Hello
I have a weird behaviour with our local GSM (3G) provider -- several
SIP clients crash on the android phone, when switching to 3G network,
and in asterisks logs it looks like this - client registers on server
successfull and then crashesh immediately.
Here's suspicious part of asterisk log:
Hello
My CLI of 1.8.5 is black and white?
How do I re-enable the color highlighting?
Thanks
Nick
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webin
Hi, my log is full of errors from this mobile user:
-- Registered SIP '0010106' at 212.93.97.135:7759
[2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804
handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms /
1ms)
[2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245
ast_sockad
actually i just noticed that it quits console because asterisk
restarts itself after:
[2011-05-16 13:48:45] ERROR[11106] tcptls.c: Unable to connect SIP
socket to 192.168.1.108:5060: Connection timed out
--
_
-- Bandwidth and Col
Hi!
I've noticed 1.8.4 keeps quitting console by itself. Is this a bug or
feature? :)
Nick
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar ever
Hi!
Here's a user with mobile phone - however why does it treat this as ERROR ?
I have a log full of that ---
-- Registered SIP '0010106' at 212.93.100.181:3698
[2011-05-12 16:07:57] NOTICE[30258]: chan_sip.c:19679
handle_response_peerpoke: Peer '0010106' is now Reachable. (212ms /
1ms)
le native formats is 0x8 (alaw) read/write = 0x8
(alaw)/0x8 (alaw)
[2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to
transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8
(alaw)/0x8 (alaw)
Hi!
Maybe someone could help me out?
When a call is routed via a2billing AGI and user does a transfer, the
call is dropped. If the trunk is called directly everyhing works.
Here's a direct scenario (working fine):
[pbx01]
exten => 101,1,Set(__TRANSFER_CONTEXT=pbx01)
exten => 101,n,Dial(S
Well, it has disappeared in further builds ;)
Thanks
2011/3/16 Leif Neland :
> Den 19-01-2011 00:19, Nick Ustinov skrev:
>>
>> Hello!
>>
>> I have just upgraded to asterisk 1.8.2.1 and see some weird messages
>> in log when client tries to register:
>>
Hello
using asterisk 1.8, compiled res_fax.so and res_fax_spandsp.so - both
loaded successfuly
in sip.conf set t38pt_udptl=yes
but faxes still don't work even in passthru mode.
if i do a 'sip show channel' on the channel via which i am sending fax it shows:
T.38 supportYes
however
These are the same for sip users and trunks
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
Who is asking to transmit frame type slin ?
Nick
On Thu, Mar 10, 2011 at 1:02 AM, Paul Belanger wrote:
> On 11-03-09 02:26 PM, Nick Ustinov wrote:
>> Using asterisk 1.8.4-rc2
>>
Hello!
Client is using ulaw, however server sometimes fills the log with following:
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit
frame type slin, while native formats is 0x8 (alaw) read/write = 0x4
(ulaw)/0x8 (alaw)
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to
Hello !
My asterisk log is full of messages like this:
[2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP
2833 Event: 009f. Not a DTMF Digit.
[2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP
2833 Event: 009f. Not a DTMF Digit.
[2011-03-06 19:01:25] DEB
Hello!
I have just upgraded to asterisk 1.8.2.1 and see some weird messages
in log when client tries to register:
[2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info
[2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now
UNREACHABLE! Last qualify: 105
[20
With asterisk 1.8+ it should be:
failregex = NOTICE.* .*: Registration from '.*' failed for
'(:[0-9]{1,5})?' - Wrong password
NOTICE.* .*: Registration from '.*' failed for
'(:[0-9]{1,5})?' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
'(:[0-9]{1,5
after some deep tracing it turned out to be a faulty router problem
thanks.
On Sun, Dec 26, 2010 at 9:38 AM, Sherwood McGowan
wrote:
> On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov wrote:
>> Hello
>>
>> We have recently upgraded to Realtime engine (sip buddies and
&
Make sure you have
dateformat=%F %T
in logger.conf
On Sun, Dec 26, 2010 at 1:04 AM, Dave George wrote:
> My server is being attached all day and fail2ban is not stopping the
> attack. I updated stamstamp to match fail2ban requirements.
>
> [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830
Hello
We have recently upgraded to Realtime engine (sip buddies and
extensions) and now have problems with calling local SIP users.
I have rtcachefriends=yes but tried with 'no' and it's even worse.
(asterisk 1.8.1.1 + realtime mysql)
Here's an example:
User 1000 registers successfully and can t
Hello!
Ater several successful SRTP-enabled calls with SRTP set to Mandatory,
asterisk starts to give the following warnings in Log:
WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure
(continiously)
and client hears no sound. After i restart the client program it works
fine again
Hello!
Ater several successful SRTP-enabled calls with SRTP set to Mandatory,
asterisk starts to give the following warnings in Log:
WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure
(continiously)
and client hears no sound. After i restart the client program it works
fine again f
Hello!
Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk
starts to give the following warnings in Log:
WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure
(continiously)
and client hears no sound. After i restart the client program it works fine
a
21 matches
Mail list logo
