]

This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play the 
dtmf tones and bridge the call to your extension afterwards.


yves

Am 07.06.2013 17:51, schrieb Sean Darcy:


I'm trying to call a conference service, wait 10 seconds, then send 
the passcode.


I've tried ww:

Dial(SIP/18005551212ww12345#@sip.com,60,r)

The sip channel didn't like that. Added 'p' , still no help.

I tried D:

Dial(SIP/18005551...@sip.com,60,rD(12345#)

The dtmf is sent too soon. I tried inserting 'ww' but that was just sent.

I tried G:

exten = 234.1.Dial(SIP/18005551...@sip.com,60,rG(next))
 same=n(next),Wait(10)
 same=n,SendDTMF(12345#)

but that didn't work at all,

This is a common use case. There must be some simple answer I'm missing.

Thanks for any help.

sean


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Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

[Yves A.]

looks yummy indeed... but how does it interact with an asterisk? phono 
uses afaiu voxeo-cloud to make place calls, send sms and so on...
I do not see a way to use phono without their cloud services, not did I 
see any hint about charges for calls...


yves

Am 03.06.2013 12:34, schrieb Lenz Emilitri:

Looks yummy! http://phono.com/webrtc


2013/5/31 Adnan 112linuxstockh...@gmail.com 
mailto:112linuxstockh...@gmail.com


Voxeo/Phono webrtc.

/Adnan


On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri
lenz.lo...@gmail.com mailto:lenz.lo...@gmail.com wrote:


Hi All,
I wonder if any of you has some suggestions on which WebRTC
client/softphone to use for a click-to-dial, webpage hosted
solution. Any suggestions?
Thanks
l.
-- 
Loway - home of QueueMetrics - http://queuemetrics.com

Test-drive WombatDialer beta @ http://wombatdialer.com

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Re: [asterisk-users] Help me understand these log messages

[Yves A.]

... an anonyous (not registerted) sip user from 188.161.238.232 was 
trying to initiate a call to

9725955 and so on...
you could enable sip tracing to get more information.

maybe you should change the 'allowguest' option in sip.conf..?

regards,
yves

Am 31.05.2013 23:57, schrieb Chris Gentle:

OK, I need a bit of help here.  I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console.  Obviously somebody was trying to take advantage of
my carelessness.  So can someone explain what would cause these types
of messages to show up on my console?

I understand that my iptables would have stopped this but I'm just
trying to understand more about the problem.  What other settings
might have stopped this?  Fail2ban was running but there were no
failed registration type messages that would have triggered it.

[May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '00972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '000972595595767' rejected
because extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.
snip


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Re: [asterisk-users] Executing a dynamic sequence of applications

[Yves A.]

Hi,

I would recommend an AGI-script or a realtime dialplan for this purpose.

yves

Am 30.05.2013 11:46, schrieb Grant Bagdasarian:


Hello,

I'm researching the possibilities of multiple communication platforms 
like Asterisk and FreeSwitch for handling a dynamic sequence of 
applications to execute, like Playback, Read, etc.


This only applies to originating a call from an external application 
by using the AMI Manager and the Originate action.


I need to know the following:

1)Does the Originate action support multiple Application keys? If so, 
how does it handle the order in which they're added to the Originate 
action?


2)If it does not support multiple Application keys, I'll have to 
instruct the Originate action to enter a context in the dialplan, and 
pass the sequence of applications in its Variable key. How would I 
configure the dialplan context to dynamically handle the sequence of 
applications to execute? I was thinking of creating a separate 
priority label for each required Application and have each application 
in the Variable key routed to the correct priority label. Is this 
possible? Are there alternatives for doing what I require?


Regards,

Grant



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Re: [asterisk-users] Sangoma Wanpipe Driver

[Yves A.]

- solved -
it turned out that libpri was not compiled correctly...
and... Asgars comment about group systax is correct.

thx 
regards,
yves
Am 13.05.2013 13:21, schrieb Yves A.:

that was the syntax before 1.8 or 11.x I think...

what about pseudo?

yves

Am 13.05.2013 13:16, schrieb Asghar Mohammad:
Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or 
Dial(DAHDI/R0/number or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number?



On Mon, May 13, 2013 at 12:53 PM, Yves A. yves...@gmx.de 
mailto:yves...@gmx.de wrote:


mmh... actually supportline is closed...

why proceeds the call to dahdi/pseudo-??

i have never seen this before...

thx.,
yves

Am 13.05.2013 11:42, schrieb Duncan Turnbull:

We have had challenges with the latest kernel versions on
Ubuntu and sangoma wanpipe drivers

An older kernel - no problem, latest ones, sometime risky.
There are release notes on their site stating the supported
versions so it might pay to check that

But if it compiled ok it might be something else

Sangoma support will dial in and help you if you ask them

Cheers Duncan

On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de
mailto:yves...@gmx.de wrote:

Hi,


I migrated from asterisk 1.6 to 11.3.
The Server has a Sangoma A104 quadPri card installed. OS
is a fresh installed Ubuntu 12.04 64bit
libpri, dahdi etc. all latest releases..

Sangoma says... driver is compatible with ANY asterisk
version...

I tried driver 3.5.8... Setup ended with error.
I tried (latest) driver 7.0.1 Setup went through,
Asterisk is showing dahdi channels... all fine I
thought... but..:

when dialing
Dial(DAHDI/i0/number)

it accepts the call, but generates a DAHDI/Pseudo channel
and the call goes not into the PSTN...

What am I doing wrong?

Has anybody successfully compiled sangoma driver 7.0.1 in
combination with an asterisk 11.3?

thanks for hints,
regards,
yves

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Re: [asterisk-users] amiDebugger - might make your life easier if you program through the AMI

[Yves A.]

thank you!
such efforts for the community are always highly appreciated! - I´ll 
give it a try.


regards,
yves
Am 13.05.2013 21:44, schrieb Lenz Emilitri:


Hi all,
I have been playing with the AMI quite a bit lately - mostly debugging 
WombatDialer in production, but that's a different story - and I have 
been frustrated by the lack of a simple way to interact CLI-like with 
the AMI itself. So I have decided to write something myself to make my 
life easier, or at least a bit less miserable.


The result is a little webapp that you can use as a sort of 
CLI-frontend to the AMI itself. It is not pretty, but pretty much 
effective. So I thought I could share it and make someone else's life 
a bit easier.


You can find it on https://github.com/l3nz/amiDebugger  - if you just 
want to test-drive it get the WAR file an put it into some webapp 
container, e.g. Tomcat.


Hope you'll like it.
l.


--
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Test-drive WombatDialer beta @ http://wombatdialer.com


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[asterisk-users] Sangoma Wanpipe Driver

[Yves A.]

Hi,


I migrated from asterisk 1.6 to 11.3.
The Server has a Sangoma A104 quadPri card installed. OS is a fresh 
installed Ubuntu 12.04 64bit

libpri, dahdi etc. all latest releases..

Sangoma says... driver is compatible with ANY asterisk version...

I tried driver 3.5.8... Setup ended with error.
I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing 
dahdi channels... all fine I thought... but..:


when dialing
Dial(DAHDI/i0/number)

it accepts the call, but generates a DAHDI/Pseudo channel and the call 
goes not into the PSTN...


What am I doing wrong?

Has anybody successfully compiled sangoma driver 7.0.1 in combination 
with an asterisk 11.3?


thanks for hints,
regards,
yves

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Re: [asterisk-users] Sangoma Wanpipe Driver

[Yves A.]

mmh... actually supportline is closed...

why proceeds the call to dahdi/pseudo-??

i have never seen this before...

thx.,
yves

Am 13.05.2013 11:42, schrieb Duncan Turnbull:

We have had challenges with the latest kernel versions on Ubuntu and sangoma 
wanpipe drivers

An older kernel - no problem, latest ones, sometime risky. There are release 
notes on their site stating the supported versions so it might pay to check that

But if it compiled ok it might be something else

Sangoma support will dial in and help you if you ask them

Cheers Duncan

On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de wrote:


Hi,


I migrated from asterisk 1.6 to 11.3.
The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed 
Ubuntu 12.04 64bit
libpri, dahdi etc. all latest releases..

Sangoma says... driver is compatible with ANY asterisk version...

I tried driver 3.5.8... Setup ended with error.
I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi 
channels... all fine I thought... but..:

when dialing
Dial(DAHDI/i0/number)

it accepts the call, but generates a DAHDI/Pseudo channel and the call goes not 
into the PSTN...

What am I doing wrong?

Has anybody successfully compiled sangoma driver 7.0.1 in combination with an 
asterisk 11.3?

thanks for hints,
regards,
yves

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Re: [asterisk-users] Sangoma Wanpipe Driver

[Yves A.]

that was the syntax before 1.8 or 11.x I think...

what about pseudo?

yves

Am 13.05.2013 13:16, schrieb Asghar Mohammad:
Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or 
Dial(DAHDI/R0/number or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number?



On Mon, May 13, 2013 at 12:53 PM, Yves A. yves...@gmx.de 
mailto:yves...@gmx.de wrote:


mmh... actually supportline is closed...

why proceeds the call to dahdi/pseudo-??

i have never seen this before...

thx.,
yves

Am 13.05.2013 11:42, schrieb Duncan Turnbull:

We have had challenges with the latest kernel versions on
Ubuntu and sangoma wanpipe drivers

An older kernel - no problem, latest ones, sometime risky.
There are release notes on their site stating the supported
versions so it might pay to check that

But if it compiled ok it might be something else

Sangoma support will dial in and help you if you ask them

Cheers Duncan

On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de
mailto:yves...@gmx.de wrote:

Hi,


I migrated from asterisk 1.6 to 11.3.
The Server has a Sangoma A104 quadPri card installed. OS
is a fresh installed Ubuntu 12.04 64bit
libpri, dahdi etc. all latest releases..

Sangoma says... driver is compatible with ANY asterisk
version...

I tried driver 3.5.8... Setup ended with error.
I tried (latest) driver 7.0.1 Setup went through, Asterisk
is showing dahdi channels... all fine I thought... but..:

when dialing
Dial(DAHDI/i0/number)

it accepts the call, but generates a DAHDI/Pseudo channel
and the call goes not into the PSTN...

What am I doing wrong?

Has anybody successfully compiled sangoma driver 7.0.1 in
combination with an asterisk 11.3?

thanks for hints,
regards,
yves

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Re: [asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1

[Yves A.]

hi,

i would try to make a symlink... link the wrong folder to the correct one...

yves

Am 02.05.2013 23:34, schrieb James Mortensen:

Hello,

I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead 
of 2.0 due to a crashing issue resulting from ICE. 
https://issues.asterisk.org/jira/browse/ASTERISK-21696


Currently, I'm systematically going through each Makefile in every 
directory in pjproject and changing the paths that exist in the 
pjproject 2.0 included with Asterisk, so that I can successfully build 
Asterisk.


I'm using the Asterisk pjproject 2.1 port from here: 
https://github.com/asterisk/pjproject


An example of the build errors I'm resolving one by one is this:

make[2]: *** No rule to make target 
`../../pjlib/lib/libpj-x86_64-unknown-linux-gnu.a', needed by 
`../lib/libpjnath-x86_64-unknown-linux-gnu.a'.  Stop.
make[1]: *** 
[/mnt/src/asterisk-11.4.0-rc1/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] 
Error 2

make: *** [res] Error 2

I'm editing the Makefiles and fixing the paths so Asterisk can find 
the target.  For all the people out there smarter than me, is there a 
better way to go about this?


I'm hoping upgrading PJSIP will resolve the crashing issue, and I'll 
continue going through Makefiles until someone smarter than me can 
enlighten me.


