Displaying 20 results from an estimated 2000 matches similar to: "Sip-Client / type=peer / Why can this client place calls?"
2010 Nov 11
3
Limit Call Duration with L-option of Dial : announcement
Hello,
Limiting the call duration with the L-option of the Dial()-command is
working fine, however the announcement is not played.
Dialplan :
exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000))
The call lasts for 11 seconds, but 5 minutes before time runs out an
announcement should come. I hear no announcement, not on caller-side nor
on
2009 Nov 09
4
local channels
I am using the AMI to dispatch (2) calls to Local/my_priority at my_context
where:
[my_context]
exten => my_priority,1,Answer()
exten => my_priority,n,Dial(${LOCAL_DIAL})
and LOCAL_DIAL has the actual phone number to dial.
The first call goes through just fine and I see DAHDI/1/XXXX being
called. The second call I see
DAHDI/2/XXXX and a message about everyone is busy on congested.
I
2017 Feb 21
3
What is the proper usage of LLVMContext?
Hi, I'm Ryo Ota. I have two questions about `llvm::LLVMContext`.
Q1) What is the difference between the following (A)`my_context` and
(B)`global_context`?
Do I have to create a LLVMContext by myself? Or use `getGlobalContext()`?
Could you tell me what situation needs a LLVMContext which is created by
myself such as (A)?
(A)
{
llvm::LLVMContext my_context;
// codes using only my_context (get
2013 Sep 03
1
Asterisk 11.5.1 / TLS and Media Encryption / Blink as Client / no audio
Hi,
I use Asterisk 11.5.1 and it works fine. :)
Now I want to use TLS and media encryption. I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
When I place a call via Blink-Client (0.5.0) I get connected and Blink
shows 2 locks. The blue lock shows "Signaling is encrypted using TLS"
and the orange lock shows "Media is encrypted using
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2011 Mar 02
1
GSM-Card for Asterisk / recommendation needed
Hi,
I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I"
from voismart (http://www.voismart.it/) but the driver is very bad
(compile-problems and no echo cancellation).
Is there anybody out there who can recommend me another piece of
hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable
and have an echo cancelltaion feature. And of course it
2015 May 28
4
Peer is UNREACHABLE
Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> The phone you gave your wife is really old. Are you sure it supports SIP
> OPTIONS? Can you make a call in or out to it? If you can, it is more
> likely that it just doesn't support that and you can't use a qualify
> statement.
No, I'm not sure.
And no, I can't make any call, right now... At least,
2012 Jun 18
4
Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
cat /etc/odbc.ini
------------------
[MySQL-asterisk]
Description = MySQL ODBC Driver
Driver = MySQL
Socket = /var/run/mysqld/mysqld.sock
Server = localhost
User = xxx
Password = xxx
Database = asterisk
Option = 3
Port =
and
/etc/odbcinst.ini
2014 Nov 26
5
Strange Issue: asterisk deleted
Hi,
I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there.
I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console.
Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too.
Now I used
2013 Aug 18
1
Asterisk SIP Trunk between two Asterisk Servers
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port
extensions.conf
[man02-trunk]
exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
exten
2017 Dec 14
4
SIP trunks going to the wrong context
Hi all,
I'm trying to resolve a weird issue with SIP routing.
I have a number of SIP trunks, from a selection of providers, all of
which are registered in sip.conf:
[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp
bindport=15060
srvlookup=yes
allowsubscribe=yes
2011 Jun 08
1
PRI hangup request, cause 18
We have 2 PRI from AT&T
And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised
[Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18
[Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,
2013 Feb 18
3
Dialplan / check / tool
Hi,
I am wondering, if there is any tool available, which performs a check
for suspicious entries in the dialplan. For example a non existing
AGI-Script or a double assigned extension ike that:
[context]
exten => *100*,1,AGI(test_app.pl)
...
exten => 190,1,Answer()
...
exten => *100*,1,AGI(never_reached.pl)
...
A "normal dialplan reload command" would echo no warning or
2015 Jun 07
3
Curious problem with NAT
Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb:
> What settings have you got for directmedia?
>
> Could you try
>
> nat=force_rport,comedia
> directmedia=no
Tried. Peer always unreachable, call not possible... :(
Other idea?
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2015 Jun 07
2
Curious problem with NAT
Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb:
> Have you tried NAT=force_rport ?
OK, tried...
I can transmit from my phone (aka: I hear my voice on another phone), but I'm
not able to receive data (aka: I cannot hear what I say on the other phone).
Other suggestion?
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2014 Jan 15
2
Asterisk ignoring nat settings
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the "sip show peer extension", I see both symmetric RTP and
Force Rport are set to yes, but asterisk seems ignoring them.
Force rport : Yes
Symmetric RTP: Yes
Asterisk is behind a
2014 Apr 16
2
FW: clients unable to auth
Hi Guys,
Just new to Asterisk and am completely stumped. I have created two accounts
as instructed. Please see below for the config of the user accounts.
[Peter]
type=friend
host=IP address
disallow=all
allow=ulaw
allow=alaw
callerid=Peter <6004>
secret=XXXXXXX
context=default
port=9060
nat=force_rport,comedia
deny=0.0.0.0
2014 Sep 18
1
Voice-Recognition / ASR / with barge in
Hi there,
I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine
:-) But I am wondering if there is a solution/application which will
enable me to implement voice recognition while playing a voice file
(barge in). So that the caller hears a voice file and can interrupt it
with his voice.
Currently (on our platform) the caller has to wait for the end of the
voicefie. Then we play
2017 Jan 24
2
Asterisk 13.13.1
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!
I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf
[1091]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1091>
disallow=all
allow=ulaw
2010 Nov 12
1
Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?
Hi,
it's me again with a cdr-issue. I have the following example
extensions.conf:
# dial in
exten => 100,1,Playback(hello)
exten => 100,n,Dial(local/200,20,rtg)
exten => 100,n,Playback(pleasewait)
exten => 100,n,wait(10)
exten => 100,n,Playback(goodbye)
exten => 100,n,Hangup
# for local dial
exten => 200,1,Playback(hello)
exten => 200,n,wait(10)
exten =>