can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents.
read more about STUN at: http://www.voip-info.org/wiki/view/STUN
and there is a list of public STUN Server.
Regards
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:rai
Am 03.11.2014 um 13:47 schrieb Rainer Piper:
Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens:
First I am new to PBX so i might be doing something fundamentally
wrong...
That being said I got a FreePBX 32bit stable 6.12.65.
I am having some issue with the NAT and sound, both phones
Piper:
Am 02.10.2014 um 15:40 schrieb Tzafrir Cohen:
On Thu, Oct 02, 2014 at 07:52:34AM +0200, Rainer Piper wrote:
Is the destination Number like Country Code +972?
+972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers]
source -http://www.wtng.info/wtng-972-il.html
That page
jail.Attempts stop reasonably quickly.
An empty room with an unlocked door is far less interesting than a
room with the door locked.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rainer
Piper
*Sent:* Friday, October 03, 2014 1
Hi Chris,
yes ... it is boring ...
I stop posting ...
;-)
Am 03.10.2014 um 20:11 schrieb Chris Bagnall:
On 3/10/14 6:52 pm, Rainer Piper wrote:
the attacking server changed the destination Number at 18:53 CEST and
he is still blocked ... LOL
972597438354 callto:00972597438354
It's
:00972595632276
Oct 3 20:26:37 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 142.0.41.179
sipcli/v1.8 rm=INVITE aU=null rU=+972595632276 callto:00972595632276
Am 03.10.2014 um 20:15 schrieb Rainer Piper:
Hi Chris,
yes ... it is boring ...
I stop posting ...
;-)
Am 03.10.2014 um
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
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
sipcli/v1.8 rm=INVITE aU=null rU=00972597613940
callto:00972597613940
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test)
XMPP: rai...@xmpp.soho-piper.de
thing is if I do realtime mysql status
It shows as connected just the timer resets.
Any idea why this is occurring?
Hi Andrew,
what balancing algorithm you use in haproxy.cfg ?
balance source
balance roundrobin
or
balance leastconn
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123
for fail2ban to catch this log:
|NOTICE.* .*: Request from '.*' failed for 'HOST(:[0-9]{1,5})?' (.*) -
No matching endpoint found
Regards|
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161 callto:004922897167161
P2P: sip:rai...@sip.soho-piper.de
Am 15.09.2014 um 15:26 schrieb Matthew Jordan:
On Mon, Sep 15, 2014 at 6:21 AM, Patrick Laimbock
patr...@laimbock.com mailto:patr...@laimbock.com wrote:
Hi Rainer,
On 15-09-14 09:07, Rainer Piper wrote:
Hi,
Info !!! not a question !!!
the pjsip logger
Hi Patrick,
github done ;-)
what is HTH ???
Am 15.09.2014 um 13:21 schrieb Patrick Laimbock:
Hi Rainer,
On 15-09-14 09:07, Rainer Piper wrote:
Hi,
Info !!! not a question !!!
the pjsip logger is different:
[Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request
from
oh ... thanks :-[
Am 15.09.2014 um 17:30 schrieb A J Stiles:
(this is not where your reply belongs)
On Monday 15 Sep 2014, Rainer Piper wrote:
Hi Patrick,
github done ;-)
what is HTH ???
HTH == Hope That Helps.
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
/
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161 callto:004922897167161
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test)
XMPP: rai...@xmpp.soho-piper.de
--
_
-- Bandwidth and Colocation
a tutorial or anything to configure that?
thank you so much
On Thu, Sep 11, 2014 at 3:46 PM, Rainer Piper
rainer.pi...@soho-piper.de mailto:rainer.pi...@soho-piper.de wrote:
Am 11.09.2014 um 10:36 schrieb rafa alfurqan:
Hi,
thank you for your repplied,
As you're on Ubuntu, you
Am 11.09.2014 um 11:28 schrieb Rainer Piper:
Am 11.09.2014 um 11:00 schrieb rafa alfurqan:
Hi Rainer,
are you sure about allowing remote access to phpmyadmin ??? think
about security first !!!
yes i'm sure coz it's not for commercial, just for my research.