Thank you for your help!

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866-707-4590
james.morten...@voicecurve.com mailto:james.morten...@voicecurve.com


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Re: [asterisk-users] looking for a way to do appointment reminders

[Yves A.]

Hi Brandon,

as you are asking for professional help for a commercial project, I 
would recommend you to place a bounty.
You can contact me directly if you want my professional help... I have 
developed exactly what you´re looking
for and this solution is running in a high-call-volume installation 
without any issues.

regards,
yves

Am 26.04.2013 03:55, schrieb Brandon Coale:

Hello,

My health care organization is looking for a way to do appointment 
reminders.  We currently have staff members who spend part of each day 
manually calling patients to remind them of their upcoming 
appointments, and we would like to automate this process.


Our electronic health record software would provide such information 
as the patient's name, phone number, and day and time of the 
appointment, and Asterisk could take this information and place an 
automated call to the patient.  We would like the reminder call to use 
text-to-speech to personalize the call, such as We have an 
appointment reminder for [first name].  The appointment is on [date] 
at [time].


I am wondering if anyone has experience with using Asterisk for this 
type of application, and would be willing to share any details of how 
you implemented it?  I am interested in any ideas, from very simple to 
feature-rich.  We would be doing a new installation of Asterisk for 
this purpose, so we could use any version of Asterisk you would 
recommend.


Thank you!
Brandon


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[asterisk-users] PRI DEBUG

[Yves A.]

hi,

strange behaviour while trying to use pri debugging on asterisk 11.x ...

please take a look:

bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
*pri intense debug span*no description available
   pri service disable channel Remove a channel from service
pri service enable channel Return a channel to service
*pri set debug {on|off*|hex|inte Enables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
 pri show channels Displays PRI channel information
*pri show debug*Displays current PRI debug settings
pri show spans Displays PRI span information
 pri show span Displays PRI span information
  pri show version Displays libpri version
bas1104*CLI help dahdi
 dahdi destroy channel Destroy a channel
 dahdi restart Fully restart DAHDI channels
 dahdi set dnd Sets/resets DND (Do Not Disturb) mode on 
a channel

  dahdi set hwgain Set hardware gain on a channel
  dahdi set swgain Set software gain on a channel
   dahdi show cadences List cadences
dahdi show channels [group|con Show active DAHDI channels
dahdi show channel Show information on a channel
 dahdi show status Show all DAHDI cards status
dahdi show version Show the DAHDI version in use
/
//currently all debug off:/

bas1104*CLI pri show debug
Span 1: Debug: No   Intense: No
Span 2: Debug: No   Intense: No
Span 3: Debug: No   Intense: No
Span 4: Debug: No   Intense: No
/
//switching it on (which currently works as expected)/


bas1104*CLI pri intense debug span 1
Enabled debugging on span 1
/
//
//oops, still shows no debug but it IS activated.../

bas1104*CLI pri show debug
Span 1: Debug: No   Intense: No
Span 2: Debug: No   Intense: No
Span 3: Debug: No   Intense: No
Span 4: Debug: No   Intense: No
/
//huh... how to disable it again? on some machines I can do so with pri 
no debug span nr but not here... gives same result (no//

//such command) and debug is still enabled.../

bas1104*CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set' for 
other possible commands)

bas1104*CLI


so... whats the right way to disable pri debugging?

thx,
yves


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Re: [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes

[Yves A.]

Hi,

I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and 
would say it is a bug...

To remotely hang up a call use
*
**hangup request channel*

where channel is the exact id of your channel as you would receive it via

*core show channels*

yves

Am 11.04.2013 10:56, schrieb Thorsten Göllner:

Hi,

I have the following setup:

Ubuntu 12.04.02 LTS (64 bit)
Asterisk 11.2.1
Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected)
WANPIPE Release: 3.5.28
DAHDI Version: 2.6.1 Echo Canceller: HWEC
libpri version: 1.4.12

I call via sip into the dialplan. Then I do a 
Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is 
fine. dahdi show channels shows me, that channel 1 is used for the 
outcall. Then I try to hangup the outcall via dahdi destroy channel 
1. Asterisk crahes immediatly. No message is logged (verbose is 10 
and debug is 10).


I get disconnected from the atserisk cli at this moment:

vlr-3*CLI dahdi destroy channel 1
vlr-3*CLI
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
voxi@vlr-3:/tmp$

Is this a bug or is this my fault?

Best regards
-Thorsten-

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Re: [asterisk-users] PRI DEBUG

[Yves A.]

thanks, that command syntax works.

yves

Am 11.04.2013 18:51, schrieb Richard Mudgett:


- Original Message -

hi,

strange behaviour while trying to use pri debugging on asterisk 11.x
...

please take a look:

bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
pri intense debug span no description available
pri service disable channel Remove a channel from service
pri service enable channel Return a channel to service
pri set debug {on|off |hex|inte Enables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
pri show channels Displays PRI channel information
pri show debug Displays current PRI debug settings
pri show spans Displays PRI span information
pri show span Displays PRI span information
pri show version Displays libpri version
bas1104*CLI help dahdi
dahdi destroy channel Destroy a channel
dahdi restart Fully restart DAHDI channels
dahdi set dnd Sets/resets DND (Do Not Disturb) mode on a channel
dahdi set hwgain Set hardware gain on a channel
dahdi set swgain Set software gain on a channel
dahdi show cadences List cadences
dahdi show channels [group|con Show active DAHDI channels
dahdi show channel Show information on a channel
dahdi show status Show all DAHDI cards status
dahdi show version Show the DAHDI version in use

currently all debug off:

bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No

switching it on (which currently works as expected)


bas1104*CLI pri intense debug span 1
Enabled debugging on span 1


oops, still shows no debug but it IS activated...

It activated a different mode of debug than what you expected
because that command is an alias that was not updated.


bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No

huh... how to disable it again? on some machines I can do so with
pri no debug span nr but not here... gives same result (no
such command) and debug is still enabled...

bas1104*CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set'
for other possible commands)
bas1104*CLI


so... whats the right way to disable pri debugging?

The correct command is pri set debug {on|off|intense} span x.
The pri intense debug span x command is an alias for
pri set debug 2 span x that didn't get updated when the real
command was changed to pri set debug intense span x.

This will show the help you need:
bas1104*CLI help pri set debug off span

Richard

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Re: [asterisk-users] dahdi-channels.conf vs. chan_dahdi.conf

[Yves A.]

hi,
chan_dahdi is some kind of generic config file, and dahdi-channels the 
config file where you configure your channels... so to say hardware 
specific.
dahdi-channels.conf is normally a generated file which in turn is 
included by chan-dahdi...
it makes sense to me to divide dahdi channel config in two files... but 
as one of them is included by the other you could merge them by hand... but
remember you have then to edit it yourself if your hardware 
configuration changes (e.g. after adding a new card, as it was in your 
case...)
so if your analog card requires drivers, install them or look in you 
/etc/dahdi/modules if you disabled the loading of the module for your newly
added card. after this run dahdi_genconf and all should be set up 
atomagically...

regards,
yves

Am 28.03.2013 14:44, schrieb Ken D'Ambrosio:
Hey, all.  Just added an analog card to our dual-T1 system... and 
clearly I'm doing something wrong.  Less interested in having the 
specifics pointed out than in finding out how/why certain things 
work.  So, really, three things:


* What the bloody Hell is the difference between dahdi-channels.conf 
and chan_dahdi.conf?  (And who thought it was a good idea to have two 
files with, apparently, different functionality, but very similar names?)


* If I'm getting power to my analog phones, but no dial tone, which 
file should I be editing?


* Likewise (and almost certainly related) if dahdi_cfg shows the 
channels, but dahdi show channels only shows my T1 spans, which file 
should I be editing?


Could someone point me to some sample analog configs?  Most of my 
searches have wound me up with GUI folks, and I'm just doing good 
ol-fashioned hand editing on an Ubuntu system.


Thanks!

-Ken





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Re: [asterisk-users] WARNING[28151] from CLI

[Yves A.]

Am 26.03.2013 17:57, schrieb Salaheddine Elharit:

Hello,

 i have all the time this warning i use asterisk 1.4 all works without 
issue i don't have any problem (i can use the inbound and outbound 
calls without issue)


i just want to know what is this WARNING

thanks and regards


 WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels 
available!  Using Primary channel 140 as D-channel anyway!


this can have different causes... mostly a wrong setting in your zaptel 
configuration file... this could be e.g.

mixing american / european settings (e1/t1),
wrong timing settings,
wrong master / source clock setting,
[...]
post more details... what span (e1 or t1), which hardware, driver 
version, asterisk version, config files...



regards,
yves



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Re: [asterisk-users] WARNING[28151] from CLI

[Yves A.]

you have already listed the two config files for using zaptel.
on first sight, they look ok to me (did not use zaptel for years now)
maybe you should definitely comment out any span that is not in use... 
or do the opposite.
i´ve seen this warning several times, but i cant remember it had 
anything to do with spans

being configured but not used.
it always had something to do with timing or even defective cards or 
cabling or even wrong

settings on providers´ site.

what changes were made to the system so that these warnings occur? or 
have they been
visible from the very start? do they affect telefony (e.g. loss of 
calls, one side audio only etc.)?
how much load (concurrent calls) is on the asterisk, does the warning 
occur periodically or

only a few times?
these are all questions you should ask yourself to help you find the 
answer yourself... it can

be very frustrating sometimes, but for me, thats all i can tell about.

regards,
yves

Am 27.03.2013 13:06, schrieb Salaheddine Elharit:
thank you for your help ,but which configure script and when i can 
find this script  ? in etc/asterisk



best regards

2013/3/27 Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com

You do use only span 1 and 6? So the other ports are not plugged?
That is the cause for the warnings. I use a Sangoma E1-Card. The
configure script gives me the option unused for any port. Maybe
your configure script offers you the same option.

Am 27.03.2013 11:54, schrieb Salaheddine Elharit:

Hi

i use 2 digium cards 1 card with 2 ports and the second card with
4 ports

but actually i use just the span 1 and span 6

Asterisk 1.4-r110474M

i use E1 ports


zaptel.conf

# Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013
-- do not hand edit

# Zaptel Configuration File

#

# This file is parsed by the Zaptel Configurator, ztcfg

#

# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED

span=1,1,0,ccs,hdb3

# termtype: te

bchan=1-15,17-31

dchan=16


# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED

span=2,2,0,ccs,hdb3

# termtype: te

bchan=32-46,48-62

dchan=47


# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3

# span=3,3,0,ccs,hdb3

# termtype: te

# bchan=63-77,79-93

# dchan=78


# Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4

# span=4,4,0,ccs,hdb3

# termtype: te

# bchan=94-108,110-124

# dchan=109


# Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1

span=5,5,0,ccs,hdb3

# termtype: te

bchan=125-139,141-155

dchan=140


# Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2

span=6,6,0,ccs,hdb3

# termtype: te

bchan=156-170,172-186

dchan=171


# Global data


loadzone= us

defaultzone= us




etc/asterisk/zapata.conf


[channels]

context=default

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

rxgain=0.0

txgain=0.0


group=1

switchtype=euroisdn

signalling=pri_cpe

callgroup=1

pickupgroup=1

immediate=no

channel = 1-15,17-31


group=2

callgroup=2

switchtype=qsig

signalling=pri_net

callerid=mycallerid

immediate=no

channel = 156-170

channel = 172-176

channel = 125-139

channel = 141-155


thanks and regards



2013/3/27 Yves A. yves...@gmx.de mailto:yves...@gmx.de

Am 26.03.2013 17:57, schrieb Salaheddine Elharit:

Hello,

 i have all the time this warning i use asterisk 1.4 all
works without issue i don't have any problem (i can use the
inbound and outbound calls without issue)

i just want to know what is this WARNING

thanks and regards


 WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No
D-channels available!  Using Primary channel 140 as
D-channel anyway!


this can have different causes... mostly a wrong setting in
your zaptel configuration file... this could be e.g.
mixing american / european settings (e1/t1),
wrong timing settings,
wrong master / source clock setting,
[...]
post more details... what span (e1 or t1), which hardware,
driver version, asterisk version, config files...


regards,






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Re: [asterisk-users] question about zapata.conf

[Yves A.]

it depends a little bit on the driver and asterisk version...
the safest way to become changes applied is to stop asterisk, reload the 
driver and than start asterisk again.


regards,
yves

btw..:
zaptel ist outdated... you should definitely upgrade using dahdi drivers...


