Has anyone figured this out yet?
Lots of places say to add the following
to sip.conf of an Asterisk 1.2 system
(current production machine/Asterisk as root):
tos=0xB8
(Hex B8 = Decimal 184 = Binary 10111000)
or if you are running Asterisk v1.4 or newer:
tos_sip=cs3 ; Sets TOS
Down for me too.
-K
- Original Message -
From: "--[ UxBoD ]--"
To:
Sent: Saturday, February 06, 2010 12:48 AM
Subject: [asterisk-users] Website Down ?
> Hi,
>
> Have I missed something as http://downloads.asterisk.org is not available
> ?
>
> --
> Thanks, Phil
>
> --
>
Hi,
Have I missed something as http://downloads.asterisk.org is not available ?
--
Thanks, Phil
--
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On Fri, Feb 05, 2010 at 01:21:38PM +, Nikhil Nair wrote:
> Hi again,
>
> OK, I've now installed a local caching nameserver, but don't see any
> change at all.
Just to add to the discussion, my setup I was using a local bind9 server
for local/authorative and recursive queries
I think from me
Nice. :-)
Didn't see that, I concede.
- Original Message -
From: "Steve Edwards"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, February 06, 2010 12:10 AM
Subject: Re: [asterisk-users] Dial script
> On Fri, 5 Feb 2010, Karl Fife wrote:
>
>> Try this:
>
> - Original Message -
> From: "Thomas Perron"
> To:
> Sent: Friday, February 05, 2010 8:54 PM
> Subject: [asterisk-users] Dial script
>
> Does anyone have a Dial script or a hint on how I can dial 1
> numbers in sequence?
> When the calls are answered, I play a .gsm or .wav.
> Then,
I think he's referring to the fact that you seem to be looking to put
together the telephone equivalent of a spam service.
I'd be advising rm -rf / as well.
On 02/06/10 16:19, Thomas Perron wrote:
> karl,
> does it make you feel good ?
> wow. pathetic.
>
>
> On Fri, Feb 5, 2010 at 11:00 PM, Kar
On Sat, 6 Feb 2010, Thomas Perron wrote:
karl,
does it make you feel good ?
wow. pathetic.
On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife wrote:
Try this:
#rm -rf /
I second that opinion. Tell us first WHY you want to dial 1 numbers
"in sequence". Without any reason, you must be assum
On Fri, 5 Feb 2010, Karl Fife wrote:
> Try this:
> #rm -rf /
Copycat!
>> On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
>>
>> > Is there any tested script available for this purpose.
>> On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards >
>> Sure. Add this to root's crontab:
>>
karl,
does it make you feel good ?
wow. pathetic.
On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife wrote:
> Try this:
> #rm -rf /
>
> - Original Message -
> From: "Thomas Perron"
> To:
> Sent: Friday, February 05, 2010 8:54 PM
> Subject: [asterisk-users] Dial script
>
>
>> Does anyone have
Try this:
#rm -rf /
- Original Message -
From: "Thomas Perron"
To:
Sent: Friday, February 05, 2010 8:54 PM
Subject: [asterisk-users] Dial script
> Does anyone have a Dial script or a hint on how I can dial 1
> numbers in sequence?
> When the calls are answered, I play a .gsm or .w
On Fri, 5 Feb 2010, Thomas Perron wrote:
> Does anyone have a Dial script or a hint on how I can dial 1
> numbers in sequence?
> When the calls are answered, I play a .gsm or .wav.
> Then, if user presses a defined digit, the call gets bridged to me.
Do you mean the dialed numbers are in sequ
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
--
_
-- Bandwidth and C
Hello everyone
I have a central Avaya S8300 with G450 Gateway, now all calls go
through the Avaya, but I need to record all calls, my questions are:
1- Can I to interconnect Asterisk with Avaya
?
2- With that tool might Asterisk record calls.
I hope your suggestions.
Thanks
Greeting
On 2010-02-05 16:20, Jeff LaCoursiere wrote:
> On Fri, 5 Feb 2010, Mark Willis wrote:
>
> - My limited math capabilities suggest 41 Mbps of RTP traffic, which
> seems like a lot, plus asterisk would be taking a single input stream
> and exploding it out to 500 endpoints.
>
> How did you get tha
Thanks everyone, I'll look at multicast. The customer prefers Snom
phones, luckily.
Mark
On 2010-02-05 16:32, Philipp von Klitzing wrote:
> Hi!
