On Wed, 8 Dec 2004, Roy Sigurd Karlsbakk wrote:
> > for call forwarding, I was told by the telco to "set the call forward
> > number" on the PRI,
> >
> > how can I do this?
>
> To answer my own message, I need to set the REDGNO (0x74) number to the
> originating number in the PRI SETUP. Example
On Wed, 8 Dec 2004, dean collins wrote:
> There doesn't seem to be any interest in using asterisk and video.
>
> I posted a $1,000 bounty to get video meet me working without a single
> reply.
>
> I have now just bumped this to $2000
> http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty
On Wed, 8 Dec 2004, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Peter Svensson <[EMAIL PROTECTED]> wrote:
> >
> > The RHEL 3 (and thus WhiteBox, TaoLinux etc) kernels also work well.
>
> Are they 2.4 or 2.6 kernels?
2.4 as is to be expected
On Wed, 8 Dec 2004, Eric Wieling aka ManxPower wrote:
> I don't really understand why do many people set their pridialplan. The
> text "PRI Dialplan: Only RARELY used for PRI." in zapata.conf.sample is
> not there to just take up space in the file.
I'd change that comment to "almost ALWAYS ne
On Wed, 8 Dec 2004, Eric Wieling aka ManxPower wrote:
> Colin Anderson wrote:
> > Some telco's do, some telco's don't. The text "Rarely used for PRI" is IMO a
> > location-centric and carrier-centric statement. As another poster suggested,
> > PRI's are picky, and some carriers are super-picky so
On Wed, 8 Dec 2004, Rich Adamson wrote:
> A fairly standard telco operating approach is _not_ to provide any answer
> supervision, and * works just fine without it for lots of folks.
Answer supervision allows you to dial several recipients at the same time
and bridge the first one to answer. Nic
On Tue, 7 Dec 2004, John Harragin wrote:
> What I have in mind is a pci card with zap-like-driver that supports digital
> phones. This eliminates (is compairable to using channel bank) additional
> delay and a primary echo source when both haves of a conversation are carried
> on the same pair
On Tue, 7 Dec 2004, Lee Howard wrote:
> On 2004.12.07 10:06 Matthew Boehm wrote:
> > Here is the setup:
> >
> > POTS -> PRI -> Asterisk -> ATA (Fax)
> >
> > The ATA is set to only 711. Asterisk's sip.conf sets this device to
> > only
> > 711. Yet, faxing works less than 50% of the time.
>
> I h
On Tue, 7 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
> So besides the Budgettone 100(or any other), there is not way to force
> asterisk to play a message. What about if the phone will be connected to
> tdm400 port?
See "immediate=yes" in the zapata.conf file.
Peter
_
On Wed, 8 Dec 2004, Steve Underwood wrote:
> Andrew Kohlsmith wrote:
> >Are you using RH's stock kernel or a plain-vanilla kernel? I have heard
> >nothing but bad things with Asterisk and RH's "custom" kernels. If you can,
> >try a stock 2.6.9.
> >
> It is just the kernels supplied with FC2 th
On Tue, 7 Dec 2004, Nick Burch wrote:
> Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines
> providing it with external connectivity. We have several analogue
> extensions spare, but no capacity to add fancier connectors to link to an
> asterisk system (as most of the PBX link
On Tue, 7 Dec 2004, el Flynn wrote:
> John Harragin wrote:
> > Are there any digital phones that run on asterisk yet? I'm talking about
> > non-IP phones here...
>
> Asterisk can work with ADSI phones, more info on the wiki at
> http://www.voip-info.org/wiki-ADSI
You can use isdn phones, if yo
On Mon, 6 Dec 2004, Kris Boutilier wrote:
> The originating PRI system passes the entire dialed number in the d-channel
> setup frame, thus the concept of a wait time for additional digits is
> meaningless. Progressive digit gathering implies that the signalling is
> occuring 'in-band' as would be
On Mon, 6 Dec 2004, Jerry Glomph Black wrote:
> Kris, thanks for the thoughful & helpful response! This makes sense in the
> same way that a dialplan on a SIP phone would behave.
>
> But... If I remove the 3-digit number (224) from the asterisk
> dialplan, I have no problem dialing 2246 suc
On Sat, 4 Dec 2004, Kevin Blackham wrote:
> Yeah, proper crossover cable. I've eliminated all cabling issues with
> the T1 analyzer. I get a full and accurate pattern back when I test
> from the cable end where it would have been connected into the T100P,
> with the channel bank in loopback. Th
On Sat, 4 Dec 2004, Rich Adamson wrote:
> > The mind boggles -- PRI is *always* out of band.
