> "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
> "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 v
> "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 th
> "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 alre
> "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 u
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. http://www.astric
> "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
> "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 h
> "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
> "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 scrip
> "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 i
> "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 u
> "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
_
> "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>
> "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, the
> "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, an
> "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
> "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 pro
> "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 As
> "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
> "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 li
> "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
_
> "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.
> "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 e
> "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 band
> "uxbod" == <[EMAIL PROTECTED]> writes:
uxbod> Hi, I have a requirement for sending and receiving faxes and
uxbod> was wondering the best way to achieve it with Asterisk as I
uxbod> only have one phone line.
uxbod> I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I
uxbod> was
> "JN" == Jordan Novak <[EMAIL PROTECTED]> writes:
JN> Okay, I get it. I still have a problem though. I have no way to
JN> wire 30% of these end-points. P{hysically impossible. They do have
JN> cat3 twisted pair to each phone. But of course they want IP. Are
JN> there any adpaters that will gi
> "PB" == Peter Bowyer <[EMAIL PROTECTED]> writes:
PB> No it can't - the latest registration 'wins'. To achieve
PB> simutaneous ringing of more than one phone (hard or soft), you
PB> need a SIP account for each and an entry in the dialplan which
PB> rings both.
Indeed, this is the limitation
> "RH" == Rizwan Hisham <[EMAIL PROTECTED]> writes:
RH> OK, but y would i want to use it. i mean y not use goto and y
RH> this? and what dialout files are you talking about?
You can't Goto in queues.conf
/Benny
___
--Bandwidth and Colocation pro
> "RH" == Russell Horn <[EMAIL PROTECTED]> writes:
RH> Hi, I've set up a Gizmo Project account for access on my Nokia E61
RH> because they work through NAT. Trouble is If I include my gizmo
RH> account in an asterisk hunt group and I'm not connected (phone is
RH> off / outside wireless coverag
> "dc" == dave cantera <[EMAIL PROTECTED]> writes:
dc> I am a bit confused about the way forward with my upgrade. I am
dc> not very good with Linux systems, but would appreciate your advice
dc> to sail through my upgrade successful.
Avoiding html in email would be a good start.
/Benny
___
> "RC" == Ricardo Carvalho <[EMAIL PROTECTED]> writes:
RC> I've also tried to do it using different contexts, but it still
RC> doesn't work. I've done like this:
RC> [default]
RC> exten => secretary_extension,1,Dial(SIP/secretary_extension)
RC> exten => boss_extension,1,Dial(SIP/secretary_ext
> "HH" == Henning Holtschneider <[EMAIL PROTECTED]> writes:
HH> MWI works on the KIRK Wireless gateways we are using.
Kirk ip600/3?
If so, how do you configure it?
/Benny
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-user
> "DC" == Dave Cotton <[EMAIL PROTECTED]> writes:
DC> I never could understand how a RAID could be made up using SCSI
DC> disks seeing that they are certainly not inexpensive.
Small Computer Systems Interface. SCSI was vastly cheaper and
(perceived as, at least) less reliable than the proper
> "SB" == Stephen Bosch <[EMAIL PROTECTED]> writes:
SB> As somebody else has already pointed out -- "There must be more to
SB> it." Let's say three of four drives failed -- the odds of them
SB> failing at the same time are vanishingly slim;
Not as slim as manufacturers want to make you belie
I have a rather interesting issue with catching calls which have been
hung up by the callee, but not by the caller. I would like those calls
to return to an IVR, and it almost works:
[incoming]
exten => 12345678,1,Goto(testIVR,s,1)
[testIVR]
exten => s,1,Answer
exten => s,n(play),Background(be
> "KD" == Klaus Darilion <[EMAIL PROTECTED]> writes:
KD> Hi! Is there unicode support in Asterisk for SIP? E.g. How can I
KD> have a displayname with special characters?
KD> E.g. if I want to have the Umlaut ä in the display name:
KD> callerid=Jeff Gräser <11>
Is your sip.conf UTF-8-encoded?