I suggest HeidiSQL Client
Am 11.09.2014 um 11:39 schrieb rafa alfurqan:
Hi Rainer,
okay, thanks for your advice.
so i think it would work for freeradius too.
I think so ... freeradius DB is mySQL or oracle.
Cheers
On Thu, Sep 11, 2014 at 4:28 PM, Rainer Piper
rainer.pi...@soho-piper.de mailto:rainer.pi...@soho
/9001-,
PJSIP/9002,20) in new stack
-- Called PJSIP/9002
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/9001-' status is
'CHANUNAVAIL'
--
Thanks,
MMEEGGAA
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123
:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just
look at the cli output *NoOp( 49${gotoadr:-11} )
Am 02.09.2014 um 17:04 schrieb Rainer Piper:
I use in *pjsip.conf
upps and delete the 49 in *Goto(49${gotoadr:-11},1) and *NoOp(
49${gotoadr:-11} )
*just look at the cli output*
Am 02.09.2014 um 17:25 schrieb Rainer Piper:
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11
script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de
mailto:rainer.pi...@soho-piper.de wrote:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
username ?
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth
contact_user in pjsip.conf has to point to the filter or to an agi in
the extentions.conf
like:
pjsip.conf
contact_user=*blablabla
extensions.conf
**exten = blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
*
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
contact_user
menuselect options from
Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only
file to copy?
I am not sure - but I would'nt do that. Make a hardcopy from your
console and transcribe the settings to your new installation. It yould
take you not more than 10 minutes.
--
*Rainer Piper
-logrotate
|
|/code
|
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Am 17.06.2014 17:36, schrieb thufir:
On Tue, 17 Jun 2014 12:14:05 +0200, Rainer Piper wrote:
git clone https://github.com/asterisk/pjproject pjproject
At the very least, thank you for pjsip. I'm not sure what it is yet, but
seems intriguing :)
I'm on Ubunutu 14.04, but will look over your
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. ;-)
wow ... early bird it is 03:36 (PDT) in the morning at your place
Thanks!
Rainer
Am 07.05.2014 12:36, schrieb Joshua Colp:
Rainer Piper wrote:
perhaps a silly
and I get ready for launch in germany at 13:15 ;-)
Am 07.05.2014 13:09, schrieb Joshua Colp:
Rainer Piper wrote:
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. ;-)
wow ... early bird it is 03:36 (PDT
upps ... off topic
and typo lunch not launch ;-)
Am 07.05.2014 13:14, schrieb Rainer Piper:
and I get ready for launch in germany at 13:15 ;-)
Am 07.05.2014 13:09, schrieb Joshua Colp:
Rainer Piper wrote:
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my
,h263,h261
transport=transport-udp
auth=auth7000
aors=7000
direct_media=no
disable_direct_media_on_nat=yes
do I have to turn on the Video Support somewhere else ?
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
is there a way to force asterisk back to do the codec translation?
Attachment:
sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the
B-Leg 7000 NativeFormats: (alaw)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
PS.
if I configure both extension 7000 and 7001 to,
disallow=all
allow=alaw
or
disallow=all
allow=g722
everything is fine. as long as the allowed codec is equal in both
extensions.
Am 07.05.2014 07:00, schrieb Rainer Piper:
Hi!
my asterisk-12.2.0 with pjsip-2.2.0 does not translate
that's funny
I recompiled asterisk without bridge_native_rtp.so
to force asterisk to go to simple_bridge and not to native_bridge...
!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu
Am 07.05.2014 07:11, schrieb Rainer Piper:
PS.
if I configure both extension 7000
[7001]
type=aor
max_contacts=10
qualify_frequency=60
Am 07.05.2014 07:35, schrieb Rainer Piper:
that's funny
I recompiled asterisk without bridge_native_rtp.so
to force asterisk to go to simple_bridge and not to native_bridge...
!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu
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