>
>
>> Has anyone done any large scale intercom deployments with Asterisk?
>> I've been asked about building a system to one-way page 500 phones
>>
Hi all,
I'm having a strange issue, wanted to see if anyone had any suggestions.
Due to the recent spike in VoIP related hacking attempts I decided to
tighten security by writing iptables scripts to only allow traffic to my
servers which is white-listed, since then I've had an issue under
certain
Hello my friends,
My asterisk is going down randomly, following you will find some errors that
i could see in the /var/log/asterisk/message at the moment of the crash:
[Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on
transmission 1850202...@10.4.1.152 for seqno 21 (Critical
Hi!
> Has anyone done any large scale intercom deployments with Asterisk?
> I've been asked about building a system to one-way page 500 phones
> simultaneously from a single server.
>
> My concerns are:
>
> - My limited math capabilities suggest 41 Mbps of RTP traffic, which
> seems like a lot
I thought of that too, but the phones will be spread over a large number
of rooms in several buildings, so that won't be too much of an issue.
Mark Willis
On 2010-02-05 15:55, jon pounder wrote:
> Mark Willis wrote:
>
> This could potentially create a very weird audio situation where the
> del
On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis wrote:
> Has anyone done any large scale intercom deployments with Asterisk? I've
> been asked about building a system to one-way page 500 phones
> simultaneously from a single server.
>
> My concerns are:
>
> - My limited math capabilities suggest 41 Mb
On Fri, 5 Feb 2010, Mark Willis wrote:
> Has anyone done any large scale intercom deployments with Asterisk? I've
> been asked about building a system to one-way page 500 phones
> simultaneously from a single server.
>
> My concerns are:
>
> - My limited math capabilities suggest 41 Mbps of RTP
Mark Willis schrieb:
> Has anyone done any large scale intercom deployments with Asterisk? I've
> been asked about building a system to one-way page 500 phones
> simultaneously from a single server.
>
> My concerns are:
>
> - My limited math capabilities suggest 41 Mbps of RTP traffic, which
>
Mark Willis wrote:
This could potentially create a very weird audio situation where the
delay between adjacent phones is audible so instead of acting like
loudspeakers in parallel on a conventional system, it just sounds like a
bunch of people talking at once and is not understandable.
> Has
Has anyone done any large scale intercom deployments with Asterisk? I've
been asked about building a system to one-way page 500 phones
simultaneously from a single server.
My concerns are:
- My limited math capabilities suggest 41 Mbps of RTP traffic, which
seems like a lot, plus asterisk woul
Hi all,
I have been running Asterisk for years (CVS-HEAD on 2005-08-24) with no
problems save a failed harddrive. I have decided to build a new box and have
Asterisk 1.6.2.2 playing nicely with mISDN after lots of changes to dialplan
syntax etc. I am struggling with SIP trunks to sipgate.co.uk and
sean darcy wrote:
> Using 1.6.2.1 with a TDM400, attached to internal analog phones and
> PSTN. When I dial out to PSTN, I cannot send tones, like push "1" for
> something stupid. The call itself works, but the DTMF tones fail.
>
> -- Starting simple switch on 'DAHDI/1-1'
> -- Executing [62
- "ABBAS SHAKEEL" wrote:
> Please some one shed some light on it..
>
Sangoma offers SS7 solutions for Asterisk using their SMG platform:
http://sangoma.com/products/software_building_blocks/ss7_solutions/ss7_signaling_and_media_gateway.html
--Tim
--
Please some one shed some light on it..
On Thu, Feb 4, 2010 at 6:48 PM, ABBAS SHAKEEL
wrote:
> Hello All,
> Please let me know Answers to the following questions .Backgroud.
>
> 1. Which one is better to use libss7 or chan_ss7. Today first time i come
> to know about it ... little bit i googled b
There you go: http://www.canall.com.br/wireshark_trace_t38_ffa.gz
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Bra
>On 5 Feb 2010, at 16:55, Greg Blakely wrote:
>> If so, how?
>NFS or rsync?
>S
Use ODBC voice message storage and realtime voicemail configuration.
- Sean
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.c
On 5 Feb 2010, at 16:55, Greg Blakely wrote:
> If so, how?
NFS or rsync?
S
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
> OK, I've now installed a local caching nameserver, but don't see any
> change at all.
> - Tested name resolution in general: working fine.
> - Turned ADSL router off and tried to make local and Zap calls: no luck.