>
> Looks like the command is documented in the current config samples.
>
> I'm not knowledgable/experienced (as yet) on where it is actually used,
> but just reading the comments in the config sample le
On Sat, 4 Dec 2004, Tracy R Reed wrote:
> I have created hint priorities in my dialplan:
>
> exten => l00,hint,SIP/100
> exten => 100,1,Macro(stdexten,100,SIP/100)
^
I guess it may just be a typo during retyping, but you have 'l' (lower
case L) in the hint line and a '1' (one) in the
>From reading the source it seems that the incoming calling number TON/NPI
are not copied from the notification from libpri.
In chan_zap.c the callerid is copied from e->ring.callingname to
pri->pvts[chanpos]->cid_num in pri_dchannel. However, the ring.callingplan
is not stored anywhere.
If o
On Wed, 1 Dec 2004, Brian C. Fertig wrote:
> You can setup recording by default. This is how I have mine setup. I
> don't believe the way app_queue is now you can have the agent press
> something to have it start recording.
Maybe the patch in
http://bugs.digium.com/bug_view_advanced_page.php
On Wed, 1 Dec 2004, Brian C. Fertig wrote:
> But now in this instance it drops them into voice mail. Is there a way
> to have them punch in there phone number so they can keep there space in
> the
> system? Like if they are #20 in queue when they left their # for call
> back
> that when they g
[moved to asterisk-users]
On Wed, 1 Dec 2004, Chris A. Icide wrote:
> currently asterisk requires that you have one D channel per PRI, and that D
> channel must be channel 24.
>
> Is it possible to support one D channel for multiple spans?
>
> It seems that you would need a bonding definition.
On Wed, 1 Dec 2004, Brian West wrote:
> Or he has a Channelized T1 with inband signaling.
Not on four BRIs he isn't, not a T1. :)
I wonder if someone runs voice channels with inband (or robbed bit!)
signalling on an bri-interface? Now that would be a weird thing.
Peter
On Wed, 1 Dec 2004, Steve Underwood wrote:
> Peter Svensson wrote:
> >Maybe he has NFAS (Non Facility Associated Signalling) where the D channel
> >on one of the BRI lines handles the signalling for the B channles on all 4
> >BRIs.
> >
> I think NFAS would be
On Wed, 1 Dec 2004, Steve Underwood wrote:
> Patrick wrote:
> >I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
> >"Anlagenanschluß" with 8 B-channels in Germany. It worked fine with Deutsche
> >Telekom, but since we switched to Arcor nothing works at all.
> >After some
On Wed, 1 Dec 2004, Enoch Root wrote:
> I'm diagnosing a problem related to PRI card. I would
> like to know the following: assuming I've got a
> working PRI card and correctly installed Linux drivers
> and a PRI line connected to the card, even without
> starting asterisk, shouldn't I hear a rin
On Wed, 1 Dec 2004, Dave Cotton wrote:
> On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
>
> > What I found on voip-info.org was that I didn't have a working timer -
> > and I had to load ztdummy module. So I did (modprobe ztdummy), started
> > asterisk again, but I'm still getting
On Tue, 30 Nov 2004, Karl Brose wrote:
>
> Why don't you just set up an extension that calls the system application
> to execute a Linux script
> Then just make a call to that extension, perhaps use disa to
> authenticate and done.
The application Authenticate() may be more suited.
Peter
__
On Mon, 29 Nov 2004, Andrew Kohlsmith wrote:
> Checking our fax logs, almost *every* company we fax (several hundred) all
> connect at > 14.4kbps and have the ECM or whatever it's called turned on.
This is our experience as well. Most companies here in Sweden seem to have
moved to laser faxes a
On Mon, 29 Nov 2004, Mark F. Vickers wrote:
> According to the FAQ "When you load the module and have no
> circuit/channel bank the LED's should flash red"
>
> I get the "knight rider" lights before the module loads, but after the
> modules are loaded I don't get any lights, other equipment plu
On Mon, 29 Nov 2004, Matthew Marlowe wrote:
> Is anyone successfully using directed call pickup with asterisk?
>
> *8 to only pick up that persons extension if the phone is
> ringing.. It says in the wiki asterisk supports it but I can not get
> it to work..