Am 25.03.2013 11:44, schrieb Salaheddine Elharit:

hello list,

i have a question related to zapata.conf,if i do any change in 
zapata.conf i must restart asterisk or just i restart zapata ,and how 
to do .


service zaptel restart or there is any other command

Thanks and regards



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Re: [asterisk-users] question about zapata.conf

[Yves A.]

hi,
migrating from zaptel to dahdi HAS an impact... new config files, new 
options and a new channeldriver that has to be
used in your dialplan ... you would have to select the DAHDI channel 
instead of your ZAP channel when dialing...
if you´re to afraid to do it... then leave it as it is and follow the 
ntars-maxime (never touch a running system)...

regards,
yves

Am 25.03.2013 16:15, schrieb Salaheddine Elharit:

thank you so much

fo the upgrade from zptel to dahdi, if there is any possibility to 
upgrade to dahdi without impacting my installation of asterisk and 
other application already installed in my server.


if you can tell how to upgrade using dahdi drivers

thanks and best regards


2013/3/25 Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com

Service asterisk stop
Service zaptel restart
Service asterisk start

-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Salaheddine Elharit
Sent: Monday, March 25, 2013 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question about zapata.conf

i use asterisk 1.4, how i can do to reload dirver

1.service asterisk stop
2 CLI reload chan_zap.so
3 service asterisk start
 that is right or i miss something ?





2013/3/25 Yves A. yves...@gmx.de mailto:yves...@gmx.de


it depends a little bit on the driver and asterisk version...
the safest way to become changes applied is to stop
asterisk, reload the driver and than start asterisk again.

regards,
yves

btw..:
zaptel ist outdated... you should definitely upgrade using
dahdi drivers...


Am 25.03.2013 11:44, schrieb Salaheddine Elharit:


hello list,

i have a question related to zapata.conf,if i do
any change in zapata.conf i must restart asterisk or just i
restart zapata ,and how to do .

service zaptel restart or there is any other command

Thanks and regards





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Re: [asterisk-users] Recording with MixMonitor and AGI

[Yves A.]

hi,

the music heard by MoH is configurable... so if you want silence...
But hold could e.g. also be done by transferring a caller into a 
dynamic meetme room...


yves

Am 14.03.2013 08:43, schrieb Henrik Westerberg:

Hi,

The idea was to record an ongoing call by three party bridging on the 
mobile phone.
Well my problem was to halt execution of the Dialplan so the server 
would not hang up the call. And I don´t want the server to say 
anything during the call.
Now I solved this case as well by using Answer and then Record in the 
dialplan . So I´m not recording with MixMonitor.


But just out of curiosity. How did you mean using hold (in 
answer/hold). Is that MusicOnHold? For me I can´t use that since I 
don´t want to make any noise. Is there another way?


exten = 111,1,Answer()
exten = 111,n,?

I have tried using Wait with a long duration but have not succeeded to 
make it work as I want.


I am using asterisk-java and originate calls to local channels.

Regards,
Henrik


Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de
Datum: söndag 10 mars 2013 11:42
Till: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
mailto:asterisk-users@lists.digium.com, Henrik Westerberg 
henrik.westerb...@ain.se mailto:henrik.westerb...@ain.se

Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI

Hi,

so if your are ok with the way you solved part 1... alright, lets go 
to part 2..

but again... hu.. I don´t understand..
what do you mean with merging to a mobile phone?
do you want do bridge the calls (three partys) or do you want to play 
the just recorded file

from your server-initiated call into a another running call?
what is by hand?
the more explicit you are, the more helpful will be the answer.

you ask but if there is a way to just
dial out and then let the server side of the call Keep the channel up 
but do nothing forever until the call is hang up


of course you can...you could e.g.:
call into a queue
call into a meetme room
call with the help of a local channel into a context where you do 
nothing but answer / hold


but as i said i did not quite catch what your objective really 
is... i just dont understand

your scenario or cant imagine its sense.

if you are a java programmer, i think your using the asterisk-java lib 
from s. reuter..
if so, you have any freedom, you could also use ami connection to 
listen to events

to start and stop recordings and so on.

regards,
yves

Am 09.03.2013 21:32, schrieb Henrik Westerberg:

Hi,

Thanks for your answer!

1.
 so you want to establish a call (triggered by ami) between two partys, record the 
conversation

 and save the file to a(nother) server (afterwards), right?

Yes this is correct, and I prefer to do the transferring of the file 
to another server with my existing AGI.

My AGIs are written in java. Today I the upload is done over http.
Today I schedule the upload in the AGI script a couple of seconds 
after the channel is hang up. But the two

lines might not be hung up at the same time.
Your suggestion of always fixing the file is wise, it now seems to 
work fine after having been processed with sox.


So now I think that this case 1 is ok for me :-)

2.
 and another task is to establish (also ami triggered) a call to a mobile and play, 
lets say a voicefile.
 this conversation should also be recorded and saved on a(nother) 
server (afterwards), right?


The idea is to perform a probe call with the only task of recording 
what the other party says.
It will be merged by hand on a mobile phone to an ongoing call with 
another party.
This could be done by calling out and letting AGI execute a RECORD 
FILE but if there is a way to just
dial out and then let the server side of the call Keep the channel 
up but do nothing forever until the call is hang up
Then I could easily use the MixMonitor and write the whole 
conversation in the dialplan with uploading similar to the first case.

Any suggestions?

Regards,
Henrik



Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
mailto:asterisk-users@lists.digium.com

Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
mailto:asterisk-users@lists.digium.com

Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI

hi,

hard to understand, what your objective is... at least for me ;-)

so you want to establish a call (triggered by ami) between two 
partys, record the conversation

and save the file to a(nother) server (afterwards), right?

and another task is to establish (also ami triggered) a call to a 
mobile and play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) 
server (afterwards), right?


let me know, if i understood you right, the solution is not so hard 
to implement.
In what language do you preferrably write your AGIs? (although

Re: [asterisk-users] Recording with MixMonitor and AGI

[Yves A.]

Hi,

so if your are ok with the way you solved part 1... alright, lets go to 
part 2..

but again... hu.. I don´t understand..
what do you mean with merging to a mobile phone?
do you want do bridge the calls (three partys) or do you want to play 
the just recorded file

from your server-initiated call into a another running call?
what is by hand?
the more explicit you are, the more helpful will be the answer.

you ask but if there is a way to just
dial out and then let the server side of the call Keep the channel up 
but do nothing forever until the call is hang up


of course you can...you could e.g.:
call into a queue
call into a meetme room
call with the help of a local channel into a context where you do 
nothing but answer / hold


but as i said i did not quite catch what your objective really is... 
i just dont understand

your scenario or cant imagine its sense.

if you are a java programmer, i think your using the asterisk-java lib 
from s. reuter..
if so, you have any freedom, you could also use ami connection to listen 
to events

to start and stop recordings and so on.

regards,
yves

Am 09.03.2013 21:32, schrieb Henrik Westerberg:

Hi,

Thanks for your answer!

1.
 so you want to establish a call (triggered by ami) between two partys, record 
the conversation
 and save the file to a(nother) server (afterwards), right?

Yes this is correct, and I prefer to do the transferring of the file 
to another server with my existing AGI.

My AGIs are written in java. Today I the upload is done over http.
Today I schedule the upload in the AGI script a couple of seconds 
after the channel is hang up. But the two

lines might not be hung up at the same time.
Your suggestion of always fixing the file is wise, it now seems to 
work fine after having been processed with sox.


So now I think that this case 1 is ok for me :-)

2.
 and another task is to establish (also ami triggered) a call to a mobile and play, lets say 
a voicefile.
 this conversation should also be recorded and saved on a(nother) 
server (afterwards), right?


The idea is to perform a probe call with the only task of recording 
what the other party says.
It will be merged by hand on a mobile phone to an ongoing call with 
another party.
This could be done by calling out and letting AGI execute a RECORD 
FILE but if there is a way to just
dial out and then let the server side of the call Keep the channel up 
but do nothing forever until the call is hang up
Then I could easily use the MixMonitor and write the whole 
conversation in the dialplan with uploading similar to the first case.

Any suggestions?

Regards,
Henrik



Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com

Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com

Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI

hi,

hard to understand, what your objective is... at least for me ;-)

so you want to establish a call (triggered by ami) between two partys, 
record the conversation

and save the file to a(nother) server (afterwards), right?

and another task is to establish (also ami triggered) a call to a 
mobile and play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) 
server (afterwards), right?


let me know, if i understood you right, the solution is not so hard to 
implement.
In what language do you preferrably write your AGIs? (although there 
is no absolute need for using an

agi... you can all write down in your dialplan...)
is there a special protocol requirement for saving/transferring the 
recorded voicefile (e.g. ftps)?
One obstacle is, that the recorded file is not fully written 
_immediately_ after stopmixmonitor or hangup...
this has to be taken care of and depending on your agi... it might be 
interrupted, if the call is hungup...

but as you did not show your agi... these are just hints..

regards,
yves



Am 07.03.2013 16:21, schrieb Henrik Westerberg:

Hi,

I am developing a call recording application on Asterisk 11.2 and 
have this configuration in my dialplan:


[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)

[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)

[outgoing-originate-rec]
exten = 
h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})


exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is 
${CC_CALLID}, CC_FILENAME is ${CC_FILENAME})

exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)

If I want to make a recorded server callout from 0 
to 08 I then originate a call via AMI to 
Local/0@outgoing-originate with context set 
to outgoing-originate-rec and extension to 08.

The result will be something like

Re: [asterisk-users] Sending SMS from asterisk

[Yves A.]

Hi there,

for sending SMSes I am using a 3G Modem and SMSlib... it is not bound to 
asterisk in any way, but I always wanted to integrate this possibility
some day. I could not do so, because our landline provider (Vodafone) 
does not support it via E1 PRI lines... by I thought (never tried...) if 
it would
be possible to use the SMS ServiceNumber from my mobile Provider...? I 
have a valid mobile contract, the number of the SMScc ,
my Cardnumber (t-mobile), my phonenumber and so on... so it should be 
possible, I think... but how? Has anybody a clue?


regards,
yves

Am 09.03.2013 11:03, schrieb Miguel Oyarzo:


Hi Bilal,

It's not necessary to use a FXS port, you can compile  install 
chan_dongle and buy a Huawei 3G dongle.
We have running here a SMS solution with four 3G dongles, which sends 
over 20.000 SMS a month.


In addition, I wrote an script able to send up to 12000 characters in 
concatenated SMS (the recipient receives a single SMS only)


chan_dongle works very well.

--
==
Miguel Oyarzo
Senior [ Network | Systems Design ] Engineer
http://www.linkedin.com/in/mikeaustralia
Linux User: # 483188 - counter.li.org
Melbourne, Australia


On 3/9/2013 1:09 PM, Gerardo Barajas wrote:

Yes, you can check solutions from sangoma and khomp.

