AB == Alan Bunch [EMAIL PROTECTED] writes:
AB Just another OpenVPN data point, and not Asterisk related but here
AB goes. I run 15 users over a DSL link on one end and a Internet T1
AB on the other with OpenVPN and it just rocks. The road warrior
AB setup is down to running one script to create
AF == Anthony Francis [EMAIL PROTECTED] writes:
AF I knew that was true about GSM networks outside of the US, but to
AF be honest, I am not concerned with those networks ^^.
On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote:
Mindfully wanting to use a + instead of knowing the international
CB == Chris Bagnall [EMAIL PROTECTED] writes:
CB We have a few FreePBX setups running in virtual machines in
CB environments where the client wants their own PBX (and web
CB interface to play with) without wanting to pay full whack for the
CB server plus hosting, etc. However, we haven't scaled
JM == Jeremy Mann [EMAIL PROTECTED] writes:
I would have answered, but I was prohibited from quoting properly:
JM If you are the intended recipient, further disclosures are
JM prohibited without proper authorization.
/Benny
___
Sign up now for
I answered because I was hoping for a repost without the licence,
perhaps through gmail. Would you have been happier not knowing that
you were missing out on something?
/Benny
___
Sign up now for AstriCon 2007! September 25-28th.
PvK == Philipp von Klitzing [EMAIL PROTECTED] writes:
PvK Some of the bigger MFC printer/copy/fax combo devices by Brother
PvK (and maybe also other vendors?) provide a fax-via-smtp feature
PvK and can built fax networks that way.
As far as I can tell, the Brother boxes require the user to
JH == John Hughes [EMAIL PROTECTED] writes:
JH And why on earth don't all the HP MFC's that have Ethernet and or
JH Wireless and a Fax modem also have T.38 built in? I've got 2 of
JH the damn things and the fax capability is useless to me.
T.38 would require actual code. T.37 is already
SB == BerkHolz, Steven [EMAIL PROTECTED] writes:
SB [..]
SB This way I can test different versions of the features of Server2
SB (clone with different IP) without affecting production. I assume
SB that I just use an IAX or SIP trunk between the two asterisk
SB servers.
SB Does this make sense?
P == Patrick [EMAIL PROTECTED] writes:
P There is a Xen page called something like cool configurations. It
P has information how you can configure access to a PCI card. Iirc it
P is even possible to assign one PCI slot/card to one virtual client
P and another PCI slot to another virtual client.
RB == Remco Barendse [EMAIL PROTECTED] writes:
RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware
RB connected to Asterisk?
Yes.
RB Any experiences / caveats?
Make sure you keep the firmware updated. It improves rapidly.
RB If anyone would be willing to share the dump of their
DL == Doug Lytle [EMAIL PROTECTED] writes:
DL Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to
the exten value, but how do I access the from number '248xxx'?
DL exten = s,1,GotoIf($[${CALLERID(number)} = 248xxx ]?2:3)
That works, of course,
PB == Phil Blundell [EMAIL PROTECTED] writes:
PB Right now I'm still using their Java thing, but it's slow enough
PB that one of these days I guess I'll crack and reimplement that
PB stuff directly in python. I think the algorithm is described on
PB the voip-info.org wiki someplace.
A trick
Tilghman Lesher [EMAIL PROTECTED] writes:
On Sunday 02 March 2008 05:33:49 Vieri wrote:
Is it possible to override the standard DB function in
Asterisk?
No.
Is it permitted to modify the astdb outside Asterisk, while Asterisk
is running? It is a SQLite file, right?
/Benny
Michiel van Baak [EMAIL PROTECTED] writes:
On 15:32, Wed 12 Mar 08, Joshua Wilson wrote:
I don't believe it supports multi-tenant as of yet. It could be requested I
am sure.
I created a new VM and installed it.
Guess what, no multi tenant support.
Too bad all them good GUI tools never
Robert Lister [EMAIL PROTECTED] writes:
So you either need to go a Goto(context,4000,1) or to drop it to the queue
with Queue(console) etc.
