Just in case anyone here hadn't noticed, Cisco is apparently making
7940/7960 SIP 8.2 firmware freely downloadable by anyone:
http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960
username: anonymous
password: your email address
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
I tried to find this in asterisk wiki, but each link I found was broken.
How can I use my Snom 190 or 360 softphone as Intercom ?
bye
Ronald Wiplinger
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
actually i do have a 1, i just removed that line because it was setting
callerid
exten = _.,1,SetCallerID(sniped)
--
~Shaun
Tim Robinson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Shaun
I agree with you - I think your dial plan is the problem.
you are stripping off the
I've just upgrade to the latest head (20843) and I get the following error
.Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new style
cdr_pgsql.so (0x0) loaded RTLD_LOCAL
Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key routine
returned NULL in module
On 17 Apr 2006, at 00:30, Steve Feinstein wrote:
Actually it makes no difference. I tried it in an attempt to get
something to happen.
Thanks,
-Steve
Eric ManxPower Wieling wrote:
What happens if you remove the r option? r is almost NEVER
useful.
Steve Feinstein wrote:
I've been
Hi,
Why does my asterisk keep forking instances at random times everyday?
When I do ps aux, I got this:
asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk
-vvvg -c
asterisk 23558 0.0 5.1 26040 12248 ? S09:57 0:00 asterisk
-vvvg -c
asterisk 29832 0.0 5.1
I had this and no one could really answer it. I only get it 1 of my
systems. I've tried a few things, from removing zaptel watchdog - since
I contacted the telco and they said I had a hung channel, to rebuilding
* with different options. Are you configuring * manually or using a
GUI?
Lee
Try, http://www.snom.com/wiki/index.php/Asterisk_1.2_Firmware_R4
On 17/04/2006, at 4:29 PM, Ronald Wiplinger wrote:
I tried to find this in asterisk wiki, but each link I found was
broken.
How can I use my Snom 190 or 360 softphone as Intercom ?
bye
Ronald Wiplinger
Hi all,
I am noob with asterisk and i am trying to install it on Debian sarge.
I know there is [EMAIL PROTECTED] but i prefere install it on my server wich
is yet running an egroupware tool.
Phones coulg register the server but when i try to call from one to
other (internal call) i get this
Hi Paul,
Thanks for the message!
On Sun, 16 Apr 2006, Paul Hewlett wrote:
[...]
I am curious..
Have you tried disabling CPU1 by setting isolcpus=1 on the kernel
command line ?
This will make the kernel ignore the second CPU - you can then run
I don't know if this only works with multiple cpus but I have HT enabled and
it shows cpu0 and cpu1 .. I tried the first part of this email and still the
kernel boots and shows 2 cpus.. Will this only work with 2 real cpus?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
On 4/11/06, Begumisa Gerald M [EMAIL PROTECTED] wrote:
Again, if the IO-APIC is reporting that the card is on its own IRQ,
it really, truly, honestly *IS* on its own IRQ. The reason that it
is suggested to disable the IO-APIC is that on many low-end systems,
Allow me to comment
On Mon, 17 Apr 2006, stoffell wrote:
Interesting. Now 'why' do they suggest it, is it because older
IO-APIC are 'broken' on some boards? I'm very curious as to 'why',
[...]
Most likely this is why.
Regards,
Gerald
___
--Bandwidth
Hi, sorry to bother again. But I still cannot make it work. I made all
acounts have canreinvite=yes, but found no option in Dial aplication to
make the phones exchange RTP directly between them. Can anyone tell me
wich option should I look at? I am stuck with this (probably simple)
problem
Chris Stenton wrote:
I've just upgrade to the latest head (20843) and I get the following error
.Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new style
cdr_pgsql.so (0x0) loaded RTLD_LOCAL
Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key routine
returned NULL
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote:
Hi, sorry to bother again. But I still cannot make it work. I made all
acounts have canreinvite=yes, but found no option in Dial aplication to
make the phones exchange RTP directly between them. Can anyone tell me
wich option should
Tiago Stein D`Agostini wrote:
Hi, sorry to bother again. But I still cannot make it work. I made all
acounts have canreinvite=yes, but found no option in Dial aplication
to make the phones exchange RTP directly between them. Can anyone
tell me wich option should I look at? I am stuck with
Tiago Stein D`Agostini wrote:
Hi, sorry to bother again. But I still cannot make it work. I made all
acounts have canreinvite=yes, but found no option in Dial aplication to
make the phones exchange RTP directly between them. Can anyone tell me
wich option should I look at? I am stuck with
Hi Emmanuel!