You could use app_intercept from
ht
On Sun, 28 Nov 2004, Lee wrote:
> So my question remains: Is PCI 2.2 a requirement to use the TDM400P
> card? If so, where is this specified? If not, is there a performance
> difference when using PCI 2.1?
PCI 2.2 is mostly a clerification on the 2.1 specification. One difference
that may be imp
On Sun, 28 Nov 2004, Lee wrote:
> On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
> > Only way that I know is to open the case and look at the slot to see if
> > there are two dividers. I would be interested in knowing this as well.
>
> I've seen many motherboards
On Sun, 28 Nov 2004 [EMAIL PROTECTED] wrote:
> This looks like a config issue, "class of service barred" but getting
> config information out of verizon is nearly impossible. I compared what
> the Mitel is sending to asterisk (since the mitel does work with the PRI)
> with what asterisk is sendin
On Sun, 28 Nov 2004, Bob Goddard wrote:
> On Sunday 28 November 2004 19:25, Steven P. Donegan wrote:
> > Well - if 2.6.etc did adopt this it isn't reflected in actual make/make
> > install world - i.e. nothing gets installed in /lib/modules/anywhere...
> > And this is with kernel source from kerne
On Sun, 28 Nov 2004, Chad Scott wrote:
>
> On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote:
> > Fair enough. If my unserstanding is correct perhaps someone can add a
> > note
> > to the wiki? It is not totally obvious.
>
> Peter, why don't *you* add
On Sun, 28 Nov 2004, Brian West wrote:
> I don't agree with this patch yet... It's the distro's fault for doing this
> wrong and I don't feel we have to work around it. The few people I talked
> to have Symlinks the "build" to /usr/src/linux or the like. Then again I
> may be wrong anyone know w
SetVar.
> >
> > Pointers would be appreciated.
> >
> > Thanks,
> > Trevor Peirce
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
&g
On Sat, 27 Nov 2004, Rich Adamson wrote:
> > You misunderstand me. I know that the buffers are larger. However, even if
> > they are 1 second deep they will eventually empty / overrun. There is no
> > way about this except to either allow data to be invented/dropped or to
> > keep the source an
On Sat, 27 Nov 2004, Rich Adamson wrote:
> > There is a
> > buffer but the buffering can only handle jitter, not compensate for
> > frequency difference.
>
> No, you're assuming a one-byte (or very small) buffer, and that's not
> what's going on in asterisk.
You misunderstand me. I know that
On Sat, 27 Nov 2004, Rich Adamson wrote:
> > True. However, you want to distribute the clocking to _all_ your
> > downstream peripherials to avoid the equivalent of frame-slips. If your
> > cards are not clocked the same exactly you will need to invent/drop a
> > freme efery now and then. That
On Sat, 27 Nov 2004, Peter Svensson wrote:
> On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote:
>
> > > How to implement some of the function into asterisk like *67 "call
> > > number blocking"
> >
> > exten => _*67*X.,1,CallerPres(32)
> &
On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote:
> > How to implement some of the function into asterisk like *67 "call
> > number blocking"
>
> exten => _*67*X.,1,CallerPres(32)
> exten => _*67*X.,1,Dial(Zap/g1/${EXTEN:4},${TIMEOUT},${DIALOPTS}))
Do you mean CallingPres? There is more informati
On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote:
> > Change this into SetVar(_SIP_CODEC=g726) and it will work.
>
> you sure?
>
> sipgw1:/usr/src/asterisk # grep -r _SIP_CODEC .
> sipgw1:/usr/src/asterisk #
The leading underscore means the variable will be inherited by the
outgoing channel. Di
On Sat, 27 Nov 2004, Rob Emanuele wrote:
> I've got a pretty easy question here I can reconfigure my configs
> pretty easily when I'm storing everything into a MySQL database. In the
> case of using the zaptel cards and zapata.conf how would I reload the
> config of an individual channel?
On Fri, 26 Nov 2004, Rich Adamson wrote:
> I've read the early posts relating to this and there still seems to be
> a misunderstanding on this clock sync issue. This stuff has been around
> for a long time in the telephony business, but it seems like not
> many people understand it on this lis
On Fri, 26 Nov 2004, "Dr. Fernando Macías Garza" wrote:
> It seems to me that if not all cards are clocked from the same source,
> then each one should be able to get its own external clock. However,
> card 0 has an external clock, but card 1 does not. Look at this:
[snip]
> I am sure the line
On Sat, 27 Nov 2004, Steve Underwood wrote:
> Peter Svensson wrote:
> >Most providers should be synchronized to a traceable time source derived
> >from UTC. I.e. they should all tick exactly the same even if they are not
> >directly interconnected.