There's also Dial(Local/[EMAIL PROTECTED]). Goto is almost always a better
idea though.
/Benny
___
--
Steve Totaro [EMAIL PROTECTED] writes:
I will probably continue this train of thought (1.2.X is more
production ready) until these threads stop popping up on the list.
I think you're being too kind to 1.2.x. It has numerous problems, most
especially with locking in chan_sip. 1.4.x is a HUGE
Matt Florell [EMAIL PROTECTED] writes:
But seriously, several of my clients use SIP exclusively, passing tens
of thousand of calls a day on Asterisk 1.2.X with no issues. I have
noticed that the load is slightly lower for SIP-only in 1.4, but I
have not noticed any stability issues revolving
Vieri [EMAIL PROTECTED] writes:
Should I interpret the above that it's in an infinite
loop trying to dial/reach SIP/4053?
They are just stuck channels. It's simply a bug in 1.2.x. Fortunately
it's fixed in 1.4.x. We upgrade our customers to 1.4.x when they hit
that bug.
/Benny
I am willing to try out an image.
/Benny
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To UNSUBSCRIBE or update options visit:
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Benny Amorsen [EMAIL PROTECTED] writes:
I am willing to try out an image.
Yes this was intended to be private mail. Alas, the Asterisk lists
suffer from broken Reply-To, and I fell for it. Sorry.
/Benny
___
-- Bandwidth and Colocation Provided
Tilghman Lesher [EMAIL PROTECTED] writes:
Oh, please, don't start THAT flame war. People who consider Reply-To
munging harmful have obviously failed to read the ENTIRE rfc (especially
the part where it specifically says the use of the Reply-To header is for use
on listservs). :-P
It's
Tilghman Lesher [EMAIL PROTECTED] writes:
And the arguments on the other side come down to I'm using an ISP
which can't correctly configure their mailserver, and I'm too lazy to set one
up myself.
How can the mail server fix a broken reply-to? It can remove it of
course, but that is rather
Tilghman Lesher [EMAIL PROTECTED] writes:
Which is why 'rm' ALWAYS prompts for confirmation, right? Oh, wait, it
doesn't. ;-)
Confirmation is useless, it just teaches everyone to blindly type y.
Analogies suck, of course.
/Benny
___
--
Steve Totaro [EMAIL PROTECTED] writes:
My dual proc, dual core AMD boxen show as four procs. I guess the AMD
architecture uses Hypertheading (or whatever the equivalent is for
AMD, I assume Intel owns the rights to the name Hyperthreading).
I think the more likely explanation is that two
Steve Totaro [EMAIL PROTECTED] writes:
But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2.
Or is C2D not four cores?
D is for duo.
/Benny
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
OCG Technical Support [EMAIL PROTECTED] writes:
Anyone tried Asterisk with Fedora 9 (recent release)?
I upgraded my home server with Asterisk to Fedora 9 the day before
yesterday. I use the Asterisk version that comes with Fedora 9.
/Benny
___
--
Lee Howard [EMAIL PROTECTED] writes:
Note that if you have a fax machine that performs some variant of T.37
(fax-over-email) and you have an on-line service provider that is
willing to work with you... then you can rather easily get your fax
machine faxing through their service. (Which is
Steve Totaro [EMAIL PROTECTED] writes:
Then check out Hylafax and IAXmodem. Hylafax has alot of client apps
too. As I said before, it is CPU intensive, so you may need separate
machines to handle fax. A direct crossover cable for network is the
best to eliminate any latency.
If you run
Gordon Henderson [EMAIL PROTECTED] writes:
RxFAX gets one step closer to the data stream (no copper/IP in the
way), so why are people using ( suggesting) Hylafax over RxFAX?