It is very hard to answer such a question without having a dialplan
(extensions.conf)or SIP configuration (sip.conf).
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Emmanuel
LAZARO
Sent: Montag, 17. April 2006 11:57
To:
So, is there any other option that prevents that from happening?
Something that I might have turned on and makes Dial work trough
asterisk? I already even removed asterisk completelyu from system and
reinstalled it to be fresh new... still all RTP goes trough Asterisk
machine. And the server
Hi Ronald!
Please check if the following points are NOT activated.
* is not using direct phone to phone RTP streams if:
-) either of the clients is configured with canreinvite=no
-) the clients cannot agree on a common set of codecs and * needs to
perform codec conversion
-) either of the
Thanks, that was the problem, I had the t option on the Dial
application. Nor that I removed them it works.
Thank you.
Rich Adamson wrote:
Tiago Stein D`Agostini wrote:
Hi, sorry to bother again. But I still cannot make it work. I made
all acounts have canreinvite=yes, but found no
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote:
So, is there any other option that prevents that from happening?
Something that I might have turned on and makes Dial work trough
asterisk? I already even removed asterisk completelyu from system and
reinstalled it to be fresh
I had log entries similar to his, bt a reload solved it -- I still
wonder what happened.
on Monday 04/17/2006 Alex Mosburger([EMAIL PROTECTED]) wrote
Hi Emmanuel!
It is very hard to answer such a question without having a dialplan
(extensions.conf)or SIP configuration (sip.conf).
Remco Barende wrote:
So, to document this, the likelihood of a fax working goes in this
order best to worse:
1. POTS - fax
2. POTS - FXO-TDM400P-FXS - fax
3. T1 - TE410P - channel bank - fax
4. T1 - TE110P - PCI - TE110P - channel bank - fax
5. T1 - TE110P - PCI -
I'd like to start a discussion about Asterisk redundancy. I know this has
been covered in the past, but would like to get an idea of what people are
doing for a production system that must be up all the time.
Assuming a single E1 out.
Here are some of my ideas.
HA Linux between the two asterisk
On 4/17/06, Alex Mosburger [EMAIL PROTECTED] wrote:
-) * needs to listen to DTMF tones during the call (for transfers or any
other features)
Does this mean you cannot do any blind or attended transfer? or only
the # transfer option (asterisk built-in, from features.conf) doesn't
work?
cheers
On Monday 17 April 2006 07:44, Rich Adamson wrote:
I don't believe you will ever get POTS - FXO-TDM400P-to-anything to
work properly due to TDM card limitations. So, move all of those to the
bottom of your list.
I *had* this working.
POTS - TDM400
TDM400 - Real_honest_fax_machine
As I'd
i tried reload but nothing ;)
maybe prob with my dialplan like says alex, i'll try to solve it
remaking my dialplan.
John covici a écrit :
I had log entries similar to his, bt a reload solved it -- I still
wonder what happened.
on Monday 04/17/2006 Alex Mosburger([EMAIL PROTECTED]) wrote
hmm
In sip.conf i have the declaration for 2 phones (i am testing asterisk
installation). No prob with phones identification (i think).
I will take a look at my extension.conf if i hadn't make a mistake (i
don't really understood how it works yet).
Alex Mosburger a écrit :
Hi Emmanuel!
Rich Adamson wrote:
I don't believe you will ever get POTS - FXO-TDM400P-to-anything to
work properly due to TDM card limitations. So, move all of those to
the bottom of your list.
If you pay close attention to those postings from the last two years
in which users say fax works, the
As a matter of curiosity, does anyone know what the E1/T1 interface in
this (redfone) box is ?