Saludos/Regards
--
Ing. Gerardo Barajas Puente

Proyectos Especiales/Preventa | www.neocenter.com 
http://www.neocenter.com

T:+52 (55)  8590-9000 x 7003


On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com 
mailto:bilmar...@yahoo.com wrote:


Hi;

If my landline service provider does not provide the ability to
send the SMS, and I need to send SMS from asterisk, then what is
the required? How?

Is it possible to send SMS from asterisk using SIM card to be
connected via GSM adaptor connected to FXS ports? Or HOW?

From the other side, this is existed only in asterisk 1.8 or it
is existed in asterisk 1.4?

Regards
Bilal

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Re: [asterisk-users] Recording with MixMonitor and AGI

[Yves A.]

hi,

hard to understand, what your objective is... at least for me ;-)

so you want to establish a call (triggered by ami) between two partys, 
record the conversation

and save the file to a(nother) server (afterwards), right?

and another task is to establish (also ami triggered) a call to a mobile 
and play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) 
server (afterwards), right?


let me know, if i understood you right, the solution is not so hard to 
implement.
In what language do you preferrably write your AGIs? (although there is 
no absolute need for using an

agi... you can all write down in your dialplan...)
is there a special protocol requirement for saving/transferring the 
recorded voicefile (e.g. ftps)?
One obstacle is, that the recorded file is not fully written 
_immediately_ after stopmixmonitor or hangup...
this has to be taken care of and depending on your agi... it might be 
interrupted, if the call is hungup...

but as you did not show your agi... these are just hints..

regards,
yves



Am 07.03.2013 16:21, schrieb Henrik Westerberg:

Hi,

I am developing a call recording application on Asterisk 11.2 and have 
this configuration in my dialplan:


[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)

[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)

[outgoing-originate-rec]
exten = 
h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})


exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is 
${CC_CALLID}, CC_FILENAME is ${CC_FILENAME})

exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)

If I want to make a recorded server callout from 0 
to 08 I then originate a call via AMI to 
Local/0@outgoing-originate with context set 
to outgoing-originate-rec and extension to 08.

The result will be something like this:

-- Executing [s@macro-ccdev2-rec:1] 
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack

  == Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] 
AGI(SIP/upps-ccm-tq01-003e, 
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script 
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, 
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1] 
AGI(SIP/upps-ccm-tq01-003f, 
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script 
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, 
returning 0

  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/upps-ccm-tq01-003f

Unfortunately I get two different calls to the h extension, but this I 
can cope with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to 
upload the file. The file will not have a duration. It works when I 
schedule the uploading a while after from my agi application but I 
would rather not rely on a timeout.


When I tried to run StopMixMonitor before the Agi call in the h 
extension, the first call fail and I never get any uploading with callid.


-- Executing [s@macro-ccdev2-rec:1] 
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack

  == Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1] 
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
  == Spawn extension (outgoing-originate-rec-dev2, h, 1) exited 
non-zero on 'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1] 
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack

  == MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2] 
AGI(SIP/upps-ccm-tq01-0043, 
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack


Am I missing something here? I also looked at the possibility to 
specify a command to execute when MixMonitor stops but I would rather 
handle the file uploading in my agi application.


I also have another case: I want to dial out a call and record it. It 
will be a oneway-call from the server to a mobile. Do I need to get 
AGI-control of it and record with an AGI command or how can I hack it 
directly in the dial plan using MixMonitor?


Best Regards,
Henrik


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Re: [asterisk-users] Extension cant pickup calls but can transfer.

[Yves A.]

do you have only ONE phone, that can´t pickup, or is this a general problem?
is pickup configured (feature.conf) AND enabled ?

regards,
yves


Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:

Greetings.

I got an extension on my Elastix who cannot pick calls on the other 
extensions, but It can transfer his calls to the other extensions. 
When this extension tries to pickup a call pressing *8  it simply does 
not pick it up. Transfering calls works just fine so dtmf may be not 
the problem.


Where should I look?

Any further information needed just ask.

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Re: [asterisk-users] Extension cant pickup calls but can transfer.

[Yves A.]

is it the same type and make of phone than one of the working ones?
- compare (dtmf) settings, firmware release etc.

- check call-group and pickup group... is the non working extension 
configured there?


regards,
yves

Am 07.03.2013 20:28, schrieb Luis H. Forchesatto:

Its only ONE phone who doesnt pickup calls.

2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de

do you have only ONE phone, that can´t pickup, or is this a
general problem?
is pickup configured (feature.conf) AND enabled ?

regards,
yves


Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:

Greetings.

I got an extension on my Elastix who cannot pick calls on the
other extensions, but It can transfer his calls to the other
extensions. When this extension tries to pickup a call pressing
*8  it simply does not pick it up. Transfering calls works just
fine so dtmf may be not the problem.

Where should I look?

Any further information needed just ask.

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Re: [asterisk-users] Extension cant pickup calls but can transfer.

[Yves A.]

mmh... should work... (i think you checked double and applied any 
changes, right..?
sometimes deleting the extension and configuring a new one can fulfil 
wonders...)


I have no further tip... maybe elastix support or forum can help... if 
you are familiar with

cli output and sip debugging... check cli output and sip debug output...

good luck.
yves

Am 07.03.2013 20:38, schrieb Luis H. Forchesatto:
Yes, both are configured in the same ata (linksys pap2) and the 
configuration options are the same. Call group and pick group are the 
same for both too.


2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de

is it the same type and make of phone than one of the working ones?
- compare (dtmf) settings, firmware release etc.

- check call-group and pickup group... is the non working
extension configured there?

regards,
yves

Am 07.03.2013 20:28, schrieb Luis H. Forchesatto:

Its only ONE phone who doesnt pickup calls.

2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de

do you have only ONE phone, that can´t pickup, or is this a
general problem?
is pickup configured (feature.conf) AND enabled ?

regards,
yves


Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:

Greetings.

I got an extension on my Elastix who cannot pick calls on
the other extensions, but It can transfer his calls to the
other extensions. When this extension tries to pickup a call
pressing *8  it simply does not pick it up. Transfering
calls works just fine so dtmf may be not the problem.

Where should I look?

Any further information needed just ask.

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*



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Re: [asterisk-users] red alarm on span - do channels in the group automatically get skipped over?

[Yves A.]

hi,

yes, this is the way, asterisk / the channeldriver handles it.
you can simulate the failure of one span by just pulling out the cable 
and see what happens..

on top, you can influence the order, the channels are used by using
dahdi/g1 or dahdi/G1...
regards,
yves

Am 05.03.2013 07:31, schrieb Hose:

Hello,

If I put two spans' worth of channels, say 1-23 from span 1 and 25-47 in
span 2, in one group, but only span 2 was showing OK and the other was
down / showing a RED alarm, would asterisk automatically skip over
trying to use channels 1-23 when doing outbound calls? e.g.,
dial(dahdi/g1/(number) would just jump to channel 25?

Testing seems to bear this out, but I'm not positive about it.

hose

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Re: [asterisk-users] ODBC and SQLIte3

[Yves A.]

hi,

if you use realtime peers, and you want to see their states, you have to 
look in the database...
if you want to see their states via cli, you have to set 
rtcachefriends=yes in your sip.conf...

there are other settings that you might be interested in...  :

rtcachefriends=yes ; Cache realtime friends by adding them to the 
internal list
; just like friends added from the 
config file only on a

; as-needed basis? (yes|no)

rtsavesysname=yes  ; Save systemname in realtime database at 
registration

; Default= no

rtupdate=yes   ; Send registry updates to database using 
realtime? (yes|no)
; If set to yes, when a SIP UA 
registers successfully, the ip address,
; the origination port, the 
registration period, and the username of
; the UA will be set to database via 
realtime.
; If not present, defaults to 'yes'. 
Note: realtime peers will
; probably not function across reloads 
in the way that you expect, if

; you turn this option off.
rtautoclear=yes; Auto-Expire friends created on the fly 
on the same schedule
; as if it had just registered? 
(yes|no|seconds)
; If set to yes, when the registration 
expires, the friend will
; vanish from the configuration until 
requested again. If set
; to an integer, friends expire within 
this number of seconds

; instead of the registration interval.

ignoreregexpire=yes; Enabling this setting has two functions:
;
; For non-realtime peers, when their 
registration expires, the
; information will _not_ be removed 
from memory or the Asterisk database
; if you attempt to place a call to the 
peer, the existing information
; will be used in spite of it having 
expired

;
; For realtime peers, when the peer is 
retrieved from realtime storage,
; the registration information will be 
used regardless of whether
; it has expired or not; if it expires 
while the realtime peer
; is still in memory (due to caching or 
other reasons), the
; information will not be removed from 
realtime storage


regards,
yves


Am 17.02.2013 12:51, schrieb termo termosel:

Hi,

I had configured Asterisk to use default database  located in 
/var/lib/asterisk/sqlite3dir/sqlite3.db. When I put odbc show in 
Asterisk's cli, It returns me that I have conected but when I put sip 
show peers,Asterisk doesn't found any peer or user.


ubuntu*CLI odbc show

ODBC DSN Settings
-

  Name:   asterisk
  DSN:asterisk-connector
Last connection attempt: 1970-01-01 01:00:00
  Pooled: No
  Connected: Yes

ubuntu*CLI sip show peers
Name/username HostDyn Forcerport 
ACL Port Status  Description  Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 
offline]



This mi configuration,

/etc/odbci.ini

[asterisk-connector]
Description = SQLite3 database
Driver  = SQLite3
Database= /var/lib/asterisk/sqlite3dir/sqlite3.db

/etc/odbcinst.ini

[SQLite3]
Description= SQLite3 ODBC Driver
Driver=/usr/local/lib/libsqlite3odbc.so
Setup=/usr/local/lib/libsqlite3odbc.so
Threading=2

/etc/asterisk/extconfig.conf

[settings]

sipusers = odbc,asterisk,sip_buddies
sippeers = odbc,asterisk,sip_buddies
sipregs = odbc,asterisk,sip_buddies

/etc/asterisk/func_odbc.conf

[SQL]
dsn=asterisk
readsql=${ARG1}

/etc/asterisk/modules.conf

autoload=yes
;preload = res_odbc.so
;preload = res_config_odbc.so
noload = pbx_gtkconsole.so
;load = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
noload = chan_capi.so
load = res_musiconhold.so
noload = chan_alsa.so
;noload = chan_oss.so
noload = cdr_sqlite.so
noload = app_directory_odbc.so
;noload = res_config_odbc.so
;noload = res_config_pgsql.so

/etc/asterisk/res_odbc.conf

[asterisk]
enabled = yes
dsn = asterisk-connector
pre-connect = yes


Can someone help me?

Thanks,
Jordi


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Re: [asterisk-users] ODBC and SQLIte3

[Yves A.]

looks like a mistake in your extconfig.conf...
do you want to use realtime extensions too?

for further instructions show us your extensions.conf and the verbose 
output of the cli showing the dialattempt...


regards,
yves

Am 17.02.2013 14:31, schrieb termo termosel:

Hi,

I have add this options into Sip.conf but the CLI continues telling 
the same message:


ubuntu*CLI sip show peers
Name/username Host Dyn Forcerport ACL Port Status Description Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 
offline]


I have two users in my slite3.db. but Asterisk doesn't show me. It is 
how asterisk can't access into this database.


When I go to call, Asterisk tells me that extension xxx is not found 
in phones context.


