On Sat, 10 Apr 2010 22:34:28 +0100 (BST), Gordon Henderson
wrote:
>
>Just a "heads-up" ... my home asterisk server is being flooded by someone
>from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it -
>they're trying to send SIP subscribes to one account - and they're
>floodi
out* of india.
On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce wrote:
> There you go. This confirms that SIP signaling determines where the calls
> should go. I would take their word with a grain of salt specially with their
> whole support center our of India. No disrespect, but it is bad service
There you go. This confirms that SIP signaling determines where the calls
should go. I would take their word with a grain of salt specially with their
whole support center our of India. No disrespect, but it is bad service
overall.
-Bruce
On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp wrote:
> --
Sorry, the parameter should be.
srvlookup=yes
Alyed
2010/4/10 Alyed
> Daniel, you are having a problem often seen in pre 1.4.14 versions.
>
> Before this release srvlookup=no was the default for sip.conf and guess
> the same for iax.conf . So if you are working with a previous release just
> a
Daniel, you are having a problem often seen in pre 1.4.14 versions.
Before this release srvlookup=no was the default for sip.conf and guess the
same for iax.conf . So if you are working with a previous release just add
this parameter .. but change it to
serverlookup=yes
under your iax.conf [
On 10/04/2010 9:24 PM, Timothy C Litwiller wrote:
> I have a did with 20 channels from didforsale. that we use to let local
> members call to listen to a conference several times a week without long
> distance charges.
>
> The upcoming call is getting more interest than usual and from places
> that
I have a did with 20 channels from didforsale. that we use to let local
members call to listen to a conference several times a week without long
distance charges.
The upcoming call is getting more interest than usual and from places
that are not local so we want to use a free conference service
On Sat, Apr 10, 2010 at 9:50 AM, James Lamanna wrote:
> Hi,
> Has anyone had any experience using DRBD to mirror an entire asterisk machine?
Entire, no. Specific/Important mounts yes.
> If so, is there a performance issue at all when people are recording
> voicemails and the like?
I haven't se
On Tue, Mar 23, 2010 at 9:06 PM, Steve Moran wrote:
> I am running Asterisk and using Answer machine detection with call files on
> a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD
> is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over
> 50,000 outb
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm trying to test with a friend who has an Asterisk in his office with
the Asterisk which I have in my house. Then I have an extension that he
is trying to register remotely.
Trying with the Twinkle client, I see that it is registered:
- --
- "Tarek Sawah" wrote:
> we started with them two days ago .. and we are facing plenty of False
> Answer cases on several destinations although ppl said they have a
> policy against FAS..
> anyway i don't know i will be looking into another method to send the
> RTP to another server,
The IP
we started with them two days ago .. and we are facing plenty of False Answer
cases on several destinations although ppl said they have a policy against FAS..
anyway i don't know i will be looking into another method to send the RTP to
another server,
thanks for the info
Hi people,
I have a Cisco ATA 186 which understands only the SCCP protocoll,
therefore I am a pure beginner and I hope that anybody of you could help
me.
How will I configure the ATA which has 2 analog ports?
For any support I would kindly thank you
Tamer
--
___
Oh, I see. I haven't done a lot of testing on this new IP since the change
of gateways happened but I did Canada calls and they go fine. However, this
exact provider lies down to their teeth when it comes to problems of call
quality and calls not routing. They never accept faults. They even have
pr
Its a good idea tos setup Fail2ban, instructions for which are on
voip-info.org. It at least blocks such IP addresses, hopefully prompting the
attackers to move their attack somewhere else and leave you alone.
Another good idea is to lookup in whois database this IP address and see if
you can find
Just a "heads-up" ... my home asterisk server is being flooded by someone
from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it -
they're trying to send SIP subscribes to one account - and they're
flooding the requests in - it's averaging some 600Kbits/sec of incoming
UDP da
you got the name EXACTLY!
i already am doing what you suggest but facing problems with some destinations
and they claim that the problem is with my Asterisk server not their routes!
