Displaying 20 results from an estimated 3000 matches similar to: "No sound from playback and voicemail"
2007 Dec 03
4
Soundcard necessary on an asterisk server to get output of playback()??
Hi,
I' still fighting the problem, that I can talk from one SIP phone to
another, but I can't hear the output of the playback or similar
applications:
exten => 202,1,ANSWER()
exten => 202,2,PLAYBACK(tt-monkeys)
exten => 202,3,HANGUP()
When I dial 202, asterisk show the following on the cli:
-- Executing [202 at local:1]
2007 Nov 12
0
No sound from playback and voicemail (Atis Lezdins)
Hello,
>> > I can talk to other SIP phones and via ISDN to the outside, but I
>> >don't hear playbacks or the voicemail messages.
>> > Asterisk show in the cli, that the corresponding files are played,
>> >but I hear nothing at all.
>> >
>> > Here is as simple example:
>> >
>> > [monkeys]
>> > exten =>
2008 Oct 31
5
twice normal beep before busy tone ??
Hi,
I have a strange problem with our Asterisk installation. Outgoing calls
are handled by the following lines:
exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)})
exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} =
0999940321]?099994030:${CALLERID(num)})})
exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr)
exten =>
2007 Dec 05
8
Strange ISDN-problem with incoming calls out of the same city
Hi,
after I fixed my problem with the playback() application, I now have the
next strange one.
When I dial the number of our client, located in another town, I get a
connection to the asterisk server, I can talk to my client or listen to
his mailbox.
If some in the town of this client calls him, he gets the ISDN error
"service not available".
Out office is connected to he office
2008 Mar 05
4
Problem between Asterisk and an Aastra 57i
Hi,
I'm currently trying to connect an Aastra 57i to our Asterisk Server.
The strange thing is, that altough I have definitely entered the correct
IP address of the server, the phone doesn't even attempt to register.
Here is the configuration file (local.cfg) of the phone:
firmware md5: dee6e938b469e217a87138076f47fe41
boot count: 1
tone set: Germany
language 1: German
time server1:
2008 Jun 02
1
Why doesn't Pickup() work??
Hi,
I'm using an Aastra 57i together with Asterisk 1.4.13. The 57i is
configured for call pickup as recommended by Aastra.
When the LED flashes, I press the corresponding button and the display
tells me "Call not possible". In the CLI is see the follwoing output.
Why isn't the call transfered to user2 and what does "No target channel
found for 31 mean"?
User2 can
2008 May 14
2
Setting CallerID UNKNOWN on an outgoing call
Hello,
on my ISDN phone I can configure that on the next outgoing call, my
telephone number should not be transmitted, instead it should be UNKNOWN.
How can I configure Asterisk to do the same? Is this a feature/parameter
of the driver (chan_capi) that I'm using?
BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any
difference.
Thanks for your help,
Stefan
--
2006 Jun 10
1
Voicemail records nonsense, but record() works (??)
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working voicemail box:
/etc/asterisk/extensions.conf
exten => 83086921,1,Answer
exten => 83086921,2,Dial(SIP/stefan,5,r)
exten => 83086921,3,VoiceMail,u111
exten => 83086921,4,Hangup
exten => 83086921,103,VoiceMail,b111
exten => 83086921,104,Hangup
/etc/asterisk/voicemail.conf
[default]
language=de
111 =>
2007 Aug 08
3
Siemens Openstage & Asterisk ?
Hi,
is anyone on the list using the Siemens Openstage phones together with
asterisk?
If yes, is it possible to use the programmable keys of these phones
together with Asterisk?
Thanks for any hints,
Stefan
--
********************************************
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
2008 Jan 13
2
Question about queues and the definition and agents
Paul wrote
>
>;Pause/unpause Queue
>exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)})
>exten => 424,2,Playback(unavailable)
>exten => 424,3,Hangup
>exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)})
>exten => 425,2,Playback(available)
>exten => 425,3,Hangup
>
Following your suggestion and a number of postings and articles I have
2008 Jan 11
1
Soundcard necessary on an asterisk server toget output of playback()?? -> Next step
Hi,
>> ATAL: Error inserting ztdummy
>> > (/lib/modules/2.6.22-14-386/misc/ztdummy.ko): Unknown symbol in
module,
>> > or unknown parameter (see dmesg)
>
>Are you sure that the source of your kernel is the same as the running
>kernel?