Thanks,
Jordi

Date: Sun, 17 Feb 2013 13:00:44 +0100
From: yves...@gmx.de
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ODBC and SQLIte3

hi,

if you use realtime peers, and you want to see their states, you have 
to look in the database...
if you want to see their states via cli, you have to set 
rtcachefriends=yes in your sip.conf...

there are other settings that you might be interested in... :

rtcachefriends=yes ; Cache realtime friends by adding them to the 
internal list
; just like friends added from the 
config file only on a

; as-needed basis? (yes|no)

rtsavesysname=yes  ; Save systemname in realtime database 
at registration

; Default= no

rtupdate=yes   ; Send registry updates to database 
using realtime? (yes|no)
; If set to yes, when a SIP UA 
registers successfully, the ip address,
; the origination port, the 
registration period, and the username of
; the UA will be set to database via 
realtime.
; If not present, defaults to 'yes'. 
Note: realtime peers will
; probably not function across reloads 
in the way that you expect, if

; you turn this option off.
rtautoclear=yes; Auto-Expire friends created on the 
fly on the same schedule
; as if it had just registered? 
(yes|no|seconds)
; If set to yes, when the registration 
expires, the friend will
; vanish from the configuration until 
requested again. If set
; to an integer, friends expire within 
this number of seconds

; instead of the registration interval.

ignoreregexpire=yes; Enabling this setting has two functions:
;
; For non-realtime peers, when their 
registration expires, the
; information will _not_ be removed 
from memory or the Asterisk database
; if you attempt to place a call to 
the peer, the existing information
; will be used in spite of it having 
expired

;
; For realtime peers, when the peer is 
retrieved from realtime storage,
; the registration information will be 
used regardless of whether
; it has expired or not; if it expires 
while the realtime peer
; is still in memory (due to caching 
or other reasons), the
; information will not be removed from 
realtime storage


regards,
yves


Am 17.02.2013 12:51, schrieb termo termosel:

Hi,

I had configured Asterisk to use default database  located in
/var/lib/asterisk/sqlite3dir/sqlite3.db. When I put odbc show in
Asterisk's cli, It returns me that I have conected but when I put
sip show peers,Asterisk doesn't found any peer or user.

ubuntu*CLI odbc show

ODBC DSN Settings
-

  Name:   asterisk
  DSN:asterisk-connector
Last connection attempt: 1970-01-01 01:00:00
  Pooled: No
  Connected: Yes

ubuntu*CLI sip show peers
Name/username HostDyn
Forcerport ACL Port Status  Description Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online,
0 offline]


This mi configuration,

/etc/odbci.ini

[asterisk-connector]
Description = SQLite3 database
Driver  = SQLite3
Database= /var/lib/asterisk/sqlite3dir/sqlite3.db

/etc/odbcinst.ini

[SQLite3]
Description= SQLite3 ODBC Driver
Driver=/usr/local/lib/libsqlite3odbc.so
Setup=/usr/local/lib/libsqlite3odbc.so
Threading=2

Re: [asterisk-users] cisco 7940 and asterisk 11

[Yves A.]

maybe not related to your problem, but I had a similar effect after 
upgrading my 1.6 to 11.
not on phones (pbx was used as ivr only), but voicefile were played in a 
robo style unconditionally.

this effect could only be gotten rid of by rebooting the server.
i had to completely clean the installation and rebuild 11 from 
scratch... i think some mp3
classes have caused the effect... (although no mp3 files were used..) 
when the effect occurred,
there was nothing that hinted to  any problem... cpu usage, network etc. 
everything was fine,
but even only restarting asterisk did not help... i think some libs got 
messed up during update.


so i´d recommend to rebuild the server complete from scratch... if the 
problem still
exists after new build... it would be interesting to know, if you just 
took the same config-files

from your previous version which would maybe cause problems...

if all fails, i would then go deeper into network analysis and trace the 
traffic.


meanwhile i administer around 10 asterisk boxes and i always use ubuntu 
12.04 lts and
latest asterisk 11 on dell r3/4/6xx servers... up to now everything runs 
fine..


regards,
yves


Am 14.02.2013 07:20, schrieb Julian Lyndon-Smith:

very polite *bump*

this is a real issue for us - anyone got _any_ clues or ideas ?

Thanks ;)


On 12 February 2013 14:29, Julian Lyndon-Smith aster...@dotr.com 
mailto:aster...@dotr.com wrote:


Ever since we upgraded to asterisk 11 we have had audio problems with
our cisco 7940 phones.

The problems manifest themselves by the conversation turning robotic
or into silence (to the extent our agents are saying hello? hello?
and the customer is saying I hear you just fine

We had to change pedantic=no in sip.conf to allow the phones to
register

We are assuming that it is the phone=asterisk combination because

a) the call recordings of the conversation are perfect (no noise on
the line, conversation is clear) but it is apparent that the agent
cannot hear the customer sometimes (Hello?)

b) we have replaced the cables and switches between the phones and
the pbx

c) we don't have the same problem with Aastra 9133i or Polycom 331
phones

Are there any settings in sip.conf that may help this , or a
particular firmware ? Are there any known audio problems with cisco
7940 and asterisk 11 ?

Many thanks

Julian

--
Julian Lyndon-Smith
IT Director, Dot R Limited

I don't care if it works on your machine!  We are not shipping
your machine!

The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg




--
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IT Director, Dot R Limited

I don't care if it works on your machine!  We are not shipping your 
machine!


The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg


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Re: [asterisk-users] Variables set by AGI lost in dialplan

[Yves A.]

are the calls always handled in the same manner, or sometimes different, 
e.g.
with conferencing, bridging, transferring, using local channels and and 
and...?
in that case try to use two underscores before the variablename to use 
inheritance.


maybe you accidentially swap the channels? you could try to set the 
variables with

help of the shared function to set it for both channels...

regards,
yves

Am 14.02.2013 10:40, schrieb Deepesh D:

Hello,

I am using asterisk 1.8.17.0 with a fast agi written in C

The following is a part of my dialplan

exten = _X.,n,MSet(my_var=0,my_var1=0)
exten = _X.,n,AGI
;; Call to a fast agi to set values of my_var my_var1
exten = _X.,n,Log(NOTICE,${my_var} ${my_var1}) ;; log the values to
asterisk messages

Inside the AGI I do some calculations and set the values of my_var and
my_var1 variables like
SET VARIABLE my_var 0.008
SET VARIABLE my_var1 0.009

The problem I am facing is that sometimes the variables are wrongly
received as 0 (zero) in the dialplan even if the AGI has set it to a
non-zero value. Inside the AGI I am logging the values of variables to
a log file, and the log file always shows non-zero values. But in my
asterisk messages file the values are zero for some calls.

This error does not happen for all calls and is not reproducible, it
is random. My asterisk server handles about 100 calls per minute, so
its impossible for me to do an 'agi set debug' and observer the output

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[asterisk-users] Asterisk Realtime Extension... strange behaviour

[Yves A.]

Hi,

I encountered a strange behaviour using realtime extensions... (on 
Asterisk 11.2)


when I use the following static dialplan, everything works as expected..:

[from-sip]
exten =  110,1,Dial(DAHDI/g0/${EXTEN})
exten =  112,1,Dial(DAHDI/g0/${EXTEN})
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _X.,1,Dial(DAHDI/g0/${EXTEN})

will say... if a sip phone calls 110 or 112 the call is routed into 
PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet 
to the corresponding SIP user
and if a sip phone calls any other number the call should be routed into 
PSTN... thats ok and works as expected.


when I change to realtime:
[from-sip]
switch = Realtime

and put the diaplan into the database
idcontextextenpriorityappappdata
1from-sip1101DialDAHDI/g0/${EXTEN}
2from-sip1121DialDAHDI/g0/${EXTEN}
3from-sip_XXX1DialSIP/${EXTEN}
4from-sip_X.1DialDAHDI/g0/${EXTEN}

only the emergency calls work and any other call goes to DAHDI... I cant 
reach any other SIP phone.

Even when swapping the content of the rows 3 and 4 in the database to
idcontextextenpriorityappappdata
1from-sip1101DialDAHDI/g0/${EXTEN}
2from-sip1121DialDAHDI/g0/${EXTEN}
3from-sip_X.1DialDAHDI/g0/${EXTEN}
4from-sip_XXX1DialSIP/${EXTEN}

makes no difference...
I thought, using realtime extensions would read the dialplan from top to 
bottom, ordered by id... but it
seems to be ignored somehow and the extension _X. catches the calls 
before the extensionpattern _XXX is reached.


I _could_ avoid this be prefixing external numbers with a leading 0 
for example... but I dont want to... as I said.. using

static extension via extensions.conf the dialplan works as expected...

Am I missing something?

regards,
yves



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Re: [asterisk-users] About Asterisk with Digium TE405P PRI ISDN cad

[Yves A.]

Hi,

I think, you mean connecting the two boxes directly with a cable... not 
via PSTN, right?


1.) You need a special cross-over cable to connect one Port directly to 
another Port...
(if you want to crimp it yourself, you can find the associated Pins via 
Google... ethernet crossover

cables do not work as they have different links)
2.) configure one end as master (CPN) and the other asterisk as Network 
(CPN), otherwise

you´ll get timing issues...

thats all...

regards,
yves

Am 11.02.2013 14:00, schrieb Shitian Long:

Hello,

I am trying to connect two asterisks with PRI connection. One asterisk 
has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port 
card.


I am wondering if there would be some step by step guide that I could 
follow to to this kind of connection?


Thanks



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Re: [asterisk-users] access control softphone registration through asterisk

[Yves A.]

Hi,

are you using realtime extensions or the classic config-file 
extension.conf ?
One way to go yould be to implement the allowed / not allowed logic in 
the context of your sip users.
check their permissions and if they are allowed to call... continue 
with the dialplan, if not, route them
to a voiceprompt saying that the call is prohibited due to whatever 
reasons...


To do so, take a look at the dialplan functions if and db. Of course 
you somehow have to set a
flag in asterisk, that decides about permissions... Don´t know which 
way you will programmatically
set or clear this flag... there are hundreds of possibilites... the 
easiest way I think would be to use the

asterisk build-in database (therefore the hint to the function db...)

regards,
yves

Am 08.02.2013 22:18, schrieb Muhammad:

Hi,
I wana control my SIP register from asterisk.
I other hand, when users login into their softphone, dont access to 
call and when I give them access, they can call.


I dont know it's right way to plan my scenario/?


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Re: [asterisk-users] special conference room

[Yves A.]

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many 
switches and options and
can help me a lot... but as you already said... does _almost_ all 
features.. unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not tried 
it in deep, because just

from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the mute switch to disable the callers to talk to 
the conference.. (so to say

it mutes the callers microphone, not his earphones am I wrong?
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:


Please study meetme application's options. You will get almost all 
feature you ask for in it


On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de 
mailto:yves...@gmx.de wrote:


Hi list,

I am in need of a special asterisk conference room with the
following constraints:

- there is one admin / moderator and several normal callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a
specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] special conference room

[Yves A.]

ok,

now i have got some very valuable information to start off with. thank 
you all.

i´ll be back to report success or further questions...

just one thing, that i think might be a showstopper that i may have not
explained clear enough...:
muting and unmuting a caller should have the effect, that the caller can 
talk

to the moderator or not... any caller should NEVER hear what other callers
are talking... may he be muted or not...

yves

Am 16.01.2013 23:01, schrieb Danny Nicholas:


From what I read, neither confbridge or meetme have the whisper 
feature built-in;  This doesn't matter because the moderator would 
have to use meetmeadmin or the confbridge equivalent to control the 
other functions.  The moderator would either need two phones or a 
phone and a web interface.  Let's say Yves' special conference is 
.  The moderator would start using this command


Exten = s,1,meetme()

The participants would do

Exten = s,1,meetme(,m) -- muted so they can listen but not talk

- there is one admin / moderator and several normal callers.
- the callers must not hear any other caller, only the moderator

The moderator would need to be able to enumerate the conference by doing

Asterisk --rx core show channels verbose|grep meetme

This is supposed to be doable from the dialplan but my google-fu 
failed me on it.