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562
LOLthis is just what I was facing 4 days ago. Unfortunately, Asterisk
doesn't provide a software feature in Zaptel to do a BUSY. But people on the
list suggest that one should call the telephone company and ask them to busy
it.
If you have the resource and don't mind the bill of calling the ba
Say, I'm looking for a simple way to dial a number repeatedly for two minutes
at a time. The purpose is to busy up a faulty analogue line in an incoming hunt
group. Tx
Shaun--
_
-- Bandwidth and Colocation Provided by http://ww
Just a week ago, I have been in the same situation. Provider was changing
from Cisco gateways to I think Nextone and hence provided me many IPs.
I found out that the media IPs don't matter and just played around with my
NAT settings and all calls can go through just fine by using simply:
host=111
Hi Guys,
I am calling out 416-999- on Channel 1 of PRI and then calling
416-999- on Channel 2 of PRI. When the two channels are going to be ZAP
native bridged, both channels hangup and CLI show PRI cause (16).
Asterisk Verbose *(Channel 1 already connected to party)*:
-- Requested tra
Don't have a system to test this right now, but read somewhere this was a 2
steps solution:
1) Leave the callerid in your tunk definition blank (in your example the 999
username)
2) Use brakets around the callerid definition of your peers: callerid= <200>
(extension 200 in your example)
Let us k
Hi,
Has anyone had any experience using DRBD to mirror an entire asterisk machine?
If so, is there a performance issue at all when people are recording
voicemails and the like?
It seems like that could generate quite a bit of traffic. Also, do you
bother to mirror the log files as well?
Thanks.
-
Hi Jerry,
On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis wrote:
> I am looking for something in asterisk that
> will let me record a wav file in asterisk (which I know how to do)
> then some other command (external or dialplan) that would read
> the wave file and tell me if a certain tone or frequen
On Sat, Apr 10, 2010 at 8:50 AM, Tarek Sawah wrote:
>
> Greetings list
> i'm trying to connect with a VoIP provider for termination.. and they have
> offered us three servers to connect with
> one SIP Signaling server and Two Media servers ..
> googled for a week and didn't find a way to do this.
Greetings list
i'm trying to connect with a VoIP provider for termination.. and they have
offered us three servers to connect with
one SIP Signaling server and Two Media servers ..
googled for a week and didn't find a way to do this.. so my question. is it
possible to be done?
Asterisk server
On Sat, Apr 10, 2010 at 6:35 AM, Adrian Marsh
wrote:
> Hi All,
>
>
>
> I’ve two asterisk servers on the same LAN, both 1.4, and I keep getting “Got
> SIP response 489 "Bad event" back from 192.168.3.10”
>
> No idea whats causing it. The only references I can find mentions NATing
> issues, but thes
Hi All,
I've two asterisk servers on the same LAN, both 1.4, and I keep getting
"Got SIP response 489 "Bad event" back from 192.168.3.10"
No idea whats causing it. The only references I can find mentions NATing
issues, but these are on the same LAN so NAT shouldn't be an issue.
3.10 does auth
Hi!
On 04/10/2010 02:04 PM, Tzafrir Cohen wrote:
> On Sat, Apr 10, 2010 at 12:32:49PM +0200, Darshaka Pathirana wrote:
>>
>> We have a problem here... Hope somebody can give us some hints.
>>
>> We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem.
>> Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+
On Fri, Apr 09, 2010 at 11:27:34AM -0400, Martin wrote:
> Are you sure writing to the right list???
Thanks for helping that mail defeat my spam filter. Please don't help
spam by quoting it.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
Hi everyone.
We have a problem here... Hope somebody can give us some hints.
We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem.
Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+lenny1) with zaptel (1.4.11) and
libpri (1.4.3) is installed.
There is a QuadBRI-Card installed:
# lspci -vv -s 06:04.
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