> >
> >
&g
On Fri, 26 Nov 2004, Andrew Kohlsmith wrote:
> There can be only one clock and you must engineer your system such that
> everything is synchronized properly. For simple systems like what we are
> describing it's not difficult but when you have multiple spans coming from
> multiple providers it
On Fri, 26 Nov 2004, Voip Business wrote:
> Guys is there any E&M available?
>
> thought it was only fxo and fxs.
E&M signalling is supported on the T1/E1 cards. There are no cards from
digium supporting analog 4-wire E&M. You need to hook up a channel bank
for that at the moment.
Peter
__
On Fri, 26 Nov 2004, Patrick wrote:
> On Fri, 2004-11-26 at 11:36 -0500, Andrew Kohlsmith wrote:
> [snip]
> > No.
> >
> > 0 = don't use the remote clock for sync (use internal clock)
> > 1 = use remote clock as card's primary clock source
> > 2 = use remote clock as card's secondary clock source
On Fri, 26 Nov 2004, Andrew Kohlsmith wrote:
> On November 26, 2004 11:06 am, Patrick wrote:
> > Doesn't "sync source" mean that the card is generating its own clocking?
> > If your telco provides the clocking, the card should not.
>
> 0 = don't use the remote clock for sync (use internal clock)
On Fri, 26 Nov 2004, Francois Fernandes wrote:
> - Caller checking:
>
> If someone calls the number of the Asterisk server it should be able to
> check if the guy is allowed to call this number. That means, that
> asterisk should pass the number to a third parity program which decides
> if the num
On Thu, 25 Nov 2004, Colin Anderson wrote:
> I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
> opinions:
>
> 1. asterisk -p == renice -20 ??
The -p option sets asterisk to realtime priority if possible. This is
different from the traditional unix nice levels. A program
On Thu, 25 Nov 2004, Rich Adamson wrote:
> > However, zttool reports card as "Internally Clocked". No matter how I've
> > tried, I cannot get card 1 to clock from the external source:
> > Sync Source:Internally clocked
> >
> > First span on card 0 is configured just the same:
> >
> > sp
On Thu, 25 Nov 2004, Ashling O'Driscoll wrote:
> So basically if I want to support approx 100 calls, I would have to
> purchase a digium PRI card and then pay eircom (or whoever my service
> provider is) approx 3000 a year for the PRI ISDN connection??
100 simultaneous calls would require 4 E1 ba
On Thu, 25 Nov 2004, Alex Barnes wrote:
> Sorry I dont have any answers, however I do have a question.
>
> I was told that ISDN-30 lines do not work during power failure. Can
> anyone with some better knowledge confirm or deny this?
> Is this because the ISDN-30 box on the wall requires power (
On Thu, 25 Nov 2004, TinKoon wrote:
> However, for the Asterisk implementation, unless you have a huge ups, you
> will not be able to make and receive any call during power failure, since
> there will be no power to the Asterisk server. And since all the incoming
> lines, be it analog lines or T1/
On Wed, 24 Nov 2004, Andrei (MPI) wrote:
> David Boyd wrote:
> >On Wed, 2004-11-24 at 04:14, Mike Dent wrote:
> >>Hi,
> >>I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and
> >>1 SIP phone.
> >>I've noticed some horrible buzz/rasping type of sounds! These seem to occur
> >>when
On Wed, 24 Nov 2004, Michael Vogel wrote:
> Soren Rathje schrieb:
>
> > Note: The Wildcard X100P/X101P only have FCC approval.
>
> What does that mean for me? Is it illegal to use it in germany or do
> they don't work in germany?
The X100 only support the US line impedance (600 ohm resistive).
On Tue, 23 Nov 2004, Asterisk wrote:
> At the moment, I have the following working scenario:
>
> isdn30B->(1)te410p(2)->merdian(B)
> isdn30A->meridian(a)->te410p(2)
>
> IOW, two isdn30 lines, one going to * span 1, the other going to the
> meridian (pri card a), which then is connected to * by p
On Tue, 23 Nov 2004, Chad Sawyer wrote:
> I have a pri comming into a t100p in my asterisk box. I have a second
> t100p configured as pri_net connected to a nas server. I can route
> modem calls to the NAS with no problem, but I am concerned about isdn
> data connections.