Did you try to make any calls before pulling the plug on the ADSL router?
dnsmasq
Speaking just for polycom 501's (3 lines), you can set MWI on two different
systems by registering a line to each system; for your question, I'd do it
like this;
Exten 100 is on system B and sends it's VM to exten 110 on System A. You
register line 1 as 100 on system b and line 2 as 110 on Syste
Searching through the archives, I couldn't find an answer for this...
I have two asterisk systems, (system A and system B), and would like to use a
single voicemail system. Phones on system B are SIP phones, registered at
system B.
Can the message-waiting indicator be activated on a SIP phone
Here's the packet trace I promised:
http://www.zshare.net/download/72186098494e6f8c/.
As this is a production system, there were a few calls along with the one that
interests us. The one you're looking for is that from 5433142...@10.150.65.16
to 5421047...@10.153.66.146. The provider has the ad
Could be. Important thing is the problem was solved :)
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54
Yes, the fax machine only transmits at 9600. That's normal and expected. I'll
capture the packets and will provide you with a link to download it in a few
minutes.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+5
On 05/02/10 16:01, Jeff LaCoursiere wrote:
>
> On Fri, 5 Feb 2010, Vinícius Fontes wrote:
>
>> I solved similar issues by setting srvlookup=no, having bind running
>> locally and just the line "nameserver 127.0.0.1" on /etc/resolv.conf.
>>
>
> Your local bind is what solved the problem. The srv
Hello,
What do you get with ipconfig from your clients?
What do you get with nslookup of a client or server?
What do you get with tracert to your server from a client?
Can you access the internet from a client? Are you isolated as a private
network?
Thanks
Larry
-Original Message-
From
On Fri, 5 Feb 2010, Vinícius Fontes wrote:
I solved similar issues by setting srvlookup=no, having bind running
locally and just the line "nameserver 127.0.0.1" on /etc/resolv.conf.
Your local bind is what solved the problem. The srvlookup=no didn't
actually help IMO.
j
Atenciosament
Steve Underwood wrote:
> Spandsp doesn't support those features. I don't know anything which
> does. It seems they can only be used with TCP. Spandsp does support
>
> T38FaxFillBitRemoval
>
> which the FAX for Asterisk package does not (according to Commetrex).
I added indication of T38FaxTran
I solved similar issues by setting srvlookup=no, having bind running locally
and just the line "nameserver 127.0.0.1" on /etc/resolv.conf.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information
On 05/02/10 14:21, Nikhil Nair wrote:
> Hi again,
>
> OK, I've now installed a local caching nameserver, but don't see any
> change at all.
>
> IN detail, what I did:
>
> - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes
> care of dynamic nameserver allocations in /etc/
DND works flawlessly, but whenever using BLF I can only tell that a line is
either in use (on a call) or not. I cannot tell a phone is on DND, or on
hold for that matter. Would be extremely useful.
Would be willing to pay for this developpement if it can be done as long as
the feature makes it
On 02/05/2010 07:40 PM, Vinícius Fontes wrote:
> This message is pointed directly to Steve Underwood. I tought it would not be
> nice to directly email him with a question that interests a good part of the
> Asterisk community, so here it is. :)
>
> Steve, remember a few days ago when we discusse
2010/2/5 Vinícius Fontes :
> Have you tried to set srvlookup=no on your sip.conf?
I think that just stops SRV lookups, not regular DNS.
/r
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asterisk-use
Have you tried to set srvlookup=no on your sip.conf?
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 21
..@macro-user-callerid:18] NoOp("SIP/105-0f643cf0",
"Using CallerID "Pete Siviter" <105>") in new stack
-- Executing [9...@from-internal:2] Set("SIP/105-0f643cf0",
"_NODEST=") in new stack
-- Executing [9...@from-internal:3] M
On Fri, 5 Feb 2010, Nikhil Nair wrote:
> Hi again,
>
> OK, I've now installed a local caching nameserver, but don't see any
> change at all.
>
> IN detail, what I did:
>
> - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes
> care of dynamic nameserver allocations in /etc/re
Hi again,
OK, I've now installed a local caching nameserver, but don't see any
change at all.
IN detail, what I did:
- Installed Debian packages resolvconf and dnsmasq (resolvconf just takes
care of dynamic nameserver allocations in /etc/resolv.conf).
- After looking at the docs, edited /etc/
On 16 January 2010 06:04, Sean Brady wrote:
>>Looking at all the docs I can find Asterisks looks like it should be
>>able to do the job and a whole lot more.