> "SS" == Stuart Sheldon <[EMAIL PROTECTED]> writes:
SS> This might sound strange, but is there anyway for Asterisk to set
SS> extra sip headers based on a sip phone returning a 302 in a
SS> dialplan?
You can detect that a redirect has occurred by looking at ${RDNIS}.
You can't tell which SIP
> "OEJ" == Olle E Johansson <[EMAIL PROTECTED]> writes:
OEJ> And for fax over VOIP, sometimes called FOIP, Asterisk 1.4.x
OEJ> supports T.38 passthrough. However, the 1.4.0 release is buggy,
OEJ> so either use 1.4 from subversion or wait for 1.4.1.
T.38 passthrough is not very exciting unless
> "MH" == Mike Hammett <[EMAIL PROTECTED]> writes:
MH> A client of mine has a Snom 320. Usually when he comes in each
MH> morning, it is asking him for a password. A power cycle brings it
MH> back to normal operation. How do I troubleshoot this further?
It isn't necessary to power cycle, it's
> "LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:
LA> If it's not a security issue I might as well have all phones with
LA> context=default in sip.conf even though voip-info specifically
LA> warns against that. Wonder why?
Random SIP calls from the internet could end up in context default, i
> "LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:
LA> I have a sip.conf with stanzas for sip phones that have
LA> 'context=sip-incoming for some Grandstream phones and another
LA> stanza for a Sipura SPA3000 with context=pstn-incoming.
LA> Reviewing the code today, I was dismayed to see that
> "BT" == Brad Templeton <[EMAIL PROTECTED]> writes:
BT> Hey, we could even build a system where DNS can be used to take
BT> any phone number and look up data about it, not just a name, but
BT> even a URI to redirect calls to for it, a source of presence info
BT> and more.
BT> What a great id
> "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
> "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
> "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 ea
> "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> Wha
> "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 c
> "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
> "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 inc
> "EW" == Eric \"ManxPower\" Wieling 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 i
> "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
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 caller
> "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.
/Benn
> "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
> "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
> "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
interr
> "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
> "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
>>>>> "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 frustratin
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 peer
> "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 quo
> "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 W
> "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
> "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 canreinvi
> "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(${CALLERIDNUM}) exten =>
CA> unknown,1,VoicemailMain(${CALL
> "KS" == "Savoy, Kevin <- Williston, ND" <[EMAIL PROTECTED]>> writes:
KS> We have been running an Asterisk box with 1.2.9.1 on it since
KS> August in a call center environment. We use the Asterisk box as an
KS> IVR and then pass the calls on to a Nortel Option 11C. Today we
KS> found in our l
> "RL" == Richard Lyman <[EMAIL PROTECTED]> writes:
RL> grr, i hate when i typo (and reply to my own posts) exten =>
RL> s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx)
Heh, if you want to chase typos, perhaps you should add an underscore
before ?
/Benny
___
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> Ok, but how does that help me? All I want to do is set a variable
DG> to be used later on in the dialplan. Eg, if someone dialls
DG> 2944000, which is in a different company...:
DG> [co1_phone-start]
DG> include => co1_did
DG> include
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
>> [example]
>> include => ctx31X
>> include => ctx3XX
>>
>> exten => _X.,1,NoOp(this gets executed first for everything)
>> exten => _X.,2,NoOp(this gets executed second only if ctx31X
>> or ctx3XX didnt match)
>> exten => _X.,3,NoOp(th
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> Benny, lets say I have this...
DG> exten => _X.,1,NoOp(1)
DG> exten => _X.,2,NoOp(2)
DG> exten => _X.,3,NoOp(3) <- Current code execution location
DG> exten => 555,1,NoOp(1)
DG> exten => 555,2,NoOp(2)
DG> exten => 555,3,NoOp(3)
DG>
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> Surely other people have hit the situation where they first check
DG> extensions within a company, and then if there's no match, you
DG> glue all the other companies dialplans together with this one.