Could the box be an embedded linux device with a PCI slot, running linux
and therefore zaptel, and therefore the PCI card could be a Digium or
sangoma card ...
Any clues ? Does anyone have such a
On 17 Apr 2006, at 12:58, Joseph Rothstein wrote:
I'd like to start a discussion about Asterisk redundancy. I know
this has
been covered in the past, but would like to get an idea of what
people are
doing for a production system that must be up all the time.
Assuming a single E1 out.
On Monday 17 April 2006 08:21, Lee Howard wrote:
I and other iaxmodem users can say fax works with analog PSTN
connections. In my case, as well as those others of which I am aware,
an X100P (clone, er winmodem) is being used.
Interesting. Do you have more information about your setup
Lee Howard wrote:
Rich Adamson wrote:
I don't believe you will ever get POTS - FXO-TDM400P-to-anything to
work properly due to TDM card limitations. So, move all of those to
the bottom of your list.
If you pay close attention to those postings from the last two years
in which users say fax
Andrew, the only two patches are the ones you mention here?(spandsp
and iaxmodem)?
no other patches?
Thanks,
On 4/15/06, alist [EMAIL PROTECTED] wrote:
I have complied the latest releases, patched with spandsp and iaxmodem
support. If you upgrade to the provided kernel you will have support
Andrew Kohlsmith wrote:
On Monday 17 April 2006 08:21, Lee Howard wrote:
I and other iaxmodem users can say fax works with analog PSTN
connections. In my case, as well as those others of which I am aware,
an X100P (clone, er winmodem) is being used.
Interesting. Do you have more
Alex Brett wrote:
The problem I have, is that the 'billsec' field in the CDR records, only
starts ticking if I accept the call, so it isn't including the time that
I have answered the call on my mobile, but not actually accepted the
call, which means if I reject the call or whatever, then it
All,I am experiencing an issue where if an agent is logged into the queue, but has their client closed. It appears that when the queue calls the agent, it goes through the macro I have setup for that user and will dump them to voicemail if unavailable. This pulls the call out of the queue, which
Asterisk is a multithreaded system. I have not in mind how many
threads open and where. But ie, if you have enabled pbx_spool.so to
generate calls from files, that module launch its own thread to
monitor the calls directory, MOH launch other thread, every channel
has its own thread, the CLI has
Carlos,
Make sure you are using the telco clock in your zaptel.conf, and use
faxdetect=both in your unicall.conf.
I had several issues before finding a combination of
asterisk+libspandsp+libunicall that worked for me.
Currently they are:
- spandsp-20060205.tar.gz
- libmfcr2-20060205.tar.gz
-
How do you use the agents? Callback or on-hook? If callback you can direct the
calls to another context that doesn't have the fail over to voicemail.
--johann
Kyle Sexton wrote:
All,
I am experiencing an issue where if an agent is logged into the queue,
but has their client closed. It
Polycom IP501
Assembly: 2345-11500-040 Rev: B
Bootrom: 3.1.0.0269
SIP Ver: 1.6.5.0043
I can dial other extensions internally, and can get to voicemail, but
when I try an outside number, I hear dial tone, the digits dialed, yet
nothing happens when I press Send.
Nothing appears on the Asterisk
Johann,I'm using callback for the login method. That definitely makes sense, I'll try it out and see. Thanks!Kyle SextonOn 4/17/06, Johann
[EMAIL PROTECTED] wrote:
How do you use the agents?Callback or on-hook?If callback you can direct thecalls to another context that doesn't have the fail over
Tofik Suleymanov wrote:
When new voicemail comes and i pick up the phone i hear special tones
indicating that the new voicemail arrived.
2. there is no new voicemail (checked mailbox on filesystem), but when i
pick up the phone i hear speial tones indicating that there is a new
message
We
I'd like to start a discussion about Asterisk redundancy. I know this has
been covered in the past, but would like to get an idea of what people are
doing for a production system that must be up all the time.