When we piloted our deployment, rxfax didn't manage to receive faxes
correctly as often as Hylafax did. We never found
Sherwood McGowan [EMAIL PROTECTED] writes:
He's right, I have a client who is running 1.2 but wishes to upgrade and
it's going to be a pretty large undertaking. If nothing else, look at
the change logs for 1.4 AND 1.6 and then decide if you're in need of an
upgrade.
In practice, upgrades
Lyndon Griffin [EMAIL PROTECTED] writes:
I will believe it's a code problem - I see that the phones are picking
up *some* of the attributes I pass in DHCP offers, like the domain
name. Not only that, but I've sniffed a phone actually trying to ARP
the address the server DHCPOFFERED to it,
Steve Totaro [EMAIL PROTECTED] writes:
I have heard many people (and even a good friend of mine and the
Asterisk community) feel ill immediately when around active DECT.
He can make a lot of money if he can demonstrate that.
/Benny
___
--
Tilghman Lesher [EMAIL PROTECTED] writes:
One very big benefit of using a database with cron jobs is that your web
application does not need to run as the same user (or otherwise weaken
security permissions) as the Asterisk daemon. If running as the same user,
you'd have to either set both
WideVOIP [EMAIL PROTECTED] writes:
Junghanns have PCI cards with GSM modules and the drivers
it works great
They have the classic problem with interrupt sharing. At least the
card we bought does. Welcome to the 21st century.
/Benny
___
--
Please don't top post.
WideVOIP [EMAIL PROTECTED] writes:
Never had this problem over HP or SuperMicro servers
If the card happens to not share interrupts, all is well.
/Benny
___
-- Bandwidth and Colocation Provided by
Michael Graves [EMAIL PROTECTED] writes:
No, TCP for media as well. I though that was the whole point of SIP
over TCP.
Hopefully not. RTP over TCP would be entirely pointless. RTP needs
packetization, doesn't mind packet loss (within reason) but hates
retransmissions. TCP doesn't provide
Coco Richard [EMAIL PROTECTED] writes:
Hi all,
How can i use different VLANs for signaling and audio, e.g vlan 100 for
sip and vlan 200 for rtp? Where can i find documentations for this?
That would be very difficult. You can do it in Linux with firewall
rules and policy routing based on
Tilghman Lesher [EMAIL PROTECTED] writes:
I'm not terribly sure that the PCI bus will stand up to that many interrupts
per second, though it's certainly possible.
The PCI bus should be rather bored with 2Mbps per card. Only one card
should interrupt, but I am not sure whether Sangoma or Digium
D.J.Sateesh [EMAIL PROTECTED] writes:
hi,
i am recieving deadlocks frequently and its calls are getting hanged .
Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for
'0xb6692258', 10 retries!
We upgrade customers who hit that bug to 1.4... The locking is greatly
improved.
Russell Bryant [EMAIL PROTECTED] writes:
That is actually more of a debug message, and is not necessarily an
indication of a problem.
It has in 95% of cases (and believe me, we have hit it a lot)
indicated that some SIP conversation has deadlocked (it seems that
it's often a registration
Philippe Sultan [EMAIL PROTECTED] writes:
Well, if someone steals the md5secret (HA1) for a given username and
realm, he can use it to authenticate to the SIP proxy or B2BUA that
serves the target user.
This is unavoidable with password-based systems.
Either you transfer the password
ronald [EMAIL PROTECTED] writes:
Is it possible to assign a plus sign on the callerid(num) ?
Yes.
currently this is what i do CALLERID(num)=+6523450017
but telco is denying calls, coz they said they are seeing bs523450017
instead of +6523450017.
Which techology? SIP? PRI? POTS? ...?
Alex Balashov [EMAIL PROTECTED] writes:
Yes, indeed. Encapsulation protocols such as IPSec/GRE won't work at
all over high RTT latency (= 400 ms).
Sure they will. They just won't benefit from TCP acceleration
performed by the satellite company. Sadly TCP itself is very slow with
high RTT, so
Anthony Francis [EMAIL PROTECTED] writes:
Lets not forget that the DECT specification does allow for data
transmission. THere is no reason that in the future you would not be
able to have integrated services over DECT.