>
> Will asterisk rout
On Tue, 23 Nov 2004, Ben Merrills wrote:
> Is there a way to log all PRI events to a logfile?
Maybe "pri intense debug span ???" is what you are after? If you set up a
logging file in /etc/asterisk/logger.conf that logs everyting you should
get all the pri events.
Peter
_
On Mon, 22 Nov 2004, Nick Bachmann wrote:
> You know you shouldn't (can't?) use the same interface for regular IP
> networking and TDMoE, right? The TDMoE should have an address-less NIC
> to itself and _really_ shouldn't run through a hub (an xover would be
> ideal). Bonding seems possible,
On Mon, 22 Nov 2004, Steve Prior wrote:
> Michael Welter wrote:
> >> echocancel=yes
> >> echocancelwhenbridged=yes
> >>
> > Steve Underwood says not to use echo cancel on a fax line.
>
> Oops, you're right. I knew I was not supposed to use echocancel, but
> somehow got these two lines backwards.
On Mon, 22 Nov 2004, Kevin Brennan wrote:
> > Using iax trunking will also loose the advantage of being tdm all the way,
> > i.e. low latancies. If the rest of the setup is tdm there is a lot of
> > value in not going to voip for one hop.
>
> This is what I was thinking, FAX would be more reliabl
On Mon, 22 Nov 2004, Jason Williams wrote:
> I recommend you use Iax trunking rather than TDMoE this would scale better.
Using iax trunking will also loose the advantage of being tdm all the way,
i.e. low latancies. If the rest of the setup is tdm there is a lot of
value in not going to voip fo
On Sat, 20 Nov 2004, Brian Roy wrote:
> I would look at putting a dual monitor on her desk. You can pick up a
> 15" flat panel and a video card for about the same cost as the SNOM.
> Not to mention, you get quite a bit more benifite from the FOP
> controls than you do busy lamp fields. It's a a ne
On Fri, 19 Nov 2004, Michael Devenijn wrote:
> We are located in Belgium and just ordered a PRA line, the telco asked
> the following questions :
>
> - 120 or 75 ohm ?
120 ohm is delivered over two balanced twisted pairs and normally
terminated in an rj45. This is what you need for th
On Thu, 18 Nov 2004, Rich Adamson wrote:
> Examples:
> 1. two-wire analog pstn lines: as soon as current draw is sensed by
> the central office, answer supervision is generated by that "central
> office", period. It has nothing to do with whether * handled it or
> whether an analog phone is hangin
On Thu, 18 Nov 2004, Daniel wrote:
> >On Thu, 2004-11-18 at 12:05, Chad Scott wrote:
> >You *can* play a welcome message without >answering the line, however,
> >this doesn't always work. eg, I tried this >config on my PRI in Australia
> >(Telstra) and:
> >
> >a) Calling from a standard analog lin
On Wed, 17 Nov 2004, Joe Greco wrote:
> > I don't think this is really a key system. AFAIK a traditional key system
> > has a one-to-one mapping between lines and the buttons. Some pbx:es offer
> > a mode where each *extension* is / can be represented by a button. This is
> > called a Busy Ligh
On Wed, 17 Nov 2004, Thomas Hutton wrote:
> Question: Does anyone know of a lightweight popup method to put an
> incoming call ID string on a client machine? Something as simple as
> winpopup would work great- for example: I have a call coming in on Zap/4
> but the phone on Zap/4 doesn't have a c
On Wed, 17 Nov 2004, Jason Becker wrote:
> > On our current phones (Iwatsu) we have a button on the
> > phones for each extension that lights up when that
> > extension is ringing or is in a call, so I can see at
> > a glance if one of my coworkers is on the phone before
> > I go barging into his
On Wed, 17 Nov 2004, Steven Critchfield wrote:
> On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
> > Thanks for your feedback, after I restarted Asterisk the card came up as
> > expected. However I am still seeing these WARNINGS when I reload *, to be
> > clear I have not made any additio
On Wed, 17 Nov 2004, Matt Riddell wrote:
> Peter Svensson wrote:
> > I guess you just have to know that Brian is a bit trigger happy sometimes.