- the moderator must be able to mute and unmute any caller at any time

Establish a maximum number of users and set this up for each one

Exten = 99,1,meetmeadmin(,M,1) let user 1 talk

Exten = 199,1,meetmeadmin(,m,1) turn user 1 back off
- the moderator must be able to talk to all callers or to a specific 
caller.


Exten = 901,1,chanspy(SIP/XXX,w)
- the modetator must be able to kick off any caller at any time...

Exten = 299,1,meetmeadmin(,k,1) kick out user 1

Exten = 666,1,meetmeadmin(,K) shut it down

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

*Sent:* Wednesday, January 16, 2013 3:34 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] special conference room

Sounds like a conference with all attendees permanently muted  (except 
the moderator).


The moderator uses whisper to communicate with individuals.

--Don

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

*Sent:* Wednesday, January 16, 2013 3:11 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] special conference room

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many 
switches and options and
can help me a lot... but as you already said... does _almost_ all 
features.. unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not 
tried it in deep, because just

from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the mute switch to disable the callers to talk to 
the conference.. (so to say

it mutes the callers microphone, not his earphones am I wrong?
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:

Please study meetme application's options. You will get almost all
feature you ask for in it

On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de
mailto:yves...@gmx.de wrote:

Hi list,

I am in need of a special asterisk conference room with the
following constraints:

- there is one admin / moderator and several normal callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a
specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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[asterisk-users] special conference room

[Yves A.]

Hi list,

I am in need of a special asterisk conference room with the following 
constraints:


- there is one admin / moderator and several normal callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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[asterisk-users] Asterisk console suddenly extremely verbose...

[Jean-Yves Avenard]

Hi there.

I started the console today to reload the extensions.conf file ; only
to be greeted with extremely verbose console.
Seems related to the zaptel card:

Example:
 Supervisory frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
 Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
 N(R): 020 P/F: 1
 0 bytes of data
voip*CLI
 [ 00 01 01 2f ]
voip*CLI
 Supervisory frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
 Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
 N(R): 023 P/F: 1
 0 bytes of data
Handling message for SAPI/TEI=0/0
-- ACKing all packets from 22 to (but not including) 23
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 timer


This is repeating every 10s or so...

Any ideas what this message means and is there a way to prevent it
from happening.
No changes has been made on this asterisk box in years (running old
1.4.25 if it ain't broken version)

Thanks in advance
JY

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Re: [asterisk-users] Asterisk console suddenly extremely verbose...

[Jean-Yves Avenard]

Hi

On 16 December 2011 13:24, Richard Mudgett rmudg...@digium.com wrote:

 You have pri intense debug span x enabled.
 Disable with pri no debug span x.

Thanks...

I couldn't find any configuration file showing this ; but ran the
command in the CLI... Seems to have done it.

I really wonder how it could have been turned on ...

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[asterisk-users] Redirecting a Channel more than three times...

[Yves A.]

  Hi folks,

could someone please try to confirm the following (mis)behaviour of my 
asterisk?

Imagine the following scenario:

Caller A calls the central.
Central picks up, talks to Caller A which wants to be connected to 
employee X.
Central puts Caller A on hold by Redirecting the Channel to a Queue.
Central calls emplyee X and bridges both channels... everybody is happy.

But..:

Caller A calls the central.
Central picks up, talks to Caller A which wants to be connected to 
employee X.
Central puts Caller A on hold by Redirecting the Channel to a Queue.
Central calls emplyee X and X doesn´t want to talk with Caller A 
Central and employee hang up..
Central pulls Caller A back from Queue (again, with Redirecting the 
channel to its own extension)
Caller A now want to talk with employee Y and so on

This game works exactly three times... when the central wants to pull 
back the Caller from the
Queue for the third time, the call is hungup.

I searched and searched, but could not find anything about a 
redirect-limit or so...
what, if there is no such limit, am I doing wrong?

If there is such a limit.. where is it configured?

thank you anyways,
yves


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Re: [asterisk-users] Redirecting a Channel more than three times...

[Yves A.]

  Hi Danny,

I decided against Parking Calls, because it seemed quite complicated and 
useless
for me... as far as i remember, parkedcalls return automagically after a 
timeout which was not desirable.

I would have to rewrite a lot of code, if i have to change... but there 
must be a reason for this misbehaviour,
and i think its hardcoded in the asterisk-source.

somewhere seems to be a counter that counts the redirects... it maybe 
useful in some
case, maybe to avoid loops or something similar to bounces in emails, 
but in my case its
undesired...

because i am using trixbox / freepbx the dialplan is very complicated, 
but it showed me no hint
of beeing responsible for this... the cli-output gives no hint.

yves


Am 24.09.2010 15:10, schrieb Danny Nicholas:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
 Sent: Friday, September 24, 2010 6:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Redirecting a Channel more than three times...

Hi folks,

 could someone please try to confirm the following (mis)behaviour of my
 asterisk?

 Imagine the following scenario:

 Caller A calls the central.
 Central picks up, talks to Caller A which wants to be connected to
 employee X.
 Central puts Caller A on hold by Redirecting the Channel to a Queue.
 Central calls emplyee X and bridges both channels... everybody is happy.

 But..:

 Caller A calls the central.
 Central picks up, talks to Caller A which wants to be connected to
 employee X.
 Central puts Caller A on hold by Redirecting the Channel to a Queue.
 Central calls emplyee X and X doesn´t want to talk with Caller A
 Central and employee hang up..
 Central pulls Caller A back from Queue (again, with Redirecting the
 channel to its own extension)
 Caller A now want to talk with employee Y and so on

 This game works exactly three times... when the central wants to pull
 back the Caller from the
 Queue for the third time, the call is hungup.

 I searched and searched, but could not find anything about a
 redirect-limit or so...
 what, if there is no such limit, am I doing wrong?

 If there is such a limit.. where is it configured?

 thank you anyways,
 yves

 #1.  Have you looked at the CLI output for this scenario
 #2.  Why don't you use Parking instead of queue?




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Re: [asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio

[Yves Arikoglu]

thanks brian,

yes, i am aware that sip is only responsible for signalling and therefor 
my conclusion was, that it
has got something to do with nat / firewall / the router...
meanwhile i´ve got it solved... although the sip-provider tried to 
convince me, that the misconfiguration
is on my asterisks´ side, i penetrated the support until they looked 
over it again and... what should i
say... finally they had to admit, that the router had a wrong acesslist. 
they corrected it and now it works.

yves

Brian schrieb:
 On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote:
   
 Hi,

 I am breaking my fingers in configuring an asterisk (1.6) to 
 successfully transmit audio with the following setup:

 asterisk, resides in local network, ip is 10.26.208.252
 versatel business router (directly connected to a dsl, configured by 
 sip-provider), WAN ip 89.244.13.25
 versatel sip-proxy ip 89.244.13.10


 in sip.conf I have:
 [general]
 bindaddr=0.0.0.0
 externip=89.244.13.25
 localnet=10.26.208.0/255.255.252.0
 nat=yes
 qualify=yes


 the local sip phones register correctly and can make calls between each 
 other with audio.
 the local sip phones CAN make outbound calls via the sip-provider... 
 will say, destination phone rings, but there is no audio (on both legs)
 after pickup...
 external phones can call my sip-number... the call comes into the 
 asterisk, the sip-extension rings, but after pickup... no audio at all.
 even if i route the call from external to a queue or something else... i 
 see, that asterisk is playing voicefiles, but the caller does not hear
 anything.
 because sip-signalling works in any ways, but audio not, i think its got 
 something to do with nat... but there is no firewall between asterisk
 and the router or between the router and the internetconnection from 
 versatel... and i already tried millions of combinations of using
 nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m 
 stuck as i was never ever stuck before :-(

 any hints? anybody?

 
 You are aware that SIP only sets up, monitors and takes the call down?
 The audio stream is RDP and on higher ports. My guess is that the audio
 stream on inbound calls is not arriving where it should be - or is
 blocked. This could be router or nat, but one thing jumps out to me:
 Does your Asterisk Server itself have something set up in the built in
 iptables firewall blocking udp inbound traffic in the port range
 15000:2? The output of the command 'iptables -nvL' will tell you
 pretty quickly.

 HTH.



   


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[asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio

[Yves Arikoglu]

Hi,

I am breaking my fingers in configuring an asterisk (1.6) to 
successfully transmit audio with the following setup:

asterisk, resides in local network, ip is 10.26.208.252
versatel business router (directly connected to a dsl, configured by 
sip-provider), WAN ip 89.244.13.25
versatel sip-proxy ip 89.244.13.10


in sip.conf I have:
[general]
bindaddr=0.0.0.0
externip=89.244.13.25
localnet=10.26.208.0/255.255.252.0
nat=yes
qualify=yes


the local sip phones register correctly and can make calls between each 
other with audio.
the local sip phones CAN make outbound calls via the sip-provider... 
will say, destination phone rings, but there is no audio (on both legs)
after pickup...
external phones can call my sip-number... the call comes into the 
asterisk, the sip-extension rings, but after pickup... no audio at all.
even if i route the call from external to a queue or something else... i 
see, that asterisk is playing voicefiles, but the caller does not hear
anything.
because sip-signalling works in any ways, but audio not, i think its got 
something to do with nat... but there is no firewall between asterisk
and the router or between the router and the internetconnection from 
versatel... and i already tried millions of combinations of using
nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m 
stuck as i was never ever stuck before :-(

any hints? anybody?

thanks,
yves


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Re: [asterisk-users] Odd error mssage on DAHDI lines

[Yves Arikoglu]

hi,

you can lookup the causes in the sources
check you dahdi-configuration (especially the groups...) is there 
everything ok? what does dahdi_tools or the other cli-commands say, that 
give you
information about the available channels?

yves

/* Causes for disconnection (from Q.931) */
   #define AST_CAUSE_UNALLOCATED1
   #define AST_CAUSE_NO_ROUTE_TRANSIT_NET  2
   #define AST_CAUSE_NO_ROUTE_DESTINATION  3
   #define AST_CAUSE_CHANNEL_UNACCEPTABLE  6
   #define AST_CAUSE_CALL_AWARDED_DELIVERED7
   #define AST_CAUSE_NORMAL_CLEARING16
   #define AST_CAUSE_USER_BUSY17
   #define AST_CAUSE_NO_USER_RESPONSE18
   #define AST_CAUSE_NO_ANSWER19
   #define AST_CAUSE_CALL_REJECTED21
   #define AST_CAUSE_NUMBER_CHANGED22
   #define AST_CAUSE_DESTINATION_OUT_OF_ORDER  27
   #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
   #define AST_CAUSE_FACILITY_REJECTED29
   #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY30
   #define AST_CAUSE_NORMAL_UNSPECIFIED31
   #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
   #define AST_CAUSE_NETWORK_OUT_OF_ORDER  38
   #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE  41
   #define AST_CAUSE_SWITCH_CONGESTION42
   #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
   #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL44
   #define AST_CAUSE_PRE_EMPTED45
   #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED   50
   #define AST_CAUSE_OUTGOING_CALL_BARRED  52
   #define AST_CAUSE_INCOMING_CALL_BARRED  54
   #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH  57
   #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58
   #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL  65
   #define AST_CAUSE_CHAN_NOT_IMPLEMENTED  66
   #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED  69
   #define AST_CAUSE_INVALID_CALL_REFERENCE81
   #define AST_CAUSE_INCOMPATIBLE_DESTINATION  88
   #define AST_CAUSE_INVALID_MSG_UNSPECIFIED   95
   #define AST_CAUSE_MANDATORY_IE_MISSING  96
   #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
   #define AST_CAUSE_WRONG_MESSAGE 98
   #define AST_CAUSE_IE_NONEXIST99
   #define AST_CAUSE_INVALID_IE_CONTENTS   100
   #define AST_CAUSE_WRONG_CALL_STATE  101
   #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE  102
   #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
   #define AST_CAUSE_PROTOCOL_ERROR111
   #define AST_CAUSE_INTERWORKING127



Richard Kenner schrieb:
 What's this:

 -- Attempting call on DAHDI/g1/9removed for application Wait(5) (Retry 
 1)
 -- Requested transfer capability: 0x00 - SPEECH
 -- Channel 0/2, span 1 got hangup, cause 44
 -- Forcing restart of channel 0/2 on span 1 since channel reported in use
 -- Hungup 'DAHDI/2-1'

 Where can I look up cause 44.  And if this is the sort of transient
 error that seems to be implied by the Forcing restart message, why
 isn't it retried?