The DECT data rate is way too low for integrated services.
/Benny
Michael Graves [EMAIL PROTECTED] writes:
DECT was designed from the start to handle voice and data.
553kbps isn't particularly useful today. Even 3G is faster.
Not that I have ever seen products offering more than ISDN speeds, and
I believe that was before 2000. DECT is excellent, but it
Rob Hillis [EMAIL PROTECTED] writes:
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good reason.
Tell that to Google.
So far, for us, 1.6 beta is running better than any of the early 1.2
releases. Perhaps even better than
[EMAIL PROTECTED] (Tony Mountifield) writes:
You mean like ONE extra message a month? For 50 messages a day, that
is approximately a 0.066% increase!
If you normally read this list via gmane and only keep a subscription
to be able to post, it's an infinite increase. Yes, I could write a
rule
Pascal Bruno [EMAIL PROTECTED] writes:
Ok very good, how about for the asterisk addonds and sounds? Can you
provide me the commands to get, build and install for the 1.4.21 version?
Thanks a lot guys.
If you can't figure that out on your own, you really should stick with
the
Stefan Gofferje [EMAIL PROTECTED] writes:
Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX
and the FIXUP SIP of the PIX makes it very easy for me to use my * as
server for external clients as well as as client for SIP providers.
A lot of phones come preconfigured for STUN
Dr. Michael J. Chudobiak [EMAIL PROTECTED] writes:
With Asterisk 1.4 I could use commands like:
/usr/sbin/asterisk -rx sip notify reboot-snom mjc_home
to reboot a snom phone. Now, with 1.6, when I try that, I get:
Unable to find notify type 'reboot-snom'
Command 'sip notify reboot-snom
Brendan Martens [EMAIL PROTECTED] writes:
On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote:
The answer you are looking for is that you should be using a
supported,
stable version, and right now, 1.4 is the only one that fits. If I
were
starting today, I'd go with 1.4.
1.6.0 has
SIP [EMAIL PROTECTED] writes:
The truth is there are plenty of email clients that CAN decode
Hotmail messages, and if you choose one that can't, you can't blame
anyone but yourself.
The truth is that there are no Netcom^WAOL^WHotmail users who write
anything worth reading. I had just
Tilghman Lesher [EMAIL PROTECTED] writes:
exten = [0-9*#+].,...
If that does not work, that is a bug and needs to be reported as such.
Sadly that matches *james and 9foo...
It would be nice if you could use normal regexes (e.g. with the pcre
library) in extensions.conf.
/Benny
Brent Davidson [EMAIL PROTECTED] writes:
I have several branch offices, each with their own Asterisk server
(version 1.4.22.1) handling their PBX functions. All of these offices
need to talk to each other. In sip.conf I created a peer entry for each
office with a username of branch-user
Steve Murphy [EMAIL PROTECTED] writes:
Other than the above, we could invent a slightly different syntax for
pcre type expressions; and you'd have to invent some sort of
disambiguation
for when multiple extensions might be matched, to choose the 'best' one.
I'd just use strict ordering from
Karl Fife [EMAIL PROTECTED] writes:
is in fact simply something like:
exten = _[0-9*#+]X.,1,NoOp(*** match ***)
As long as you're happy to match *9foo and not match **123, then yes,
that will work.
/Benny
___
-- Bandwidth and Colocation Provided
Gordon Henderson [EMAIL PROTECTED] writes:
Hm. Drayteks are on my list of modems to turn any SIP ALG off on! You must
have a goodun :)
Drayteks do indeed mess with SIP packets. If you keep STUN/ICE off on
the phone and let Draytek mangle the packets, there is a chance that
things will work.
Steve Underwood [EMAIL PROTECTED] writes:
That list rather poorly supports your argument. The PAP2 and the PAP2T
do *not* support T.38, despite numerous arguments you'll find to the
contrary. Personally I believe Linksys, the manual, and the menus.
