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lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 2:59 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer '0049351111111' is now UNREACHABLE! Last qualify: 0 In the CLI I can see: Name/username Host Dyn Nat ACL Port Status 0049351111111/00493511111 192.168.200.11 D 5060 UNREACHABLE 0049351222222/00493512222 192.168.200.10 D 5060 OK (17 ms) 0049351333333 (Unspecified) D 5060 UNKNOWN 1234 (Unspecified) D 5060 UNKNOWN messagenet/1234567890 212.97.59.76 5061 Unmonitored pbxanika/00493511111111 172.16.34.132 5060 Unmonitored pbxfax/00493513333333 172.16.34.132 5060 Unmonitored pbxluca/00493512222222 172.16.34.132 5060 Unmonitored 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] Asterisk connects to another Test-VM with AsteriskNOW and to the italian provider Messagenet. Can someone suggest me, what can I do? I can send the configuration file, if they are needed. Thanks Luca Bertoncello (lucabert@lucabert.de) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

kevin.larsen at pionee...
Guest

Posted: Thu May 28, 2015 3:05 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Quote:

I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.

The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always UNREACHABLE.
I can just see in the log:

[May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
'0049351111111' is now UNREACHABLE! Last qualify: 0

In the CLI I can see:

Name/username Host Dyn Nat ACL Port Status
0049351111111/00493511111 192.168.200.11 D 5060 UNREACHABLE
0049351222222/00493512222 192.168.200.10 D 5060 OK (17 ms)
0049351333333 (Unspecified) D 5060 UNKNOWN
1234 (Unspecified) D 5060 UNKNOWN
messagenet/1234567890 212.97.59.76 5061 Unmonitored
pbxanika/00493511111111 172.16.34.132 5060 Unmonitored
pbxfax/00493513333333 172.16.34.132 5060 Unmonitored
pbxluca/00493512222222 172.16.34.132 5060 Unmonitored
8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline]

Asterisk connects to another Test-VM with AsteriskNOW and to the italian
provider Messagenet.

Can someone suggest me, what can I do?
I can send the configuration file, if they are needed.

What kind of phone are we talking about, both yours that works and your wife's that does not?

Can you ping the unreachable phone and does it respond to a ping?

Many phones will have a network test function built in to them to help you determine if the phone is properly connected to the network.

Do you see anything in the asterisk logs or the logs of the phone itself (providing the phone puts logs somewhere) that indicate a failure to register or to resolve the ip address of the asterisk server?

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 3:08 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

What kind of phone are we talking about, both yours that works and your
wife's that does not?

Right!

Quote:

Can you ping the unreachable phone and does it respond to a ping?

I can ping both phones from the VM

Quote:

Many phones will have a network test function built in to them to help you
determine if the phone is properly connected to the network.

Unfortunately not that...
I tried with Twinkle from my PC, using the same account of my wife
(configured IDENTICALLY to my account, just another username). It don't
work...
I presume, I configured something wrong in Asterisk...

Quote:

Do you see anything in the asterisk logs or the logs of the phone itself
(providing the phone puts logs somewhere) that indicate a failure to
register or to resolve the ip address of the asterisk server?

Unfortunately not... Just UNREACHABLE...

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
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kevin.larsen at pionee...
Guest

Posted: Thu May 28, 2015 3:22 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Quote:

What kind of phone are we talking about, both yours that works and your
wife's that does not?

Right!

Quote:

Can you ping the unreachable phone and does it respond to a ping?

I can ping both phones from the VM

Quote:

Many phones will have a network test function built in to them to help you
determine if the phone is properly connected to the network.

Quote:

Unfortunately not... Just UNREACHABLE...

Can you post the Manufacturer and Model of your phones (both of them if they are different)? That will help us look up what diagnostics/log files there might be on the phones.

Does the Twinkle software on the PC show any error messages?

If you watch the CLI in asterisk, does anything go by in there regarding a failed registration? If I get one of my phones programmed with an incorrect username/secret, it will try to register with the server, but can't. Those failed registrations do show up in the CLI.

Double check that you are not mistyping the credentials somewhere. If you do post the relevant parts of your config in here, you might want to obscure the secret.

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 3:30 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

Can you post the Manufacturer and Model of your phones (both of them if
they are different)? That will help us look up what diagnostics/log files
there might be on the phones.

Of course!
My phone is a Thomson ST2022 and my wife has a KE1020A

Quote:

Does the Twinkle software on the PC show any error messages?

Nope, just trying and then say "unable to connect"...

Quote:

If you watch the CLI in asterisk, does anything go by in there regarding a
failed registration? If I get one of my phones programmed with an
incorrect username/secret, it will try to register with the server, but
can't. Those failed registrations do show up in the CLI.

That's very strange... I expected these errors, but in the console I can't
see anything...