   


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Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

[Yves Arikoglu]

do you use the

qualify=yes

option for your endpoints?

y.


Peter Childs schrieb:
 Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash

 I've managed to get a basic system set up. and can now take and make
 sip calls over the sip trunk I've got from sipgate.co.uk for testing
 purposes

 Anyway I can make calls fine (if only to the testing line and other
 sipgate lines as I have not set up any credit), and I can take calls
 but only if someone phones me within 2 minutes of doing a sip reload
 otherwise I just get a dead line.

 I'm thinking this is something to do with registration or Nat, but
 I've set my Nat up to forward everything, and it all works for
 2minutes.



 Peter.

   


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[asterisk-users] sip.conf with versatel and two NICs very strange problem

[Yves Arikoglu]

Hi

My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has 
the local ip 10.26.208.252
and the external ip 89.244.x.y

eth0 of the server is configured to 10.26.192.107

The Problem:
SIP registration works, phone rings in- and outbound, but there is no 
audio, nor the caller neither the callee
can hear anything.
So i am quite sure that is has something to do with firewalls, natting 
and so on but i?ve read hundreds of
pages and tried thousands of setting but i cant get audio to work..
the strange thing is... when i call the versatel-sip-number from my 
mobile phone, i see the call coming in
in the cli, i see the voiceprompts that asterisk plays, but even there I 
cant hear anything on my mobile.
next strange thing:
i defined 2 sip-extensions. both are registered... everything is fine... 
routes are ok, they can call out
and can be called from external and from internal (sip phones call each 
other).. but the same... no audio.
but when one sip extension calls a wrong number... the cannot be 
completed message is hearable.
i configured a queue with moh and even this works... but why cant to 
sip-phones talk to each other?
why cant an external caller hear any audio?

if i make sip debug, i see traffic (and due to extension is calling i 
think that on the sip-level everything
is okay...) how can i see, which port and interface is chosen for audio 
when a call comes in?

thanks,
yves


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Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem

[Yves Arikoglu]

thanks, i tried this already but unfortunately no change.
any further suggestions or answers concerning my other questions?

thanx, yves

Cary Fitch schrieb:
 As a guess, they can both talk to the server, but can't talk to each other.


 What is common to that is they may be trying to reinvite each other, and
 there is no path through the respective routers/firewalls to the other.

 So if reinvite is set to yes, set it to no, in both phone profiles on the
 server.

 Cary Fitch



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu
 Sent: Monday, January 25, 2010 7:28 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] sip.conf with versatel and two NICs very
 strangeproblem

 Hi

 My System is:
 Asterisk 1.6 running on a Dell Server with two network interfaces.
 eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has 
 the local ip 10.26.208.252
 and the external ip 89.244.x.y

 eth0 of the server is configured to 10.26.192.107

 The Problem:
 SIP registration works, phone rings in- and outbound, but there is no 
 audio, nor the caller neither the callee
 can hear anything.
 So i am quite sure that is has something to do with firewalls, natting 
 and so on but i?ve read hundreds of
 pages and tried thousands of setting but i cant get audio to work..
 the strange thing is... when i call the versatel-sip-number from my 
 mobile phone, i see the call coming in
 in the cli, i see the voiceprompts that asterisk plays, but even there I 
 cant hear anything on my mobile.
 next strange thing:
 i defined 2 sip-extensions. both are registered... everything is fine... 
 routes are ok, they can call out
 and can be called from external and from internal (sip phones call each 
 other).. but the same... no audio.
 but when one sip extension calls a wrong number... the cannot be 
 completed message is hearable.
 i configured a queue with moh and even this works... but why cant to 
 sip-phones talk to each other?
 why cant an external caller hear any audio?

 if i make sip debug, i see traffic (and due to extension is calling i 
 think that on the sip-level everything
 is okay...) how can i see, which port and interface is chosen for audio 
 when a call comes in?

 thanks,
 yves


   


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Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem

[Yves Arikoglu]

thanx... a typo... the routers local ip is 10.26.208.253


yves


Tim Nelson schrieb:
 - Yves Arikoglu yves...@gmx.de wrote:
   
 Hi

 My System is:
 Asterisk 1.6 running on a Dell Server with two network interfaces.
 eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has

 the local ip 10.26.208.252
 and the external ip 89.244.x.y

 

 Either a typo or you have an IP conflict?

 --Tim

   


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[asterisk-users] DTMF reception during WaitForSilence

[Yves Arikoglu]

Hello,

I wrote a little AGI-Script that implements an IVR (using asterisk 1.6).
The whole conversation is recorded and at some points the caller should 
tell some information.
I detect the silence (WaitForSilence) to go to the next step in the IVR. 
Until now everything is OK, but...
some information the user gives (or speaks) is numeric... some users 
have the habit, to enter numeric
information via the phonekeypad (ergo creating dtmf-tones) but I cant 
process DTMF-Input
during WaitForSilence.
How can I achive that both works simultaneously? I mean recording the 
spoken digits AND detecting
DTMF-Input AND detecting silence to know, when Input has finished... 
(I want to avoid that users
have to finish their input with the pound-key...) ?

Btw.: why are the DTMF-Tones, that a user enters, not hearable in the 
recording?

Thanks for your help and hints,
Yves

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[asterisk-users] Zaptel - DAHDI: now echo

[Jean-Yves Avenard]

Hello

I have upgraded our asterisk box from zaptel to dhadi two weeks ago...

Since, there has been quite a significant amount of echo when making a
call. Only for the local outgoing call, the person on the other side
doesn't hear any echo.

This is with a TE-110P ISDN PRI card ..

I've pretty much took the original zaptel configuration and used it
as-is with the dahdi one ; to no available..

Any help would be greatly appreciated.

Here is what zaptel.conf and zapata.conf used to be:
/etc/zaptel.conf:
loadzone = au
defaultzone=au
#TE110P
span=1,1,0,ccs,hdb3,crc4
bchan=1-10
dchan=16

/etc/asterisk/zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callerid=asreceived
restrictcid=no
usecallingpres=yes

; ISDN Exchange Lines (Fractional E1 PRA10)
switchtype=euroisdn
signalling=pri_cpe
immediate=no
pridialplan=unknown
prilocaldialplan=unknown
overlapdial=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=256
rxgain=1.0
txgain=8.0
context=incoming
faxdetect=incoming
group=1
channel=1-10


now for the dahdi configuration:
/etc/dahdi/system.conf:
loadzone = au
defaultzone=au
#TE110P
span=1,1,0,ccs,hdb3,crc4
bchan=1-10
dchan=16

/etc/asterisk/chan_dahdi.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callerid=asreceived
restrictcid=no
usecallingpres=yes

switchtype=euroisdn
signalling=pri_cpe
immediate=no
pridialplan=unknown
prilocaldialplan=unknown

echocancel=yes
echocancelwhenbridged=yes
echotraining=256

rxgain=1.0
txgain=8.0
context=incoming
faxdetect=incoming
group=1
channel=1-10

---

From reading the various documentation, I was convinced that moving
from zaptel to dahdi was almost just a matter of renaming the
configuration file... Am I mistaken ?

Thank you in advance for any help.

Jean-Yves

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Re: [asterisk-users] Zaptel - DAHDI: now echo

[Jean-Yves Avenard]

Hi

That was a fast answer, impressive !
2009/8/18 Kevin P. Fleming kpflem...@digium.com:

 Did you read the upgrade documentation that comes with DAHDI,
 specifically from UPGRADE.txt:

I did, but I guess I did not pay enough attention...


 * It is no longer possible to select a software echo canceler at
   compile time to build into dahdi.ko; all four included echo
   cancelers (MG2, KB1, SEC and SEC2) are built as loadable modules,
   and if the Digium HPEC binary object file has been placed into the
   proper directory the HPEC module will be built as well. Any or all
   of these modules can be loaded at the same time, and the echo
   canceler to be used on the system's channels can be configured using
   the dahdi_cfg tool from the dahdi-tools package.

   Note: It is *mandatory* to configure an echo canceler for the
   system's channels using dahdi_cfg unless the interface cards in use
   have echo canceler modules available and enabled. There is *no*
   default software echo canceler with DAHDI.



So, knowing my card (a Digium TE-110P, which AFAIK doesn't have any
hardware echo cancellation module)...
Which software echo canceller should I be using ?

Is see that there are particular software configuration available ,
but I haven't had a clue on what they are for, nor did I find
documentation about it...

I'm not building asterisk nor dahdi myself, but instead rely on
packaged from ATrpms.conf

Thank you
Jean-Yves

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Re: [asterisk-users] Zaptel - DAHDI: now echo

[Jean-Yves Avenard]

Hi

2009/8/18 Tzafrir Cohen tzafrir.co...@xorcom.com:
 Something is missing here...

 http://docs.tzafrir.org.il/dahdi-tools/#_echo_canceller_modules

Thanks ..

I added to /etc/dahdi/system.conf the following:
echocanceller=mg2,1-10

However, I have no clue about the various echo canceller, between mg2,
kb1, sec2, and sec which one will provide the best performance ?
(knowing that I was happy with whatever zaptel was doing before)

JY

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Re: [asterisk-users] Read Command

[Yves Räber]

You first use the Read application : 

exten = s,n,Read(ANS|filetoplay)

And then use GotoIfs by checking the ${ANS} variable to do the logic
(re-ask if bad response, else continue in dialplan).


On Sun, 2008-08-24 at 23:10 -0700, Joe Carroll wrote:
 I’ve search the world over….  but I haven’t figured out a way to have
 valid/invalid options for entry when using the Read command…
 
  
 
 I need to set a variable, but only want to allow certain values to be
 valid options for that variable…   
 
  
 
 Any ideas? 
 
  
 
 Thanks in advance..
 
  
 
 -JC
 
  
 
  
 
  
 
  
 
 
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Re: [asterisk-users] IVR question

[Yves Räber]

You could use func_odbc in your dialplan, check here : 

http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc

Yves.

On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote:
 Hi!
 
 I'm setting up my IVR system, how can I register in a mysql database the 
 IVR menus accessed by the clients ?
 
 Thanks a lot,
 
 Szasz Szabolcs
 
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Re: [asterisk-users] IVR question

[Yves Räber]

Sorry, maybe I misunderstood your question.

If you want the dialplan to be in a MySQL dabtase, check here : 
http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database

Works great, but the documentation is sometimes a bit outdated.

Good luck.

Yves.

On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote:
 Hi!
 
 I'm setting up my IVR system, how can I register in a mysql database
 the 
 IVR menus accessed by the clients ?
 