The manuals and the menus for PAP2T talk
Daniel Hazelbaker [EMAIL PROTECTED] writes:
I can answer both of those with a single point. We just switched
(entirely) to Asterisk a few weeks ago. We looked, very briefly, at
various ways to get rid of the physical, analog, fax machines. They all
ended with the answer People can't figure
Steve Underwood [EMAIL PROTECTED] writes:
Even the big floor standing office MFPs typically only offer T.37 or
T.38 only through an expensive option card.
Medium MFP's almost all support T.37. They call it scan to email,
but they do it (as far as I can tell) in a way that is compliant with
Steve Underwood [EMAIL PROTECTED] writes:
A number of people say this, but when you ask them to point out the
exact location in the menus, they can't find it.
You are right. I am very sorry for contradicting you.
I got fooled enough to even put FAX_Enable_T38 No /FAX_Enable_T38
in when
Lincoln King-Cliby [EMAIL PROTECTED] writes:
Periodically I'm seeing calls placed from the 7961s through anything
on the PBX that requires digit entry (the Auto Attendant, Voicemail,
etc.) 'randomly' drop; extension-to-extension calls
extension-to-PSTN, and PSTN-to-extension calls never have
SU == Steve Underwood [EMAIL PROTECTED] writes:
SU G.729 isn't the best. Its just the one you need to be compatible
SU with the other end. G.729 is the lock-in choice, not the quality
SU choice.
What is the best codec with asterisk on a slightly lossy link (0.1%
packet loss), if bandwidth is
SB == Stephen Bosch [EMAIL PROTECTED] writes:
SB And it will mean that calls answered by SIP/line1 will roll over
SB to SIP/line2 after the caller hangs up, so you'll get a lot of
SB nuisance rings.
That has not been my experience. When either party hangs up, the call
goes to the h extension,
SIP == SIP [EMAIL PROTECTED] writes:
SIP Premium Rate Services think like 900 and 976 numbers in the
SIP US, but not every country allocates a particular block of numbers
SIP or prefixes to its premium rate services, so with some, they're
SIP pretty close to impossible to block.
Perhaps
CSB == CSB [EMAIL PROTECTED] writes:
CSB But I want to be a bit more selective: tcpdump -C 100 -W 10 -w
CSB /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060
= redirects stdout to a file named =. Possibly not what you want.
/Benny
___
--Bandwidth
KM == Knud Müller [EMAIL PROTECTED] writes:
KM Hi, there are some interesting figures on
KM http://www.thrallingpenguin.com/articles/asterisk-solaris.htm.
It's hard to take them as more than a lower bound on that particular
hardware. No attempt is made at figuring out what actually limits
NM == Noah Miller [EMAIL PROTECTED] writes:
NM If it helps at all, I read a study that said that SSL VPN's can
NM actually help with jitter problems. So it might be preferable to
NM implement something with OpenVPN (uses SSL) rather than an
NM IPSec-based VPN. I found the link:
Only if you use
MB == Matt Brown [EMAIL PROTECTED] writes:
MB Does anyone have any experience with a GSM card, preferably Quad
MB Span (4 GSM modules or higher) for use in the UK. I have seen the
MB Junghanns* version but I am not keen on the limitation of having
MB to use a BriStuffed version of Asterisk.
The
SD == Stephen Davies [EMAIL PROTECTED] writes:
SD Hi, I want to quickly mention that I've had great success with
SD running Asterisk in the under-appreciated Linux-VServer
SD environment.
I just want to do an AOL here: me too! Linux-vserver is great for
asterisk, although we will probably
MR == Matthew Rubenstein [EMAIL PROTECTED] writes:
MR And if you've got GigE installed, not 10/100Mb, and your LAN
MR doesn't have a switch that can handle a phone's lower bitrate
MR without bringing down the whole LAN's rate.
