Displaying 20 results from an estimated 3000 matches similar to: "SIP peer disappearing"
2010 Apr 17
1
Realtime changes not reflected realtime
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<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
Using Asterisk 1.4.25.1<br>
Using realtime sip_buddies<br>
<br>
I notice
2010 Sep 06
1
Asterisk Fax
Hi
I know that this topic was on the list maybe dozen of times. But I
have a question regarding the fax support in asterisk, because all the
information I could get does not give me the clear view of if. I read
that Asterisk 1.8 will have strong fax (t.38) support, but I want to
know if these four scenarios will be possible to achieve:
fax machine (phone+fax) connected to ATA --- SPA2102 ATA ---
2016 Aug 15
5
Realtime SIP peers do not register any more after upgrade to Asterisk 13
Hello
after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
none of my realtime SIP peers (saved in MySQL DB) register anymore.
[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'<sip:testacc77 at 178.19.90.240>' failed for '78.119.140.190:5076' - Wrong
password
[Aug 15 22:04:13] NOTICE[30098]:
2011 Dec 27
3
how to stop hacking of my server
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account only. bcoz I can't apply iptables rules on server it's
remote server of server provider and we used it for making voip call for
customers.
for the time been i have close all sip accounts. but can't stop for more
then
2013 Apr 09
0
realtime peer w/ callbackextension does not register after 'sip reload'
Hello everybody,
I am having a problem with realtime SIP peers.
On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register.
Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'
2010 Dec 25
2
sip attack.. fail2ban not stopping attack
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
handle_request_register: Registration from '"7002" <sip:7002 at x.x.x.x>'
failed for '38.108.40.94' - No matching peer found
[2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830
2010 Jan 29
2
Questions about asterisk and spa2102
Hi there! First mail on the list :)
1.- is it possible to use an spa2102 to make and revice calls from a
"normal" phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.
2.- I'm trying to unlock the spa2102 with no succes at the moment, any
links or hint will be very
2014 Apr 04
4
Asterisk 1.6
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '"4941" <sip:4941 at public_ip>' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '"4941"
2015 Feb 16
1
Asterisk 11.6. SIP realtime lost peers after 'sip reload'
Hi, list.
We have a problem with loss peers after 'sip reload', our configuration:
Asterisk 11.6-cert1, SIP realtime peers, sip.conf:
- rtcachefriends=yes
- rtsavesysname=yes
- rtupdate=yes
- rtautoclear=yes
When we do 'sip reload' , peers are removing from available.
Before `sip reload` :
srv-pbx2*CLI> sip show peers
Name/username Host
2011 Mar 06
0
imsdroid on droidX to asterisk: No matching peer found
sip.conf:
[imsdroid]
type=friend
;;auth=md5
;;defaultuser=imsdroid
secret=mysecret
host=dynamic
context=cloud-out
qualify=60
dtmfmode=auto
insecure=port,invite
callerid="IMSDroid client" <imsdroid>
disallow=all
allow=ulaw
I've tried with and without defaultuser and secret.
sip show peer imsdroid:
* Name : imsdroid
Secret : <Set>
MD5Secret :
2006 Apr 06
2
build error : xapian-core-0.9.4_svn6707
I posted this message first. But it was missing so I post again.
I think there is a some bug in build configration files.
Can you check up the following errors and fix it?
The error has no concern with UTF-8 patch.
Environment : CentOS 4.3 x86_64
Sungsoo Kim
----------------------------------
[root at saturn xapian-core-0.9.4_svn6707]# make
...
...
mkdir .libs/libxapian.lax/libqueryparser.a
2011 Jan 26
0
list of errorswhile registering client at asterisk with sipp
Hi every one,
Hello i am doing project on evaluating the sip proxy
performances like asterisk, openims and opensips using the traffic generator
SIPp.
I am using 2 computers of same configuration as SIPp clients one as uac and
other as uas... and one laptop for asterisk server......
UAC:192.168.1.99------------------------>Asterisk
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
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I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G
2010 Feb 20
1
Fax, T38 and NAT
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches to T38
and fax goes through.
Sending a from 0197673581 to 0851711201, no problem as long as i dont
enable T38 on 0197673581.
But, if i enable T38
2010 Jun 08
6
reloading realtime sip peers
Hello,
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
When changing for instance the mailbox-parameter in the realtime
sip_buddies table, the change is not applied with a 'sip reload'.
For every change there is a complete 'reload' necessary.
Why does a 'sip
2006 May 30
4
javaw.exe exits with InvocationTargetException
Running Ubuntu Dapper Drake 6.06
wine version 0.9.14
Trying to install Rosetta Stone Application version 2.1.3A
Calling the Setup.exe on the application disk of the Rosetta Stone
package
results in the InstallAnywhere window popping up and reaching 100%
completion
of the install preparation, then the window goes away, the console
hangs for a few
more seconds, and then the following error and
2008 Mar 17
1
getting a LAX error
I am getting the following error when trying to install an application
that should launch a java application:
I did set my display to the localhost:0.0 but to no avail.
Stack Trace:
java.lang.NoClassDefFoundError
at java.lang.Class.forName0(Native Method)
at java.lang.Class.forName(Unknown Source)
at java.awt.Toolkit$2.run(Unknown Source)
at java.security.AccessController.doPrivileged(Native
2009 Aug 21
1
Help with building sieve
Am building sieve plugin on AIX. This is revision 1095:8010f1eb66b3. I
see the following error when doing the build:
rm -fr .libs/lib90_sieve_plugin.lax^M
mkdir .libs/lib90_sieve_plugin.lax^M
rm -fr .libs/lib90_sieve_plugin.lax/libsieve.a^M
mkdir .libs/lib90_sieve_plugin.lax/libsieve.a^M
(cd .libs/lib90_sieve_plugin.lax/libsieve.a && ar x /usr/sadmin/src/
2009 Feb 24
0
Legacy Cisco ATA186I1
The ATA186I1 is way past end of life,
it was the first version of the ATA186,
but it's still working with only a couple
adjustments to a factory default configuration.
Unlike the brand spanking new Cisco/Linksys SPA2102.
Which has a number outstanding issues with Cisco/Linksys
tech support.
It's taking a lot longer than I was expecting to get
the new? box working.? And I may continue
2011 Jun 09
0
Insert name in SIP registry
Via: SIP/2.0/UDP 10.11.22.161:10000;branch=z9hG4bK-a860600e\x0d\x0a
From: Jian Gao
<sip:8181234567 at my.provider.com>;tag=7e9c4091bfc704bco0\x0d\x0a
To: Jian Gao <sip:8181234567 at my.provider.com>\x0d\x0a
Call-ID: daf96244-769f952c at 10.11.22.161\x0d\x0a
CSeq: 48998 REGISTER\x0d\x0a
Max-Forwards: 70\x0d\x0a
Contact: Jian Gao <sip:8181234567 at