asterisk users - May 2015 - Peer is UNREACHABLE

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> The phone you gave your wife is really old. Are you sure it supports SIP 
> OPTIONS? Can you make a call in or out to it? If you can, it is more 
> likely that it just doesn't support that and you can't use a
qualify
> statement.
No, I'm not sure.
And no, I can't make any call, right now... At least, not connected to my
Asterisk...
If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but
NOT my phone connected on my Asterisk, using the "proxy".
I can see that in the log:

[May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
mismatch, have <1234>, digest has <luca>
[May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
Failed to authenticate device "Test1" <sip:1234 at
172.16.34.132>;tag=as6dd12e05

Thanks
Luca Bertoncello
(lucabert at lucabert.de)

Ahh. Seen that before! That suggests to me that you don't have your 
sip.conf records setup right.

What's your sip.conf look like?


On 15-05-28 04:51 PM, Luca Bertoncello wrote:
> Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
>
>> The phone you gave your wife is really old. Are you sure it supports
SIP
>> OPTIONS? Can you make a call in or out to it? If you can, it is more
>> likely that it just doesn't support that and you can't use a
qualify
>> statement.
> No, I'm not sure.
> And no, I can't make any call, right now... At least, not connected to
my
> Asterisk...
> If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but
> NOT my phone connected on my Asterisk, using the "proxy".
> I can see that in the log:
>
> [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
> mismatch, have <1234>, digest has <luca>
> [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
> Failed to authenticate device "Test1" <sip:1234 at
172.16.34.132>;tag=as6dd12e05
>
> Thanks
> Luca Bertoncello
> (lucabert at lucabert.de)
>

> No, I'm not sure.
> And no, I can't make any call, right now... At least, not connected to 
my
> Asterisk...
> If I connect it to the other VM with AsteriskNOW I can call my Twinkle, 
but
> NOT my phone connected on my Asterisk, using the "proxy".
> I can see that in the log:
> 
> [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
> mismatch, have <1234>, digest has <luca>
> [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
> Failed to authenticate device "Test1" 
<sip:1234 at 172.16.34.132>;tag=as6dd12e05
> 
I know from your previous email that you are new to Asterisk. Have you 
created a dialplan that would allow you to call from one extension to 
another without going through your phone company? That is to say, call 
from your phone through Asterisk to your wife's phone?

You have two parts that you need to have in place for the basics to work. 
You need your sip.conf in order to tell asterisk what devices and phone 
trunks you have and you need extensions.conf to tell Asterisk how to route 
calls. Since you are new to this, you can start by getting the two phones 
to both register (sounds like one of them is and one probably is not). 
Then you get to where you can dial from one phone to the other and vice 
versa. From there you can add in the telephone company lines and the 
ability to dial in and out to the world.

I am still curious why you have both an Asterisk setup and an AsteriskNow 
setup? Is that just to play around with? At the end of the day you should 
just need one or the other.
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Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your 
> sip.conf records setup right.
> 
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:

register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register => 00493513333333:MYSECRET at pbxanika/00493513333333
register => 4444444444:MYSECRET at messagenet/4444444444

[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite

[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite

[pbxanika]
type=peer
defaultuser=00493513333333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493513333333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite

[messagenet]
type=peer
defaultuser=4444444444
secret=MYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming
outboundproxy=sip.messagenet.it
port=5061
fromuser=4444444444
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite


Here my extensions.conf:

[stdexten]
include => luca_incoming
include => fax_incoming
include => anika_incoming
include => messagenet_incoming

[luca_incoming]
exten => _00493511111111,1,Verbose(2,Call for Luca)
exten => _00493511111111,n,Dial(SIP/00493511111111)
exten => _00493511111111,n,Hangup

[fax_incoming]
exten => _00493512222222,1,Verbose(2,Call for FAX)
exten => _00493512222222,n,Dial(SIP/00493512222222)
exten => _00493512222222,n,Hangup

[anika_incoming]
exten => _00493513333333,1,Verbose(2,Call for Anika)
exten => _00493513333333,n,Dial(SIP/00493513333333)
exten => _00493513333333,n,Hangup

[messagenet_incoming]
exten => _4444444444,1,Verbose(2,Call from Messagenet)
exten => _4444444444,n,Dial(SIP/00493511111111)
exten => _4444444444,n,Hangup

[myproxy]
exten => _X.,1,Verbose(2,Call from ${CALLERID(num)} to ${EXTEN})
exten => _X.,n,GotoIf($["${CALLERID(num)}" =
"00493511111111"]?dialluca)
exten => _X.,n,GotoIf($["${CALLERID(num)}" =
"00493512222222"]?dialfax)
exten => _X.,n,GotoIf($["${CALLERID(num)}" =
"00493513333333"]?dialanika)
exten => _X.,n,Dial(SIP/pbxluca/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialluca),Verbose(2,Outgoing using pbxluca)
exten => _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialfax),Verbose(2,Outgoing using pbxfax)
exten => _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika)
exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r)
exten => _X.,n,Hangup

And here my users.conf:

[00493511111111]
fullname = luca
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgrouppickupgroupdial=SIP/00493511111111

[00493512222222]
fullname = fax
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgrouppickupgroupdial=SIP/00493512222222

[00493513333333]
fullname = anika
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgrouppickupgroupdial=SIP/00493513333333


Now I see this: if I call my phone (00493511111111) from Twinkle it works.
If I call it from the phone of my wife, logged in on the same AsteriskNOW of
Twinkle and able to speak with Twinkle, it does NOT work and I see that in the
Log of my Asterisk:

  == Using SIP RTP CoS mark 5
[May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch,
have <1234>, digest has <luca>
[May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed
to authenticate device "Test1" <sip:1234 at
172.16.34.132>;tag=as7855ffe5

(the phone of my wife is now logged in on AsteriskNOW with the user
"1234" and try
to call my phone with the same number I use from Twinkle, which works).

Very puzzled...

Thanks
Luca Bertoncello
(lucabert at lucabert.de)

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> I am still curious why you have both an Asterisk setup and an AsteriskNow 
> setup? Is that just to play around with? At the end of the day you should 
> just need one or the other.
Just why I need a second SIP-provider to check if all works, when Deutsche
Telekom activate the new line...
So I installed AsteriskNOW on a VM and configured it to serve a couple of
number.
Then I installed Asterisk on a second VM and configured it to connect to
AsteriskNOW (later will be Telekom) and Messagenet.

Dialplan and the other configuration were already sent...

Thanks
Luca Bertoncello
(lucabert at lucabert.de)