[Asterisk-Users] SwissVoice MGCP IP10S

Marian Danisek majo at sunteq.sk
Sat Nov 1 08:29:04 MST 2003


rnc Info Lists wrote:
>>Hi,
>>
>>
>>>-----Original Message-----
>>>
>>>>The portion of extensions.conf is:
>>>>exten => 3001,1,Dial(MGCP/aaln1,20)
>>>
>>>exten => 3001,1,Dial(MGCP/aaln/1 at ip10,20)
>>
>>Or aaln/1@<ip> should do just fine. However this doesn't explain why there
>>is no dialtone on the phone..
>>
>>Oh, one thought: Did you set your toneconfiguration to Europe or US ? If
>>you
>>choose custom you need to configure it another way...
>>
>>Florian
>>
> 
> Update:
> I changed the tone config to USA to match Asterisk. No change.  I did
> notice that when I booted up everythign tonight that the MGCP SHOW
> ENDPOINTS now shows:
> Gateway 'ip10' at 0.0.0.0 (Dynamic)
>    -- 'aaln/1 at ip10 in 'from-sip' is idle
> 
> In the messages at start up there is:
> == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
>     -- MGCP Auditing endpoint aaln/1 at ip10 for hookstate
>  [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099
> (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1')
> does not exist
>     -- Setting hookstate of aaln/1 at ip10 to ONHOOK
> 
> 
> MGCP DEBUG shows the below lines repeating every couple of seconds:
> from 192.168.0.5:2427MGCP read:
> RSIP 1375 aaln/1@[192.168.0.5] MGCP 1.0 NCS 1.0
> RM: restart
> 
> from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint:
> 'aaln/1@[192.168.0.5]', Version: 'MGCP 1.0'
> 2 headers, 0 lines
> 
> Still no dialtone and not able to send or receive calls.
> 
> Evidently there is a problem finding the phone.  I can ping it from the
> Asterisk server so isn't a raw IP issue.  On the phone there is the
> message "Waiting for call manager"
> 
> Additional ideas are appreciated. Will keep plugging away at it.

in sending you my mgcp.conf file, my ip10s mostly working fine...

regards Marian

-------mgcp.conf-----------------

[general]
port = 2427
bindaddr = 192.168.1.253


[192.168.1.92]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
host=192.168.1.92
context=local
nat=no
;dtmf=inband
disallow=all
allow=g711
allow=alaw
callerid = John <92>
line => aaln/1

[192.168.1.91]
threewaycalling=yes
transfer=yes
callwaiting=no
callwaitingcallerid=no
host=192.168.1.91
context=local
nat=no
;dtmf=inband
disallow=all
allow=g711
allow=alaw
callerid = Mary <91>
line => aaln/1


> 
> Robert
> 
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> Asterisk-Users at lists.digium.com
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