SOMETIMES, but just sometimes, if the phone of my wife tries to connect, I
see something like "connecting from 192.168.200.11" (I can't find the error
message anymore), and then:

[May 28 21:46:27] NOTICE[3592] chan_sip.c: Peer '00493512222222' is now
UNREACHABLE! Last qualify: 0

Quote:

Double check that you are not mistyping the credentials somewhere. If you
do post the relevant parts of your config in here, you might want to
obscure the secret.

Which part of the configuration do you need?

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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darryl at moores.ca
Guest

Posted: Thu May 28, 2015 3:37 pm Post subject: [asterisk-users] Peer is UNREACHABLE

I'd start by turning on sip debugging in asterisk

Quote:

sip set debug ip [your_phone_ip]

and use tcpdump or wireshark to see what the OS sees

tcpdump host [your_phone_ip] and udp port 5060

On 15-05-28 03:58 PM, Luca Bertoncello wrote:

Quote:

Hi list!

I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.

The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always UNREACHABLE.
I can just see in the log:

[May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
'0049351111111' is now UNREACHABLE! Last qualify: 0

In the CLI I can see:

Name/username Host Dyn Nat ACL Port Status
0049351111111/00493511111 192.168.200.11 D 5060 UNREACHABLE
0049351222222/00493512222 192.168.200.10 D 5060 OK (17 ms)
0049351333333 (Unspecified) D 5060 UNKNOWN
1234 (Unspecified) D 5060 UNKNOWN
messagenet/1234567890 212.97.59.76 5061 Unmonitored
pbxanika/00493511111111 172.16.34.132 5060 Unmonitored
pbxfax/00493513333333 172.16.34.132 5060 Unmonitored
pbxluca/00493512222222 172.16.34.132 5060 Unmonitored
8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline]

Asterisk connects to another Test-VM with AsteriskNOW and to the italian
provider Messagenet.

Can someone suggest me, what can I do?
I can send the configuration file, if they are needed.

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 3:41 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Darryl Moore <darryl@moores.ca> schrieb:

Quote:

I'd start by turning on sip debugging in asterisk

Quote:

sip set debug ip [your_phone_ip]

Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.16.34.133' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.200.11:5060:
OPTIONS sip:00493512222222@192.168.200.11:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.34.133>;tag=as1215345d
To: <sip:00493512222222@192.168.200.11:5060>
Contact: <sip:asterisk@172.16.34.133>
Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6@172.16.34.133
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
Date: Thu, 28 May 2015 20:39:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

repeated in loop...
Help that?

192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 the IP of the Asterisk server.

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

kevin.larsen at pionee...
Guest

Posted: Thu May 28, 2015 3:45 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Quote:

Darryl Moore <darryl@moores.ca> schrieb:

Quote:

I'd start by turning on sip debugging in asterisk

Quote:

sip set debug ip [your_phone_ip]

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.

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 3:51 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

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@172.16.34.132>;tag=as6dd12e05

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

darryl at moores.ca
Guest

Posted: Thu May 28, 2015 3:55 pm Post subject: [asterisk-users] Peer is UNREACHABLE

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:

Quote:

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

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.

kevin.larsen at pionee...
Guest

Posted: Thu May 28, 2015 4:02 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Quote:

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.

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 4:09 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Darryl Moore <darryl@moores.ca> schrieb:

Quote:

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@pbxluca/00493511111111
register => 00493512222222:MYSECRET@pbxfax/00493512222222
register => 00493513333333:MYSECRET@pbxanika/00493513333333
register => 4444444444:MYSECRET@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
callgroup=
pickupgroup=
dial=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
callgroup=
pickupgroup=
dial=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
callgroup=
pickupgroup=
dial=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@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@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 4:11 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

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@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

darryl at moores.ca
Guest

Posted: Thu May 28, 2015 4:35 pm Post subject: [asterisk-users] Peer is UNREACHABLE

I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a call from it?

On 15-05-28 05:09 PM, Luca Bertoncello wrote:

Quote:

Darryl Moore <darryl@moores.ca> schrieb:

Quote:

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?

lucabert at lucabert.de
Guest

Posted: Thu May 28, 2015 4:49 pm Post subject: [asterisk-users] Peer is UNREACHABLE

Darryl Moore <darryl@moores.ca> schrieb:

Quote:

Well, right now this phone USES the username 1234, on the AsteriskNOW (the
"later Telekom").
I really don't know why it tries to authenticate to my "own Asterisk"...

What I see right now, if I try to connect the phone of my wife to "my own
Asterisk":

-- Registered SIP '00493512222222' at 192.168.200.11 port 5060
[May 28 23:46:01] NOTICE[1350]: chan_sip.c:22933 sip_poke_noanswer: Peer '00493512222222' is now UNREACHABLE! Last qualify: 0

But, as I said, right now the phone is connected to the AsteriskNOW...

Well, now I must sleep...
Hope someone can suggest me something that I can try tomorrow.

Thanks a lot
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
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