On Thu, Nov 06, 2008 at 04:43:07PM -0700, Wilton Helm wrote:
The linksys phones annoy me because they cannot implement southern
hemisphere DST properly.
I was shocked the first time I had to write firmware for an
international project. Not only is there the southern hemisphere
issue of
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Your monitoring app is not sending valid IAX2 packets to the server. If
it was sending a true IAX2 POKE, it would be a valid packet and wouldn't
generate this warning.
Could asterisk at
Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP Office
talk to asterisk with SIP Trunking. I had successful calls from asterisk to
Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or the path to resolve
it.
The error I
On 7 Nov 2008, at 08:49, Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these
warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]:
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk configuration files support #include and a number of other
Thanks, I also ported my app to 1.6.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tilghman
Lesher
Enviado el: Friday, November 07, 2008 2:51 AM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] ODBCExec and
Hi!
Is it possible to deactivate RTCP? (I am using 1.6)
thanks
klaus
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Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP Office
talk to asterisk with SIP Trunking. I had successful calls from asterisk to
Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or the path to resolve
it.
The error I get
Hi!
For T.38 Asterisk uses the port defined in udptl.conf. Is there a
workaround (I am using 1.6) for using the same port as RTP also for UDPTL?
regards
klaus
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asterisk-users
I'm pretty sure it is a VERY OLD ISDN router and you can't use it with
*. The voice ports have no VoIP capabilities, they are just used
directly from the ISDN line.
Ronny Julian wrote:
I found this at a local sale. I need to find a power supply for it.
Before I do I wonder if anyone can
Anyone is using 1.6 in production??
Is it ready?
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2008/11/6 Steve Murphy [EMAIL PROTECTED]
On Thu, 2008-11-06 at 13:55 +0100, Olivier wrote:
Yes, you're right : NoOp needs verbosity of 3 and above.
Thanks for helping.
The surprising thing is that AEL Verbose prints output whatever the
verbosity level is (even with 0).
Would you
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B
single port card. When installing the card, the slot on the card doesn't quite
line up with the tab in the PCI-E slot. If I loosen the front plate on the
card, Ican sort of make it plug in, however, the card won't
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
calls another machine
On 7 Nov 2008, at 09:57, Louis-David Mitterrand wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
Your monitoring app is not sending valid IAX2 packets to the
server. If
it was sending a true IAX2 POKE, it would be a valid packet and
wouldn't
generate this warning.
Jerry Geis wrote:
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial this is when it messes up.
when I start entering the number first then click dial this successfull
Jerry Geis wrote:
How do I turn off this overlap dial?
You need to review the dialing rules for the Polycoms.
They'd be located in the ftp directory that you've setup for your
Polycoms to pull their configs from. It's located in the sip.cfg.
Look for the line:
digitmap
Hi folks,
I've been using * for quite a few years and everyday it surprises me more.
I was recently analysing some captures with ethereal/wireshark and found out
that * was doing DNS A queries for domain names like
channel.mydomain.comwhere channel is the typical string of the
dstchannel or
Wolfgang Pichler wrote:
Hi all,
we have the following setup
PSTN 3 PRI Lines --- Asterisk (1.4.22) --- Siemens HiCom
--- Bosch Integral
The Asterisk Machine does play the man in the middle - and adds some
extra functionality to the system (SIP users...) - the normal calls
On Fri, Nov 7, 2008 at 11:20 AM, Brent Davidson [EMAIL PROTECTED]
wrote:
Jerry Geis wrote:
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial
On Fri, Nov 7, 2008 at 11:27 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Jerry Geis wrote:
How do I turn off this overlap dial?
You need to review the dialing rules for the Polycoms.
They'd be located in the ftp directory that you've setup for your
Polycoms to pull their configs from. It's
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
Your monitoring app is not sending valid IAX2 packets to the
server. If
it was sending a true IAX2 POKE, it would be a valid packet and
wouldn't
generate this warning.
Could asterisk at least _not_ report this harmless,
Sebastian Gutierrez wrote:
Anyone is using 1.6 in production??
Is it ready?
I have a number of people using 1.6 in production doing SS7-SIP,
SS7-IAX, and SS7-ISDN gatewaying.
One example (doing SS7-IAX):
System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds
8617029 calls
Steve Totaro wrote:
For two phones, I would just use the web interface.. That is of
course if you plan on keeping a small amount of phones.
Or, if you absolutely hate the web interface :-P
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
So, you don't want any media? No audio, video, just sip packets? If you
just want a sip router with no media look into SER.
Klaus Darilion wrote:
Hi!
Is it possible to deactivate RTCP? (I am using 1.6)
thanks
klaus
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On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED]wrote:
Sebastian Gutierrez wrote:
Anyone is using 1.6 in production??
Is it ready?
I have a number of people using 1.6 in production doing SS7-SIP,
SS7-IAX, and SS7-ISDN gatewaying.
One example (doing SS7-IAX):
Hello,
We've had this problem happen twice with retail customers already and
still have no solution. Basically there are times when customers can't
get any ring at all. It happens that they call our switch and even
though we are receiving ring from the carrier they hear no ring. We have
even put
Steve Totaro wrote:
On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Sebastian Gutierrez wrote:
Anyone is using 1.6 in production??
