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lucabert at lucabert.de
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 Posted: Thu May 28, 2015 5:14 pm Post subject: [asterisk-users] Peer is UNREACHABLE |
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Darryl Moore <darryl@moores.ca> schrieb:
| Quote: | 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?
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Well, another information (then I **MUST** go sleep...):
I tried to use my mobile phone logging to my "own Asterisk" with the login
data of my wife's telefon.
Now this user is REACHABLE... So I think, it was a problem on her phone...
I can't call and receive calls. I think, that it's a problem of my Dialplan.
If I try to call the mobile phone from AsteriskNOW (later: "the world"), I
see that in Asterisk's log ("my own Asterisk"):
== Using SIP RTP CoS mark 5
[May 29 00:07:49] NOTICE[1106]: chan_sip.c:20163 handle_request_invite: Call
from '00493511111111' to extension '00493512222222' rejected because
extension not found.
That's very strange, since I call from Twinkle and it has the number "1234"...
If I call my mobile phone using my VoIP-phone (connected on the same "my own
Asterisk") I get that:
== Using SIP RTP CoS mark 5
== Call from 00493511111111 to 00493512222222
== Outgoing using pbxluca
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
[May 29 00:09:25] WARNING[1106]: chan_sip.c:12800 check_auth: username
mismatch, have <00493511111111>, digest has <00493512222222> [May 29
00:09:25] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to
authenticate device "00493511111111"
<sip:00493511111111@172.16.34.132>;tag=as058adbf2 == Everyone is
busy/congested at this time (1:0/1/0) == Spawn extension (myproxy,
00493512222222, 9) exited non-zero on 'SIP/00493511111111-00000004'
Maybe this is the same problem, since I didn't configured my own Asterisk to
manage "internal calls" (since I don't need to call my wife on VoIP...  )
And, last but not least, if I try to call from my mobile phone Twinkle I get
this:
== Using SIP RTP CoS mark 5
== Call from 00493512222222 to 1234
== Outgoing using pbxanika
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/1/0)
== Spawn extension (myproxy, 1234, 15) exited non-zero on
'SIP/00493512222222-00000006'
And if I try to call my VoIP-phone I get that:
== Using SIP RTP CoS mark 5
== Call from 00493512222222 to 00493511111111
== Outgoing using pbxanika
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
[May 29 00:12:02] WARNING[1106]: chan_sip.c:12800 check_auth: username mismatch, have <00493512222222>, digest has <00493511111111>
[May 29 00:12:02] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "00493512222222" <sip:00493512222222@172.16.34.132>;tag=as193c26b0
== Everyone is busy/congested at this time (1:0/1/0)
== Spawn extension (myproxy, 00493511111111, 15) exited non-zero on 'SIP/00493512222222-0000000a'
Maybe can these information help someone helping me?
Thanks a lot!
Luca Bertoncello
(lucabert@lucabert.de)
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adrian-lists at wombit...
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| Posted: Fri May 29, 2015 8:07 am Post subject: [asterisk-users] Peer is UNREACHABLE |
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Maybe shut off qualify for the peer? I think I tried twinkle a few
years ago and it didna (yes didna) like the qualify packet. the sip
options qualify packet is only needed to keep the UDP state tables in a
firewall if the peer is remote
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lucabert at lucabert.de
Guest
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| Posted: Fri May 29, 2015 8:12 am Post subject: [asterisk-users] Peer is UNREACHABLE |
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Zitat von Adrian Serafini <adrian-lists@wombit.com>:
| Quote: | Maybe shut off qualify for the peer? I think I tried twinkle a few
years ago and it didna (yes didna) like the qualify packet. the sip
options qualify packet is only needed to keep the UDP state tables
in a firewall if the peer is remote
|
Well, the same happens with my wife's phone...
I'll try later again...
Regards
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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