I'm going to pipe in on this one. Asterisk redundancy is a huge
discussion on this
On Monday 17 April 2006 10:17, Jim Rice wrote:
I can dial other extensions internally, and can get to voicemail, but
when I try an outside number, I hear dial tone, the digits dialed, yet
nothing happens when I press Send.
a sip debug on the asterisk console will give you a ton of data if it's
Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that
On Wed, 12 Apr 2006 [EMAIL PROTECTED] wrote:
Hi
I've got a dell 2550 with an Eicon Diva server PRI card plugged into it.
I can call out using the acopy2 test utility.
I'm having trouble with asterisk making calls however... my capi.conf
and modules.conf looks correct by the wiki
John Rich a écrit :
Hi Folks,
I have posted a couple of message to the list and do see them, even
after waitin for long time (2 days). Can someone please point me to
the rules for posting to this list? I think it had to do with the
subject that I was looking for. I was looking for IAX
Is there any h323 channel driver that supports DTMF inband signalization?
Thank you for your answer!
--
Tomislav
___
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To UNSUBSCRIBE or update options visit:
You could try chan_oh323.so and chan_h323.so. I think also ooh323
supports inband DTMFs.
Regards
Alberto Sagredo
Tomislav Parčina escribió:
Is there any h323 channel driver that supports DTMF inband signalization?
Thank you for your answer!
--
Tomislav
We have the same problem!
And our question was a technical one about Snom 360...
Was the mailing list server down?
I'll go search in my sent items and try to send it again...
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Jean-Michel Hiver
Verzonden:
I can dial other extensions internally, and can get to voicemail, but
when I try an outside number, I hear dial tone, the digits dialed, yet
nothing happens when I press Send.
Nothing appears on the Asterisk CLI screen.
Did the upgrade modify the dialplan setting on your phone? This
Thanks Do you have any suggestion on which news group I should target? Thanks John,Jean-Michel Hiver [EMAIL PROTECTED] wrote: John Rich a écrit : Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me
By now, every Snom fan
should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware
The XML minibrowser is very
cool and opens a lot of possibilities!
One of my ideas is
rich messaging, so you can send fully formatted messages to a Snom
360
Hi, Can someone plesae recommend a good IAX hard phone? And/or IAX ATA? Thanks John
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1/min.___
--Bandwidth and Colocation provided by Easynews.com --
TWV wrote:
By now, every Snom fan should have installed the 6.0 (beta) firmware :-)
See http://www.snom.com/wiki/index.php/Beta_Firmware
I had to revert back to 5.5, because 6.0 kept garbling my LCD screen
(the screen would become unreadable). You might want to wait for 6.0.1 :-)
- Mike
I need
to manually set certain CDR fields.
1).
Callers are allowed to call someone within the same organisation by using a 4
digit extension. A database lookup maps the 4 digit extension to the real
number. However, a CDR for this call shows the original 4 digit extension still.
What
On Mon, 2006-04-17 at 10:45 -0400, Andrew Kohlsmith wrote:
a sip debug on the asterisk console will give you a ton of data if it's
getting to the asterisk box...
*CLI sip debug
... stuff scrolled off screen ...
m=audio 2230 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
I'm sorry to hear that, but I didn't experience such a problem, 6.0 seems to
work quite well on my phone.
Do you have a suggestion for my question?
Or alternative:
Is it possible to send a custom SIP NOTIFY message (with XML body) to an
asterisk sip client?
- Frederic
-Oorspronkelijk
On Monday 17 April 2006 16:03, Moises Silva wrote:
Asterisk is a multithreaded system. I have not in mind how many
threads open and where. But ie, if you have enabled pbx_spool.so to
generate calls from files, that module launch its own thread to
monitor the calls directory, MOH launch other
Rich Adamson wrote:
Chris Stenton wrote:
I've just upgrade to the latest head (20843) and I get the following
error
.Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new
style cdr_pgsql.so (0x0) loaded RTLD_LOCAL
Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key
Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance.
___
--Bandwidth and Colocation provided by Easynews.com --
On Monday 17 April 2006 12:39, Anton Krall wrote:
I don't know if this only works with multiple cpus but I have HT enabled
and it shows cpu0 and cpu1 .. I tried the first part of this email and
still the kernel boots and shows 2 cpus.. Will this only work with 2 real
cpus?
I believe so.