 Thanks a lot,
 
 Szasz Szabolcs
 
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[asterisk-users] Playback don't play the beginning if a sound file

[Yves Räber]

Hello,

I'm using this dialplan to let user record messages. The recording part
works quite fine, but there is something strange :

When Asterisk plays vm-torerecord, it misses the beginning, I only hear
the few last seconds (vm-torerecord is a sound file that was in the
asterisk-sounds cvs repo, but I simply renamed it). 

I've looked on voip-info.org, googled anything I could think about and
checked on bugs.digium.com, I don't have any clue of what's going on.

Does anyone has an idea ? Thanks.


Here is my dialplan :

[record]
exten = s,1,Answer
exten = s,n,Set(counter=1)
exten = s,n,NoOp(${counter})
exten = s,n,GotoIf($[${counter} = 1]?record)
exten = s,n(next),System(/bin/rm
-f /var/lib/asterisk/sounds/${RECORDED_FILE}.wav)
exten = s,n(record),Set(counter=$[${counter}+1]);
exten = s,n,GotoIf($[${counter}  3]?i,1)
exten = s,n,Playback(vm-intro)
exten = s,n,Record(webrecord%d:wav,10,60)
exten = s,n,Wait(1)
exten = s,n,Set(CDR(userfield)=${RECORDED_FILE})
exten = s,n,Playback(${RECORDED_FILE})
exten = s,n(askretry),Background(vm-torerecord)
exten = s,n,WaitExten(5)
exten = i,1,Goto(s,askretry)
exten = 3,1,Goto(s,next)
exten = t,1,Set(CDR(userfield)=${RECORDED_FILE})
exten = t,n,Hangup


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Re: [asterisk-users] Playback don't play the beginning if a sound file

[Yves Räber]

It seems this has something to do with the Wait() before the Playback
(Background behaves the same). 

If I remove the Wait, the next Playback is just fine, otherwise it
truncates the beginning of the message.

On Mon, 2008-05-05 at 10:41 +0200, Yves Räber wrote:
 Hello,
 
 I'm using this dialplan to let user record messages. The recording part
 works quite fine, but there is something strange :
 
 When Asterisk plays vm-torerecord, it misses the beginning, I only hear
 the few last seconds (vm-torerecord is a sound file that was in the
 asterisk-sounds cvs repo, but I simply renamed it). 
 
 I've looked on voip-info.org, googled anything I could think about and
 checked on bugs.digium.com, I don't have any clue of what's going on.
 
 Does anyone has an idea ? Thanks.
 
 
 Here is my dialplan :
 
 [record]
 exten = s,1,Answer
 exten = s,n,Set(counter=1)
 exten = s,n,NoOp(${counter})
 exten = s,n,GotoIf($[${counter} = 1]?record)
 exten = s,n(next),System(/bin/rm
 -f /var/lib/asterisk/sounds/${RECORDED_FILE}.wav)
 exten = s,n(record),Set(counter=$[${counter}+1]);
 exten = s,n,GotoIf($[${counter}  3]?i,1)
 exten = s,n,Playback(vm-intro)
 exten = s,n,Record(webrecord%d:wav,10,60)
 exten = s,n,Wait(1)
 exten = s,n,Set(CDR(userfield)=${RECORDED_FILE})
 exten = s,n,Playback(${RECORDED_FILE})
 exten = s,n(askretry),Background(vm-torerecord)
 exten = s,n,WaitExten(5)
 exten = i,1,Goto(s,askretry)
 exten = 3,1,Goto(s,next)
 exten = t,1,Set(CDR(userfield)=${RECORDED_FILE})
 exten = t,n,Hangup
 
 
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Re: [asterisk-users] Goto in Realtime extensions

[Yves Räber]

That's very unfortunate.

I use now a workaround : I'm just switching (with gotos) between
extensions and use some macros but always within the same context.

I'll try to remeber it for next time :)

Cheers,

Yves.

On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote:
 On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote:
  * Version: Asterisk 1.4.14
 
  * Commas instead of pipes = already tried, this is not working at all
 
  * Realtime switch for script_13_0 = No, should I ? This would be really
  a shame, I want to use realtime BECAUSE I don't want to play with my
  extensions.conf file. (I'm building a web interface that has to generate
  the contexts).
 
 Yes, unfortuneately that's the thing you have to do. You have to add
 each context you want - in static conf file like this:
 
 [db_na]
 switch = Realtime/db_na
 
 [db_busy]
 switch = Realtime/db_busy
 
 You can have as many extensions you like with whatever commands, but
 contexts still should be registered. Generally editing and debugging
 of complete dialplan in DB is not so easy - so you should keep your
 main logic in static, but use realtime for data that actually changes.
 
 Regards,
 Atis
 
 
  * Using numbers instead of 's' = already tried, no changes
 
  * Renaming contexts without underscores = tried it right now, no
  changes
 
  Thanks for all those ideas.
 
  Yves.
 
  On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote:
   On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote:
I would have been happy ... but it's not that. This query gives me the
right row (I double checked).
   
On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote:
 On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
  Hello,
 
  I'm having troubles while using the Goto function in a realtime
  extension. Here is the error message :
 
  -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1)
  -- Goto (script_13_0,s,1)
  [Feb  7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel
  'SIP/siemens1-081f56b0' sent into invalid extension 's' in context
  'script_13_0', but no invalid handler
 
  And I definitively have a row in my extensions table with context
  script_13_0, exten s and priority 1 !
 
  I also tried to goto in another context that is in my 
  extensions.conf
  file, and it works.
 
  Is this a restriction or a bug ? It seems that it's not possible to
  Goto to another context within the realtime extensions.

 It's impossible to guess what might be wrong, because you haven't 
 included
 a dump from your table.  Try a:

 SELECT * FROM extensions_table WHERE exten='s' AND 
 context='script_13_0'
 AND priority='1'

 If that fails, you have your answer.

  
   What version? You could try replacing pipes with commas. Do you have
   realtime switch statement for script_13_0? Can you try renaming
   context to not use underscores? Try using not s but any number (and
   create extension _X.)
  
   Regards,
   Atis
  
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 


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[asterisk-users] Goto in Realtime extensions

[Yves Räber]

Hello,

I'm having troubles while using the Goto function in a realtime
extension. Here is the error message :

-- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1)
-- Goto (script_13_0,s,1)
[Feb  7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel
'SIP/siemens1-081f56b0' sent into invalid extension 's' in context
'script_13_0', but no invalid handler

And I definitively have a row in my extensions table with context
script_13_0, exten s and priority 1 !

I also tried to goto in another context that is in my extensions.conf
file, and it works. 

Is this a restriction or a bug ? It seems that it's not possible to
Goto to another context within the realtime extensions.

Cheers,

Yves.



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Re: [asterisk-users] Goto in Realtime extensions

[Yves Räber]

I would have been happy ... but it's not that. This query gives me the
right row (I double checked).

On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote:
 On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
  Hello,
 
  I'm having troubles while using the Goto function in a realtime
  extension. Here is the error message :
 
  -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1)
  -- Goto (script_13_0,s,1)
  [Feb  7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel
  'SIP/siemens1-081f56b0' sent into invalid extension 's' in context
  'script_13_0', but no invalid handler
 
  And I definitively have a row in my extensions table with context
  script_13_0, exten s and priority 1 !
 
  I also tried to goto in another context that is in my extensions.conf
  file, and it works.
 
  Is this a restriction or a bug ? It seems that it's not possible to
  Goto to another context within the realtime extensions.
 
 It's impossible to guess what might be wrong, because you haven't included
 a dump from your table.  Try a:
 
 SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0'
 AND priority='1'
 
 If that fails, you have your answer.
 


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Re: [asterisk-users] Goto in Realtime extensions

[Yves Räber]

* Version: Asterisk 1.4.14

* Commas instead of pipes = already tried, this is not working at all

* Realtime switch for script_13_0 = No, should I ? This would be really
a shame, I want to use realtime BECAUSE I don't want to play with my
extensions.conf file. (I'm building a web interface that has to generate
the contexts).

* Using numbers instead of 's' = already tried, no changes

* Renaming contexts without underscores = tried it right now, no
changes

Thanks for all those ideas.

Yves.

On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote:
 On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote:
  I would have been happy ... but it's not that. This query gives me the
  right row (I double checked).
 
  On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote:
   On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
Hello,
   
I'm having troubles while using the Goto function in a realtime
extension. Here is the error message :
   
-- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1)
-- Goto (script_13_0,s,1)
[Feb  7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel
'SIP/siemens1-081f56b0' sent into invalid extension 's' in context
'script_13_0', but no invalid handler
   
And I definitively have a row in my extensions table with context
script_13_0, exten s and priority 1 !
   
I also tried to goto in another context that is in my extensions.conf
file, and it works.
   
Is this a restriction or a bug ? It seems that it's not possible to
Goto to another context within the realtime extensions.
  
   It's impossible to guess what might be wrong, because you haven't included
   a dump from your table.  Try a:
  
   SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0'
   AND priority='1'
  
   If that fails, you have your answer.
  
 
 What version? You could try replacing pipes with commas. Do you have
 realtime switch statement for script_13_0? Can you try renaming
 context to not use underscores? Try using not s but any number (and
 create extension _X.)
 
 Regards,
 Atis
 


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Re: [asterisk-users] Goto in Realtime extensions

[Yves Räber]

I'm not using labels at all (but I've also tried with :))

On Thu, 2008-02-07 at 16:39 -0800, Grey Man wrote:
 
 Make sure you don't have any labels on the prioritys. When loading
 extensions from realtime labels aren't supported.
 
 Replace:
 
 exten = _X.,1(mylabel),...
 
 with
 
 exten = _X.,1,...
 
 You'll have to make your Goto's use the prioritty instead of the label
 afterward.
 
 Regards,
 
 Greyman.
 


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Re: [asterisk-users] How to prevent logging of some entries in CDR

[Jean-Yves Avenard]

Hi

On Jan 21, 2008 11:05 PM, Jean-Yves Avenard [EMAIL PROTECTED] wrote:
 This works great. However in the CDR, than seeing one entry for each
 call, I see several entries in the CDR
 Worse, if I do something like:
 Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])


 40. 2008-01-21 13:59:34 Local/1...  04  MOB.
 04 04DialSIP/ipp100SIP/ipp100.1 100   
   NO
 ANSWER  00:11   incoming-zap
 41. 2008-01-21 13:59:34 SIP/ipp...  04  04
   s
 NO ANSWER   00:11
 42. 2008-01-21 13:59:34 SIP/ipp...  04  04
   s
 NO ANSWER   00:11
 43. 2008-01-21 13:59:33 Zap/7-1...  04  MOB. 
 04
 04DialLocal/[EMAIL PROTECTED]|10|tr286 
 ANSWERED00:12
 incoming-zap
 44. 2008-01-21 13:49:39 Local/1...  04  MOB. 
 04
 04DialSIP/ipp100SIP/ipp100.1 100 NO 
 ANSWER   00:05
 102 NO ANSWER   00:05
 52. 2008-01-21 13:49:39 Local/1...  04  MOB. 
 04
 04DialSIP/ipp119SIP/ipp119.1 119 NO 
 ANSWER   00:05


No one else is seeing this issue ?

Jean-Yves

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Re: [asterisk-users] How to prevent logging of some entries in CDR

[Jean-Yves Avenard]

Hi

On Jan 25, 2008 4:58 AM, John Faubion [EMAIL PROTECTED] wrote:
 I have the same issue but I haven't put much effort into solving it yet. Too
 many other issues seem to get in the way.


If you do, please post your results !

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