>
>I.E. Have a look at the source it is using while compiling Asterisk and
>compare that to uname -a
>
yes,
2006 Jun 12
2
Bug in Voicemail ??
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working voicemail box:
/etc/asterisk/extensions.conf
exten => 83086921,1,Answer
exten => 83086921,2,Dial(SIP/stefan,5,r)
exten => 83086921,3,VoiceMail,u111
exten => 83086921,4,Hangup
exten => 83086921,103,VoiceMail,b111
exten => 83086921,104,Hangup
/etc/asterisk/voicemail.conf
[default]
language=de
111 =>
2007 Oct 21
1
Sometimes echoes & Asterisk sometimes connects too early
Hello,
I have read the articles on echo cancellation
(http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a
solution to my problem.
We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M
PCI (current driver from EICON) and some SNOM 300/360.
There are few clients where we recognize echoes on both sides when we
call them via ISDN.
With some of these clients we
2007 Dec 04
1
Soundcard necessary on an asterisk server toget output of playback()??
Hi,
>However, I believe that zaptel >= 1.4.6 or zaptel 1.2 >= 1.2.21 should
>support hires timers for timing on kernel >= 2.6.22 .
>
>What version of Zaptel do you use?
>
I was using version 1.4.5.1
I just downloaded and installed version 1.4.7, configure/make/make
install finished without an error, but when is used
modprobe ztdummy
the system said:
FATAL: Error
2007 Nov 12
0
No sound from playback and voicemail (Carlos Chavez)
>On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote:
>> > Hello,
>> >
>> > I have a strange situation:
>> >
>> > I can talk to other SIP phones and via ISDN to the outside, but I
>don't hear
>> > playbacks or the voicemail messages.
>> > Asterisk show in the cli, that the corresponding files are played,
2006 Dec 07
1
Asterisk accepting calls to fast
Hi,
the german telco Colt Telekom has assigned the phone number block 56830-xxx to
one of our customers. In the diaplan we have setup extensions like the
following ones:
exten => 56830910,1,Answer()
exten => 56830910,2,Dial(SIP/bduerring,10,tr)
exten => 56830910,3,VoiceMail,u20
exten => 56830910,4,hangup
exten => 56830910,103,VoiceMail,b20
exten => 56830910,104,hangup
exten
2008 Jan 13
0
Soundcard necessary on an asterisk server to get output of playback()?? -> Next step
Tzafrir Cohen wrote:
> > The agent picks up the phone but neither the agent nor the caller >
> > here anything.
>So please provide a more complte trace and a the relevant partt of your
>dialplan.
>
Here is the relevant part of the dialplan:
[local]
exten => 98,1,Dial(SIP/sguenther,20,tr)
exten => 98,2,VoiceMail(98|u)
exten => 98,3,hangup
exten =>
2008 Jan 11
2
Question about queues and the definition of agents
Hi,
I have a question about the definition of agents.
The agents.conf file looks like this:
[general]
persistentagents=yes
[agents]
maxlogintries=5
ackcall=no
wrapuptime=500
musiconhold => default
group = 1
agent => 1311,1311,Tom
agent => 1531,1531,Tim
and here is the queues.conf:
[general]
persistentmembers = yes
[queue1]
musiconhold = default
strategy = rrmemory
servicelevel = 60
2009 Mar 24
2
HW-Recommendation: cell/mobile phone, capable of WLAN and SIP ??
Hello,
is anyone on the list using a normal cell/mobile phone which is able to
act as a SIP client over WLAN?
Or has anyone heard of a SIP client for cell/mobile phones running
windows mobile 6.x?
The phone should use SIP, when the asterisk server is reachable and
should automatically switch to a German telco if it is not reachable.
Thanks for any hints,
Stefan
--
2008 Feb 05
1
Mistake in the wiki's description of cmd Pickup() ?
Hi,
according to the description of Pickup() on page
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
I can use this command to pickup a call at a certain extensions.
When I try this with e.g.
exten => *8200,1,Pickup(200)
Asterisk tells me that the highest value for the Pickup command is 63.
Wenn I enter the number of a callgroup instead of an extension, I can
pickup the call.