Question

Tenor AS200 and Asterisk SIP

Asked by: mark_06

Hi,

I am currently trying to get my asterisk box to work properly with my Tenor AS200 (using SIP). Currently I can call the phones on the Tenor AS200 but I cannot call from the AS200 to any of my SIP phones.

My sip.conf is:

[1001] ;sip phone
type=friend
context=internal
username=1001
secret=welcome
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
outgoinglimit=1
incominglimit=2
;disallow=all
allow = g729
allow=ulaw
amaflags=default

[1002] ;sip phone
type=friend
context=internal
username=1002
secret=welcome1
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
outgoinglimit=1
incominglimit=2
;disallow=all
allow = g729
allow=ulaw
amaflags=default

[1003] ;Berwick
type=friend
context=internal
username=1001
secret=welcome2
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
outgoinglimit=2
incominglimit=2
;disallow=all
allow = g729
allow=ulaw
amaflags=default

[1004] ;Rye
type=friend
context=internal
username=1001
secret=welcome3
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
outgoinglimit=2
incominglimit=2
;disallow=all
allow = g729
allow=ulaw
amaflags=default

my extensions.conf is:
[default]
exten => 1001,1,Dial(SIP/1001|30)
exten => 1002,1,Dial(SIP/1002|30)
exten => 1003,1,Dial(SIP/1003|30)
exten => 1004,1,Dial(SIP/1004|30)
exten => 97073333,1,Dial(SIP/1003|30)
exten => 59851165,1,Dial(SIP/1004|30)
exten => 100,1,Dial(SIP/1003|30)
exten => 97072222,1,Dial(SIP/1002|30)

[internal]
include => default

Does anyone know what I am doing wrong? My thought is maybe I have missed something on the AS200 config.

Thanks

Mark

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Asked On
2007-02-25 at 04:06:12ID22411876
Tags

tenor

,

asterisk

Topic

Voice Over IP

Participating Experts
1
Points
500
Comments
2

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Answers

 

by: feptiasPosted on 2007-02-25 at 07:02:19ID: 18605143

To check if the SIP Invite requests are reaching Asterisk, you could enable SIP debug in Asterisk. The command, from the CLI prompt, is "sip debug". To switch it off again is "sip no debug".

You have provided a copy of the entries from SIP.CONF for the SIP phones, but the section that is more relevant is the one nearer the start of the SIP.CONF file: Please can you copy the [general] section, especially the settings like these:
domain = <ip_address>,context1
domain = <dns_domain>,context2

In the Tenor dial plan, what sip address do you use to forward calls to Asterisk? Is it an IP address or is it a DNS domain name?

 

by: mark_06Posted on 2007-02-27 at 04:09:08ID: 18616106

Thanks. I found the problem with the debug command. The Tenor was putting the country and area code infront of each number it was dialing.

Thanks

20120131-EE-VQP-002

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