I bet you can't find a switch which acts that way. It existed
RS == Ryan Stille [EMAIL PROTECTED] writes:
RS I am a new trixbox user. One of the things I'd like to get working
RS is being able to tell if a user is calling me by directly dialing
RS my extension, or if they pressed 1 for sales. When they press 1,
RS it rings a group of phones, and the call
KW == Kevin Withnall [EMAIL PROTECTED] writes:
KW Using trixbox (or a custom dialplan if needed) has anyone been
KW able to convert a number dialled like +61242110 to something
KW like 02422110 ie (remove the +61 and replace with 0)
KW i just dont know how to set it up, there seems to
ND == Nitesh Divecha [EMAIL PROTECTED] writes:
ND Hello All, Recently I added some Nokia N95 customers and it worked
ND pretty good. Now the customers are complaining about the dialing
ND rules... They are used to dialing +12486543210 and +4479XX for
ND long distance calls.
ND Is there
MM == Marco Mouta [EMAIL PROTECTED] writes:
MM Siemens GigaSet SL75
The SL75 is DECT, not Wifi.
Apart from that, was it really necessary to quote 20 lines and add a
ridiculous 15 line disclaimer telling me that I'm not allowed to read
the message?
/Benny
MF == Marcus Franke [EMAIL PROTECTED] writes:
MF There is a GigaSet SL75 WLAN.
MF http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html
MF Hmm, I did not see any DECT SL75..
You are indeed correct, and I apologise.
I was thinking of the SL37; how I messed them up I don't
SB == Stephen Bosch [EMAIL PROTECTED] writes:
SB Hi, folks: I remain intrigued by the gap in BRI implementation
SB between North America and Europe, and I wanted to get feedback
SB from the list members on the matter. I'm seriously considering
SB making the leap in our office.
BRI is being
CA == Colin Anderson [EMAIL PROTECTED] writes:
CA Sometimes it's asterisk, sometimes it's unknown sometimes,
CA it's Unknown so:
CA exten = asterisk,1,VoicemailMain(${CALLERIDNUM}) exten =
CA Unknown,1,VoicemailMain(${CALLERIDNUM}) exten =
CA unknown,1,VoicemailMain(${CALLERIDNUM})
CA This
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is
PC working fine in my Setup, but I want Extern1 to talk to Extern2
PC directly whitout going over Asterisk as the uplink is slow.
PC When I set for Extern1/2 canreinvite=yes it
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC But then all RTP Traffic of my internal phones will go over
PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern
PC and Extern-to-Extern should go P2P and Intern-2-Extern should
PC go over Asterisk, see picture
I understand what you
ER == Eric Rousse [EMAIL PROTECTED] writes:
ER Hi, I was planning on getting a Dell PowerEdge 2950 for our new
ER Asterisk configuration. But while searching for documentation
ER about it and/or reported issues, I found this:
ER http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING -
ER
CB == Chris Bagnall [EMAIL PROTECTED] writes:
CB I have run a few speed tests from the sites in question (iperf to
CB the machine in the datacentre) and I'm consistently getting around
CB 380k upstream and 5.5mbit downstream, even during peak hours. Some
CB distance away from the quoted speeds,
I have a problem with asterisk-1.2.13, where it retries SIP INVITEs
too quickly. It happens when qualify is on, and the server it tries to
reach is only 1ms away according to qualify.
The time between the first SIP INVITE and the 7th (last) is then only
64ms, and that can be too short for the
BA == Benny Amorsen [EMAIL PROTECTED] writes:
BA I reported this bug in much more detail in bugs.digium.com, but
BA the bug is gone now without even an email saying where it went. I
BA don't remember the issue number. Somewhat frustrating.
Yay my bug is there after all! ID is 9020. It doesn't
M == Matt [EMAIL PROTECTED] writes:
M I talked to Dell technical support and they said oh all our new
M machines share IRQs like that, the way you are trying to do it is
M archaic.
The technical support is right. Digium should fix their driver (or
possibly the card). Perhaps it's fixed in the
M == Matt [EMAIL PROTECTED] writes:
M According to him, and he backed everything up (and I have no doubts
M about how PCIe is working) with PCIe devices can share a single
M IRQ, because there is so much more throughput. However, obviously
M with PCI this does not work.