Hi All,
I have a project about IP telephony, we want to build a switch phone using
Asterisk, the amount of users is of 15000 and also it was needed to connect
with other PSTN using SS7 my consult is: asterisk can be used for this
ammount?, Can I to build a switch and using SS7 for example
On Mon, 2006-04-17 at 19:12 +0200, Paul Hewlett wrote:
This is incorrect. Asterisk is a multithreaded system but how the threads
are handled by the OS depends on the version of threads that is being used.
For Linuxthreads (kernel 2.4), one would see a separate entry for each
thread
On Monday 17 April 2006 12:53, Jim Rice wrote:
Looking for 6600546 in office (domain 10.0.0.1)
Reliably Transmitting (no NAT) to 10.0.0.201:5060:
SIP/2.0 484 Address Incomplete
It looks like whatever you're dialing from the IP501, Asterisk doesn't like.
Did the number you called (6600546)
Hi Waldo,
The best I've seen so far is about 100 concurrent calls on a single
Xeon 2.4Ghz. The CPU was 100% but this does not mean anything since
this is due to GSM encoding which happens sequentially and always
leaves capture work in priority. I'm sure it can do more than that,
it's just not
Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg
doesn't elicit any response from the phone using fw 8.2?
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138
Thanks for clarifying that Paul. my output for getconf is:
linuxthreads-0.10
so i guess is normal to have several threads shown by ps axu right?
On 4/17/06, Dave Cotton [EMAIL PROTECTED] wrote:
On Mon, 2006-04-17 at 19:12 +0200, Paul Hewlett wrote:
This is incorrect. Asterisk is a
In Cisco land you would send a command to the phone via a long URL so
the idea was to send a HTTP/POST to
ipaddress/ciscoservices/directory/browser/index.html?url=http://somedomain/services/app.php
this is not exact as it changes often.
I am reading on the snoms I am sure their system is much
http://www.asterisk2billing.org/
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, April 17, 2006 1:13
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Billing
Server Open Source
Any know of any working smart
dovb wrote:
That fix would be great!!!
To press # and be able to get the call back and terminate the transfer...
I had to implement an horrible workaround to emulate this functionality
Well, as it stands now, to hangup while you are doing a transfer, you
using the hangup feature code (in
FYI, this is more of a question for the
asterisk-biz list.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, April 17, 2006 1:13
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Billing
Server Open Source
Any
On Mon, 2006-04-17 at 11:48 -0400, Jonathan k. Creasy wrote:
Did the upgrade modify the dialplan setting on your phone? This sounds
suspiciously like trying to dial a number that is not matched or allowed
by the dialplan.
I'm not sure. I think it the default dialplan:
dialplan
We have four settings for the codec.
How will it be negotiated?
How should it be negotiated in relation to the available bandwidth?
Is there an influence by using canreinvite=yes ?
Phone A has a setting for the priority of codec
Sip.conf has (maybe even different) settings for the priority of
hi all,
i use asterisk-1.2.0-beta and asterisk-oh323-0.7.3 with
openh323_v1_17_1 . when i dial out with H323, it take so long time to
start a 323 call , usually 80 seconds .
i havn't used gk.
i trace the code , and found out it block at
h323_make_call(
this
I (cannot sleep and I) am thinking if there is a way to make inwards
billing easy possible.
To dial out we use something like:
exten =
_9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF})
(I have an extra field TARIFF, what allows me to use different prices
for
On Mon, 2006-04-17 at 13:19 -0400, Andrew Kohlsmith wrote:
Did the number you called (6600546) have a match in the dialplan context it's
entering in to (office)?
-A.
It's supposed to going outbound.
[office] is the internal extensions context.
And no, it wouldn't find a match there.
Jim
I want to thank you for the suggestions. The office is in the UK, so
probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for
the line so that bandwidth should not be a problem, the internal LAN
will be Gbit as said so the QoS as suggested will be only on the
firewall (linux). I
Has anyone heard anything about these guys? Anyone seen anything
like this?
http://www.orative.com/solutions.php
Its seems very cool, basically uses GPRS as a digital
overlay on your mobile phone for additional functionality such as presence and
IM though Im sure they have some other
On Monday 17 April 2006 14:50, Jim Rice wrote:
It's supposed to going outbound.