Is it ready?
I have a number of people using 1.6 in production doing
I've set up an architecture in which OpenSER acts as a registrar and
load balancing server for Asterisk machines. I currently have only one
Asterisk machine serving as a Media Gateway.
My problem is that when A calls B, and then A makes a blind transfer
to C, everything works: REFER goes to
On Nov 7, 2008, at 8:29 AM, samuel wrote:
Hi folks,
I've been using * for quite a few years and everyday it surprises me
more.
I was recently analysing some captures with ethereal/wireshark and
found out that * was doing DNS A queries for domain names like
channel.mydomain.com
What Hardware? For that performance?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew
Fredrickson
Enviado el: Friday, November 07, 2008 3:18 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] 1.6
I think he wants to leave RTP turned on, but turn off RTCP statistics
collection and offers.
Sorry I don't have an answer for the actual question, though. Seems
reasonable, though perhaps selectable on a per-connection basis. Is
RTCP crashing your remote end?
JT
On Nov 7, 2008, at
I am using it at home with FreePBX on a Clarkconnect 4.3 community server. Only
about 10 calls a day but it is doing IAX to SIP. Not had any real problems.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian
Gutierrez
Sent: Friday, November 07,
Jerry Geis wrote:
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial this is when it messes up.
when I start entering the number first then click
Krishna Sumanth Chava wrote:
Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP
Office talk to asterisk with SIP Trunking. I had successful calls from
asterisk to Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or
Not as impressive as matthew's ref but just to add to the picture.
System uptime: 17 weeks, 7 hours, 30 minutes, 51 seconds
342277 calls processed
Asterisk SVN-branch-1.6.0-r117951 built by root @ localhost.localdomain
on a i686 running Linux on 2008-05-22 21:13:46 UTC
using and old Dell 1750
Tim Panton [EMAIL PROTECTED] writes:
I always want to know when I get malformed protocol packets in.
It's easy for an attacker to fill your log drive then.
/Benny
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Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using SIP
(3 trunks now,
The timezone only tells the system with what offset to show
the time when asked for local time.
Sadly some operating systems have this strange concept that changing a
time zone means changing the system clock itself. This makes it a huge
change indeed.
Agreed. The firmware I design works the
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get malformed protocol packets in. It is
always bad news, mostly either a misconfiguration (your
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using
On Fri, Nov 07, 2008 at 12:43:47PM -0700, Wilton Helm wrote:
The timezone only tells the system with what offset to show
the time when asked for local time.
Sadly some operating systems have this strange concept that changing a
time zone means changing the system clock itself. This makes it
At 14:15 11/7/2008, SIP wrote:
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check
back to see if it worked. Would be nice if it did :)
Thanks,
Kurt
On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote:
At 14:15 11/7/2008, SIP wrote:
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind a
That seems to have sort of worked. It seems the phone decided to end
the call this time, instead of Asterisk and now the call is dangling
inside of 'sip show channels'.
So that solution didn't work :(
On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
Ok, recompiling it now
Sebastian Gutierrez wrote:
What Hardware? For that performance?
It's a dual core 1.8 GHz Opteron with 2 TE420P cards and 4 GB of RAM.
Oh yeah, those numbers indicate averaging over 110,000 calls per day
(the ones I posted below) :-)
--
Matthew Fredrickosn
Digium, Inc.
-Mensaje
Sebastian Gutierrez wrote:
What Hardware? For that performance?
It's a dual core 1.8 GHz Opteron with 2 TE420P cards and 4 GB of RAM.
Oh yeah, those numbers indicate averaging over 110,000 calls per day
(the ones I posted below) :-)
--
Matthew Fredrickosn
Digium, Inc.
-Mensaje
Rob Hillis wrote:
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Your monitoring app is not sending valid IAX2 packets to the server. If
it was sending a true IAX2 POKE, it would be a valid packet and wouldn't
generate this
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B
single port card. When installing the card, the slot on the card doesn't quite
line up with the tab in the PCI-E slot. If I loosen the front plate on the
card,
Ican sort of make it plug in, however, the card won't go
Tzafrir Cohen wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get malformed protocol packets in. It is
always bad news, mostly either
To get to the bottom of it I'd recommend determining why the ACKs are
not getting through to Asterisk rather than trying to work around it.
I'm actually suprised Asterisk terminates the call by default when it
doesn't get the ACK to it's 200 Ok response that must be new for
1.4.22 as I haven't
Hi Igor,
We had an interconnect with a carrier that generated early media for
progress indications but the carrier's switch, in this case a Cerpack,
would only start sending the RTP for the early media AFTER it received
an RTP packet from the Asterisk end. Completely stupid behaviour since
early
Thanks a lot Grey. I'll look into it.
Regards,
--
Igor Hernandez
Escape Communications
http://www.escapetel.com
Grey Man wrote:
Hi Igor,
We had an interconnect with a carrier that generated early media for
progress indications but the carrier's switch, in this case a Cerpack,
would only
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get
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