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asterisk01 at in-put.de
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 Posted: Fri Jan 11, 2008 11:21 am Post subject: [asterisk-users] Soundcard necessary on an asterisk server t |
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Hi,
| Quote: | | Quote: | ATAL: Error inserting ztdummy
| Quote: | (/lib/modules/2.6.22-14-386/misc/ztdummy.ko): Unknown symbol in
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module,
| Quote: | | Quote: | | Quote: | or unknown parameter (see dmesg)
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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
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yes, I'm sure. There is only one source code directory on this system
and I have already used it to compile the driver for an EICON/DIALOGIC
card. And these drivers works perfectly.
BTW: Do I need a soundcard to use the local channel in a queue?
Asterisk forwards calls to agents like this:
-- Executing [98 at local:1] Answer("Local/98 at local-a81e,2", "") in new stack
The agent picks up the phone but neither the agent nor the caller here
anything.
Thanks for your help,
Stefan
--
********************************************
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
********************************************
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
******************************************** |
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tzafrir.cohen at xorco...
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| Posted: Fri Jan 11, 2008 6:43 pm Post subject: [asterisk-users] Soundcard necessary on an asterisk server t |
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On Fri, Jan 11, 2008 at 05:21:27PM +0100, Stefan Guenther wrote:
| Quote: | BTW: Do I need a soundcard to use the local channel in a queue?
Asterisk forwards calls to agents like this:
-- Executing [98 at local:1] Answer("Local/98 at local-a81e,2", "") in new stack
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No. A sound card is not needed.
| Quote: |
The agent picks up the phone but neither the agent nor the caller here
anything.
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So please provide a more complte trace and a the relevant partt of your
dialplan.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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asterisk01 at in-put.de
Guest
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| Posted: Sun Jan 13, 2008 9:20 am Post subject: [asterisk-users] Soundcard necessary on an asterisk server t |
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Tzafrir Cohen wrote:
| Quote: | | Quote: | The agent picks up the phone but neither the agent nor the caller >
here anything.
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| Quote: | So please provide a more complte trace and a the relevant partt of your
dialplan.
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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 => 98,101,VoiceMail(98|b)
exten => 98,102,Hangup
; QUEUES
exten => 6661,1,ANSWER()
exten => 6661,2,Queue(queue1|t)
; AgentLogin
exten => 6662,1,ANSWER()
exten => 6662,2,AGENTCALLBACKLOGIN(||${CALLERID(num)}@local)
exten => 6663,3,HANGUP()
Here is the ouput of "agent show"
intranet*CLI> agent show
666 (ceo) not logged in (musiconhold is 'default')
777 (smguenther) available at '98 at local' (musiconhold is
'default')
888 (michaela) not logged in (musiconhold is 'default')
999 (user1) not logged in (musiconhold is 'default')
4 agents configured [1 online , 3 offline]
And here is the output of the cli (debug level 5). user dials 6661, the
number of the queue, the phone of sguenther rings and I pick up the
phone, but you hear nothing on either side.
-- Executing [6661 at local:1] Answer("SIP/user1-081dda50", "") in new stack
-- Executing [6661 at local:2] Queue("SIP/user1-081dda50", "queue1|t") in
new stack
-- Started music on hold, class 'default', on SIP/user1-081dda50
-- outgoing agentcall, to agent '777', on 'Local/98 at local-c3b1,1'
-- Called Agent/777
-- Executing [98 at local:1] Dial("Local/98 at local-c3b1,2",
"SIP/sguenther|20|tr") in new stack
-- Called sguenther
-- Agent/777 is ringing
-- SIP/sguenther-082093a0 is ringing
-- SIP/sguenther-082093a0 is ringing
-- SIP/sguenther-082093a0 is ringing
-- SIP/sguenther-082093a0 answered Local/98 at local-c3b1,2
-- Agent/777 answered SIP/user1-081dda50
-- Stopped music on hold on SIP/user1-081dda50
When user1 calls the number 98 directly, everything is okay, sound on
both sides of the line.
Thanks for your help,
stefan
--
********************************************
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
********************************************
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
******************************************** |
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