[office] is the internal extensions context.
And no, it wouldn't find a match there.
How do you have your polycoms set up such that different numbers go to
different parts of the dialplan? Every single
On 4/17/06, Simone [EMAIL PROTECTED] wrote:
look at the wiki and the phones suggested, we'd definitely like phones
with internal ethernet switch and PoE capable, I'll try to get an idea
of what could work for us.
I just have a few suggestions on the phones.. First of all, try using
1 model for
Jeez.
Why does every startup in the universe have to be in the bay area.
:(
-Original Message-From: Dean Collins
[mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:00
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] Orative
The weather isn't as good in Indiana.
Douglas Garstang wrote:
Jeez. Why does every startup in the universe have to be in the bay
area. :(
-Original Message-
*From:* Dean Collins [mailto:[EMAIL PROTECTED]
*Sent:* Monday, April 17, 2006 1:00 PM
*To:* Asterisk Users
Erick,
That's it for now. Let me know offlist if you have any requests...
Andrew
Erick Perez wrote:
Andrew, the only two patches are the ones you mention here?(spandsp
and iaxmodem)?
no other patches?
Thanks,
On 4/15/06, alist [EMAIL PROTECTED] wrote:
I have complied the latest
And they have better sushi than Montana and Wyoming
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew D
Kirch
Sent: Monday, April 17, 2006 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Orative
The
I had the same question, and I want to make sure I'm clear. This
implies to me that Asterisk itself doesn't use SMTP, but rather dumps a
message into some directory that Sendmail on the same box will see and
process? I have no problem getting Sendmail to use a smarthost, but am
I
Ronald Wiplinger wrote:
I (cannot sleep and I) am thinking if there is a way to make inwards
billing easy possible.
To dial out we use something like:
exten =
_9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF})
(I have an extra field TARIFF, what allows me to use
Andrew D Kirch wrote:
The weather isn't as good in Indiana.
Hear hear!!
There had to have been a hundred thousand lightning strikes last night
within a mile of my house. . .
B.
___
--Bandwidth and Colocation provided by Easynews.com --
Easy access to talent.. and lets face it where else
are you going to start up? NY.. I hate it and I have to
live here.
BTW another neat start up I saw at CTIA last week is www.savaje.com Java based mobile OS being sold
to MVNO on other companies hardware, highly customisable and OTA
Armin Schindler wrote:
The configuration is as easy as with BRI lines. Can you provide more (like
your confs and verbose/debug output)?
Also (this isn't directed at you Armin, but I found your email to reply
off of to maintain the threading), I created a Wiki page over at the
freePBX
Southern California would make me happy, maybe the north west.
:)
-Original Message-From: Dean Collins
[mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:49
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users]
Orative
Easy
Hi,
Anyone know how to compile asterisk for a hyperthreaded processor? Thnx
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I resent that, the weather here is wonderful today
On 4/17/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
The weather isn't as good in Indiana.
Douglas Garstang wrote:
Jeez. Why does every startup in the universe have to be in the bay
area. :(
-Original Message-
Wai Wu wrote:
Hi,
Anyone know how to compile asterisk for a hyperthreaded processor? Thnx
Wai,
Asterisk makes heavy use of threads, so all that's required to take
advantage of HyperThreading is an SMP kernel. The kernel itself will
take care of scheduling the threads on the different
I highly recommed against using hyperthreading. It always seems to cause
intermittent kernel panics for me when I forget to turn it off.
--
Justin Tunney
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We've been running Asterisk on P4s with HyperThreading turned on with
an SMP kernel for almost two years now. Currently 12 production
servers, No problems and slightly higher capacity.
MATT---
On 4/17/06, Justin Tunney [EMAIL PROTECTED] wrote:
I highly recommed against using hyperthreading. It
Hi,
like to know which configurations are most suitable for roaming users
accessing from various external environments? As an example, should I
use nat=yes in sip.conf when the end device could be connecting from
behind nat with private ip or with a public ip?
Appreciate any suggestions. Thanks.
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