Of course we have. Just Goto(glued
>>>>> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
>> -----Original Message- From: Benny Amorsen
>> [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006
>> 6:16 AM To: asterisk-users@lists.digium.com Subject:
>> [asterisk
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> If I pass a priority, we're right back at square one, we're I'm
DG> stuck in a priority and can't get back to an extension.
You ALWAYS have both a priority and an extension. There is no such
thing as "being stuck in a priority".
/Be
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> So, in the event that the logic flows beyond
DG> coo1_OnNet, we want to reset the caller id of say, 3254001 ,
DG> to 3254000 .
DG> exten => 3254101,1,Dial(SIP/3254101,20,tr)
DG> exten => 3254102,1,Dial(SIP/3254102,20,tr)
DG> exten =>
> "PJ" == Pavel Jezek <[EMAIL PROTECTED]> writes:
PJ> tunneling small rtp packets through vpn has big overhead, better
PJ> to use application level encryption - encrypted iax or srtp.
IPSEC in transport mode without NAT has a very low overhead.
/Benny
_
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> When I exited the CLI and re-entered and pressed ctrl-c,
That's where your problem is. Use exit and not ctrl-c to leave
asterisk -r.
/Benny
___
--Bandwidth and Colocation provided by Eas
> "ZZ" == Zeeshan Zakaria <[EMAIL PROTECTED]> writes:
ZZ> Switches should be Layer 2 or Layer 3, and what's the difference.
You really should hire someone to do the design.
ZZ> Another question I have is about 10/100/1000 Mbps. In a standard
ZZ> switch, ports don't actually work at 100 Mbps.
> "JO" == J Oquendo <[EMAIL PROTECTED]> writes:
JO> I don't follow... Remove the mechanical lifter? Then do what, go
JO> from the Plantronic to the headset jack on the Snom, leave the
JO> receiver in its normal port? If I do this, the person has to hit
JO> the "headset button" on the Snom...
> "JO" == J Oquendo <[EMAIL PROTECTED]> writes:
JO> Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset
JO> --> Plantronic jack (bottom base in the front) If I placed
JO> Plantronic(RJ11) --> Snom's Headset port, the auto lift on the
JO> Plantronic wouldn't work until the person pr
> "CG" == Csibra Gergo <[EMAIL PROTECTED]> writes:
CG> Well, I think there's far more htan 1000 interrupts come from an
CG> T1/E1 card. Or do you think 1000/channel?
No I mean 1000 interrupts total, across all channels on all cards.
Otherwise the driver is just broken. Of course in a sane sys
> "CG" == Csibra Gergo <[EMAIL PROTECTED]> writes:
CG> Well, the data bandwidth is only one. The irq is the other, and
CG> that is the bottleneck.
You get 1000 interrupts per second. If that's a bottleneck then
there's something fundamentally wrong with your system.
/Benny
___
> "AMH" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes:
AMH> Well, perhaps the IF hinders evaluation from happening? It is
AMH> by far not as elegant, but you could try
AMH> exten=>123456,1,GotoIf($[${REGEX("^0..)} = 1]?2:3)
AMH> exten=>123456,2,Set(CALLERID(num)=44${CALLERID
> "JS" == Jon Schøpzinsky <[EMAIL PROTECTED]> writes:
JS> I would just guess that the PCI bus would be pretty busy, with 3
JS> T1 cards. Couldn't that be a problem? Jon
A T1 is less than 2Mbps. The PCI bus can just about handle 1Gbps
ethernet. That's a LOT of T1's.
/Benny
_
> "JO" == J Oquendo <[EMAIL PROTECTED]> writes:
JO> One thing I noticed about Asterisk and group rings is, if a phone
JO> is not registered but in the group, sometimes it won't ring.
What did you expect? If it isn't registered, Asterisk doesn't know how
to reach it, and therefore it doesn't r
> "MG" == Michael Graves <[EMAIL PROTECTED]> writes:
MG> Who will benefit as long as calls must typically pass into
MG> existing PSTN infrstructure, and so be transcoded into G.711? It
MG> seems to me that only systems that are IP end-to-end stand to show
MG> the improvements...or am I mistund
> "CS" == Curt Shaffer <[EMAIL PROTECTED]> writes:
CS> And to respond to Alex, The box is only doing Asterisk. 2.8Ghz
CS> proc with 1GB of RAM. The iptables is on the server itself.