Why does it not work
M == Matt [EMAIL PROTECTED] writes:
M I guess the question is... is it even possible to have a real-time
M VoIP card running on PCIe? Or with 1,000 Interrupts a second.. does
M it simply need to have its own IRQ?
I don't know why you are so fixated on PCIe. PCIe can do shared
interrupts,
M == Matt [EMAIL PROTECTED] writes:
M PCI has always had the ability to do shared IRQs, you are correct,
M however, that is a bad idea for any real time application,
M especially VoIP. If you have your NIC card and Digium TDM2400
M sharing an IRQ, when you get a bunch of calls going over your
M == Matt [EMAIL PROTECTED] writes:
M Benny, I've seen it happen, though, with other things. For
M instance, with a mouse and modem on the same IRQ... you can end up
M getting disconnected from the Internet when the mouse is moved, if
M the modem is trying to access resources at the same time
M == Matt [EMAIL PROTECTED] writes:
M I plan to call Digium about this tomorrow... and find out what the
M official word is on using the PCI cards in PCIe slots,
That's easy. The card won't fit. It will fit in PCI-X and work, but
PCIe isn't backwards compatible hardware-wise.
/Benny
I have a challenge that is ending up quite interesting. I need to
identify which SIP phone touched a call last, that is, which phone did
the last transfer or dialed the original call if no transfers were
done.
It is easy in the case of a regular, non-transfered call. Just put
something in
ML == Mailing Lists [EMAIL PROTECTED] writes:
ML Yes, it would be wrong to expect performance near that mark. Most
ML systems cannot handle the TCP processing load generated by a
ML gigabit ethernet interface, let alone process everything that goes
ML along with calls associated with that
EW == Eric \ManxPower\ Wieling Eric writes:
EW All of our SIP phones dial instantly when the users finished
EW dialing. We can do this because we have no ambiguous extension
EW lengths. i.e. no _XXX and _ and we don't use the . pattern
EW match.
If you have managed that even for
RS == Rob Schall [EMAIL PROTECTED] writes:
RS Example: John calls in from the outside using (213-555-1234) and
RS he calls into the asterisk system (actually the operator). The
RS operator (a real person) answers the call and presses transfer on
RS her polycom 501 phone. I see an incoming call
MB == Michael Boers [EMAIL PROTECTED] writes:
MB I have setup an asterisk based phone system using snom-320 (SIP
MB based) phones.
MB I would like to change what seems to be the default procedure for
MB an attended call transfer. Right now, the phone user places the
MB call on hold, calls the
CA == Carlos Alperin [EMAIL PROTECTED] writes:
CA The error is that when I run modprobe the result is FATAL NO
CA ZAPTEL MODULE FOUND.
CA Any clue about this?
It is important that you do not rephrase error messages, but copy them
directly.
I probably can't help you even with the correct
ML == Mike Lynchfield [EMAIL PROTECTED] writes:
ML With all other things said.. you might want a professional service
ML for this like targusinfo.com
ML Maintaining and running an operation like a cname web lookup thing
ML is REALLY high overhead in terms of web traffic etc
ML What happens
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL TP'n to follow flow just like DNS, the 'root servers' would still
RL see the high request hits, prior to passing off to local caching
RL app.
The DNS root servers are almost only loaded by irrelevant traffic. The
root information is easily
ML == Mike Lynchfield [EMAIL PROTECTED] writes:
ML Well caching is the way to go., bu then again most of the current
ML solutions have this problem.
ML John smit has a DID.. 514 555 1234 and closes account.. did sleeps
ML for 3 months and new client Jane doe takes it..
ML Now how long should
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL everytime you make a dns request, i agreed that it does not hit
RL the root servers, but every time you request a NON-cached one you
RL DO.
Nope. If you request foo.com and you have up to two days earlier
visited bar.com, you won't hit the root
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