>
>>This is for a small call centre so ideally we want all the features of
>>an average call centre, ACD, Call Recording, Queue's etc etc.
>
This message is pointed directly to Steve Underwood. I tought it would not be
nice to directly email him with a question that interests a good part of the
Asterisk community, so here it is. :)
Steve, remember a few days ago when we discussed about issues on Asterisk
1.6.1.13 and T.38 fax recept
Hi all,
Many thanks for all your very fast and really helpful replies!
Now I know about the asynchronous DNS issue, this all starts to make
sense: presumably, when I disabled eth1 completely, the DNS queries just
failed immediately, so didn't hold anything else up, whereas in the other
scenari
On Fri, Feb 5, 2010 at 9:38 AM, David de Boer wrote:
> I'm currently working on a PHP web interface to show (1) the registered
> endpoints and (2) their status: available, outgoing call or incoming call.
> (In the future, this interface should also be able to redirect calls etc.)
>
> (1) The inte
On Fri, Feb 5, 2010 at 10:39 AM, --[ UxBoD ]-- wrote:
> Doh! :) My philosophy has always been to install a local named server,
> whether it be for Asterisk or something else, as most of the time everything
> I do is behind NAT and I prefer to resolve internal addresses. This also
> help if yo
5 feb 2010 kl. 10.36 skrev Philipp von Klitzing:
> Hi!
>
OpenVPN by default uses UDP, but can be configured to use TCP.
>>
>> So what's the configuration on the Snom? Can I change it?
>
> Google is your friend:
> http://wiki.snom.com/Networking/VPN
>
So what you're saying is that you hav
5 feb 2010 kl. 10.37 skrev Randy R:
>>> Why not run a internal DNS with forwarders to your ISP ? That way Asterisk
>>> can still resolve itself and hosts internally.
>>>
>> See above:
you need a local
resolver, like a caching BIND server, on the same host.
>
> Nice, but still, it rui
- "Randy R" wrote:
> Nice, but still, it ruins the "all in one" concept. Isn't there a
> lighter solution?
Nice and lite DNS server ?
http://www.nlnetlabs.nl/projects/nsd/
--
Thanks, Phil
--
_
-- Bandwidth and Colocation
- "Olle E. Johansson" wrote:
> 5 feb 2010 kl. 09.28 skrev --[ UxBoD ]--:
>
> > - "Randy R" wrote:
> >
> >> On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson
> >> wrote:
> > What I have seen on my asterisk box when I had a up/down adsl
> line
> >> was
> > that the asterisk box
>> Why not run a internal DNS with forwarders to your ISP ? That way Asterisk
>> can still resolve itself and hosts internally.
>>
> See above:
>>> you need a local
>>> resolver, like a caching BIND server, on the same host.
Nice, but still, it ruins the "all in one" concept. Isn't there a
lighte
Hi!
> >> OpenVPN by default uses UDP, but can be configured to use TCP.
>
> So what's the configuration on the Snom? Can I change it?
Google is your friend:
http://wiki.snom.com/Networking/VPN
Philipp
--
_
-- Bandwidth and C
5 feb 2010 kl. 09.28 skrev --[ UxBoD ]--:
> - "Randy R" wrote:
>
>> On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson
>> wrote:
> What I have seen on my asterisk box when I had a up/down adsl line
>> was
> that the asterisk box couldn't do dns resolution and would hang(
>> well no
- "Randy R" wrote:
> On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson
> wrote:
> >>> What I have seen on my asterisk box when I had a up/down adsl line
> was
> >>> that the asterisk box couldn't do dns resolution and would hang(
> well no
> >>> other internal calls could be made, seemed lik
I'm currently working on a PHP web interface to show (1) the registered
endpoints and (2) their status: available, outgoing call or incoming call. (In
the future, this interface should also be able to redirect calls etc.)
(1) The interface already shows a list of all registered endpoints. For th
On 05/02/10 09:15, Randy R wrote:
> On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson wrote:
What I have seen on my asterisk box when I had a up/down adsl line was
that the asterisk box couldn't do dns resolution and would hang( well no
other internal calls could be made, seemed lik
On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson wrote:
>>> What I have seen on my asterisk box when I had a up/down adsl line was
>>> that the asterisk box couldn't do dns resolution and would hang( well no
>>> other internal calls could be made, seemed like some sort of semaphore
>>> was stuck)
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