Please don't top-post.
What about your OUTGOING chain, how does that look?
/Benny
> "EW" == Eric \"ManxPower\" Wieling writes:
EW> exten => _+NXXNXX,1,Goto(${EXTEN:1},1)
You are assuming a fixed-length number here...
The basic problem is to ensure that the extension gets run first,
since asterisk has its very own ideas about what goes first. So put
the incoming calls
> "DL" == Doug Lytle <[EMAIL PROTECTED]> writes:
DL> Tim Uckun wrote:
>> Judging by the lack of response here it seems like this is broken
>> and nobody knows how to fix it.
DL> 98% of the people here don't use Trixbox.
I have the same issue as reported earlier. I have never tried Trixbox.
> "TU" == Tim Uckun <[EMAIL PROTECTED]> writes:
TU> I am seeing the following in my log file (standard trixbox
TU> install). One seems to be complaining about an error in the
TU> dialplan but it won't tell me what file or what line. The other
TU> (maybe related) is complaining about a channel
>>>>> "BA" == Benny Amorsen <[EMAIL PROTECTED]> writes:
BA> An alternative is to make an extension which goes to voice mail
BA> directly, and simply redirect the phone to that extension. It's a
BA> bit more than one button, but at least the Snom 360
> "NZ" == Norbert Zawodsky <[EMAIL PROTECTED]> writes:
NZ> If I leave my desk I press this button. A light should show up as
NZ> an indicator/reminder. From this moment all calls to my extension
NZ> should immediately be transferred to my voicemail box.
NZ> When I return I press the button ag
> "SP" == Scott Pinhorne <[EMAIL PROTECTED]> writes:
SP> I am setting up my phones so that if the callerID is 3 digits the
SP> phones ring one way if it is more than 3 digits it rings another
SP> i.e. internal calls and external calls.
SP> exten => ,1,GotoIf($["${CALLERIDNUM}" = ""]?5
> "MW" == Mike Williams <[EMAIL PROTECTED]> writes:
MW> The control connection (port 5060) obviously goes via the asterisk
MW> server as it has to work out where to send the control to, but I
MW> could quite easily imagine the audio going directly handset to
MW> remote server or handset to ast
> "BT" == Brad Templeton <[EMAIL PROTECTED]> writes:
BT> The correct behaviour, as I see it is:
BT> a) Native bridge when connecting two external channels --
BT> everybody is on the real internet b) Native bridge when connecting
BT> two internal channels -- everybody is on the 192.168.* n
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> If you include 10 contexts, and each one of those has a realtime
DG> switch, than that's 10 times that Asterisk has to query the
DG> database, for a single call.
Not that I would make extensions.conf realtime, but...
One trick to avo
> "DL" == Doug Lytle <[EMAIL PROTECTED]> writes:
DL> I don't specify it on the phone. My Asterisk server changes it's
DL> time and all of the phones pick it up.
The phones get their time from Asterisk? Which protocol do they use
for that?
/Benny
___
I have a bunch of Snom phones. When I press the mute button, the phone
stops sending RTP frames. If I have rtptimeout set, that means that
the connection will eventually be cut off. It also affects sound
generated by asterisk, since timing is generated from the incoming
frames.
Are there any worka
> "HK" == Håkan Källberg <[EMAIL PROTECTED]> writes:
HK> Problem B - Quick Dial Buttons:
HK> I have used the programmable function keys together with the hint
HK> system in * to monitor local lines. It works very well,
HK> impressive! But people like to use these buttons as quick dial
HK> but
> "MJ" == Martin Joseph <[EMAIL PROTECTED]> writes:
MJ> I added the rtptimeout=60 to my general section in sip.conf, and
MJ> now when the e60 goes out of wifi range, 61 seconds later, my
MJ> channels are clear! Sweet.
Does this work with canreinvite=yes? (I can't see how it could, but
I'd lik
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