Displaying 20 results from an estimated 10000 matches similar to: "Hangup()-command does not hang up the line"
2010 Apr 13
2
SNOM M9 base station A to base station B
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<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
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I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2009 Sep 07
2
features.conf : feature map ==> getting feature to work
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =>
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten => s,n,Dial(SIP/grandstream,30)
I want the callee to be able to press #3 to be able
2010 Mar 01
2
Is answer() necessary ?
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other hand, sometimes an incoming call is send to a macro where
the caller is given the
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2009 Oct 21
3
Searching on how to keep local calls... local
Hi list.
Does anyone know how to keep calls between 2 local SIP-phones on the
local private network when the 2 local IP-phones are registered to an
online public Asterisk-server ??
What network-element / router do I need to install to prevent the
RTP-traffic from flowing via the internet ?
Config :
Asterisk --internet-- > router/firewall --> connected local IP-phones
Internal call :
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2009 May 30
2
Simplex voice on TDM410P
Hello,
I am working on a trixbox based system with a TDM410P connected to 3
phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
with some polycom and Aastra SIP phones. In general everything works.
the problem I am trying to solve is that if both parties to a call speak
at the same time one of the voices gets cut out such that the talker A
cannot hear what talker B is
2009 May 08
2
Not receiving voicemail message in mailbox
It should be as simple as editing voicemail.conf :
; Voicemail Configuration
;
[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
serveremail=asterisk-voicemail
; Should the email contain the voicemail as an attachment
attach=yes
;
2009 Oct 14
2
FXS to SIP gateway
Hello list !
I don't have the money to test out all the products and reading the
manuals is not always that enlightening...
Does someone here know a good gateway-product that lets analogue
telephones communicate with an Asterisk-server.
I have found the Grandstream GXW-400x to be able to add SIP-accounts to
analogue telephone devices that are connected to the FXS-ports. Moreover
this
2009 Aug 29
1
GoToIfTime : how to define sep 25th till oct 10th ?
Hi list,
quick question :
With GoToIfTime, how to define a period of holiday that starts at the
end of the month and ends at the beginning of the next month ??
Like September 25th till October 10th when incoming calls need to go to
the voicemail...
Greetingz,
Jonas.
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2009 Aug 18
2
You do not appear to have the sources for the 2.6.20-prep kernel installed
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
"You do not appear to have the sources for the 2.6.20-prep kernel
installed."
I have installed :
- kernel-headers-2.6.18-128.4.1.el5.x86_64
- kernel-devel-2.6.18-128.4.1.el5.x86_64
- kernel-xen-devel-2.6.18-128.4.1.el5.x86_64
bash-3.2# uname -r
2.6.20-prep
bash-3.2# ls -l
2009 Aug 30
2
MySQL syntax error : I really don't see where...
Hi list,
I'm stuck for the moment @ the following :
My Query (in a macro) :
exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\
vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\
SIPACCOUNT="${ARG1}")
Asterisk CLI :
[Aug 30 14:07:42] -- Executing [s at macro-vakantie:2]
MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set
2009 Sep 20
1
Experience with Sangoma's USBfxo
Hi,
I've seen this USB product from Sangoma :
http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html
Is it working ok ?
Is it easy to integrate it with Asterisk ?
How would you rate your experience with it ?
Regards
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2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
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2013 Jun 06
1
Hangup cause 111 after call pickup
Hello,
when picking up an incoming call from one ip phone on another ip phone,
the call terminates after about 5 to 10 seconds.
When reading out the hangup cause variable in the h-extention of the
dialplan, the hangup cause seems to be 111.
In the dialplan output, you can see that SIP-peer sipacc3 picks up the
incoming channel SipAgenT01-00001454, and the call is answered. After 7
seconds,
2009 May 22
1
VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
Don't be afraid about the info that I'm going to post in this mail, but
I want you to give as much info as possible. Also I want to show you
what I've tried.
What do I want
When a voicemail-message is left via the Voicemail()-application, I want
the .wav-file send to my mail-address as an attachment.
My mail-setup
I'm not using sendmail as MTA. I have msmtp as MTA and mutt as
2009 Jul 06
2
SIP registry fails during night
Every morning I check my SIP registry to the SIP-provider. And I must
conclude that during the night somewhere registry has failed.
asterisk*CLI> sip show registry
Host Username Refresh State
Reg.Time
85.119.188.3:5060 092779077 105 Failed Sun,
05 Jul 2009 23:11:40
asterisk*CLI> sip reload
[Jul 6 10:30:43]
2009 Sep 17
1
I'm not getting the ability to leave a voicemail-message
I'm having a little problem with voicemail. Actually I'm not getting the
ability to leave a voicemail-message.
This is part of the dialplan :
> exten => s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1})
> exten => s,n,NoOp(${ARG1}@boxes)
> exten => s,n,Voicemail(${ARG1}@boxes)
> exten => s,n,Hangup()
> exten => s,n,MacroExit
This is the
2009 Apr 26
1
file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
part of extensions.conf:
exten => 11,1,Answer()
exten => 11,n,NoOp(CallerID : ${CALLERID(all)})
exten => 11,n,Playback(/tmp/welkom-tcs.alaw)
exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten => 11,n,NoOp(Oproep tijdens winkel gesloten)
exten => 11,n,Playback(/tmp/winkel-gesloten.alaw)
exten =>
2012 Mar 07
1
Finish ChanSpy() when channel spied hangs up
Is there any way to do this?
Thanks
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