> > It has it's ups and downs. Things get fixed quickly, but sometimes his
> > instinct is wrong.
>
> I was beginning
On Wed, 17 Nov 2004, Matt Riddell wrote:
> Régis MARTIN wrote:
> > When I first read the answer, I look at it like another quick answer with no
> > understanding of my problem.
>
> Aha! But you didn't notice that it was Brian West (bkw) who gave you
> the answer!
>
> He is one of few able to
On Tue, 16 Nov 2004, Tobias Jönsson wrote:
> On Mon, 15 Nov 2004, Jason Williams wrote:
> > After the Authenticte why not do a Playtones(Dial) this will give
> > dialtone
>
> The dialtone won't stop after pressing first digit then. If course you can
> have an X extension that will do a StopPlay
On Mon, 15 Nov 2004, Jim Dossey wrote:
> I have a client who currently has a Toshiba PBX. We are trying to
> replace it with an Asterisk system. One of the features that they have
> on their current PBX is the ability to select a POTS line by pressing a
> button on their phones. They have 10 PO
On Mon, 15 Nov 2004, Brian West wrote:
> Ok to cut confusion here
>
> Its:
> Variable: _ALERT_INFO
> Value: somevalue
>
> Its always var/val via manager.
Not in the Originate action it isn't. This is what both the help
show manager command originate
say and what reading the source indicate
On Mon, 15 Nov 2004, Peter Osborne wrote:
> I am using the Asterisk Manager API to originate calls and it is working
> well,
> when a call is placed the local phone rings, once you pick it up you can here
> the call ringing the other end. Now, I am using Polycom IP 300 and I have
> them setup
On Tue, 9 Nov 2004, Henry Devito wrote:
> HI I am trying to use the outcall going by the wiki.(
> http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the
> errors below. Here is a sample of a callout file. What am I doing wrong?
> Begin Outgoing.call
> Channel: sip/
On Tue, 9 Nov 2004, Kristian Kielhofner wrote:
> G.711 is a standard that defines Ulaw and Alaw, commonly called Ulaw
> and Alaw. But last I checked Meetme transcodes all codecs to Ulaw for
> the purposes of the conference. So, I suppose G.711u would be your best
> bet for low processor
On Sun, 7 Nov 2004, Reid A. Forrest wrote:
> > Currently, our office phone systems have 6 outside lines
> > coming in. The
> > actual phones have lights ( indicators ) for these lines, so matter
> > where you are in the office, you can look at the phones and see that
> > someone is on line #2
On Sat, 6 Nov 2004, William M. Sandiford wrote:
> Excuse the newbie nature of the question, but can you elaborate a little
> further. Sorry...I am pretty new
There is a block in the queues.conf.sample file in the Asterisk
distribution that reads:
; A context may be specified, in which if the u
On Sat, 6 Nov 2004, William M. Sandiford wrote:
> Hello All:
>
> I need some help. I am trying to configure * so that users that are
> placed in a call are able to break out of the queue and go to voicemail
> if they no longer wish to wait in the queue. I read the cmd options for
> the Queue co
On Fri, 5 Nov 2004, Ryan Thrash wrote:
> What about an expensive Supermicro dual Xeon PCI-X system with 1GB ECC
> RAM and a hardware RAID controller (it was SATA, though)?
>
> Echo was noticeable even on SIP-to-SIP calls internally with the
> system, with all sorts fo tweaks to tx/rx gain. Supe
On Fri, 5 Nov 2004, Matthew Marlowe wrote:
> This seem to be fixed in CVS 11/05 - Altho ALERT_INFO is still broken
> in CVS 11/05
Isn't this an effect of the new automatic variable inheritance? Since
ALERT_INFO is used in the called channel you would have to set _ALERT_INFO
instead of ALERT_INF
On Fri, 5 Nov 2004, Kurt Bauer wrote:
> --On Thursday, November 04, 2004 04:41:53 PM +0100 Peter Svensson
> <[EMAIL PROTECTED]> wrote:
>
> > On Thu, 4 Nov 2004, Kurt Bauer wrote:
> >> connection is to a Ericsson MD110 wich is set as network, * is set as
> >
On Thu, 4 Nov 2004, Nate Carlson wrote:
> Area you using a PRI line, or what?
>
> If a PRI, you need your provider to allow you to set the outgoing CallerID
> to whatever you'd like, instead of just one of your own numbers.
>
> If BRI, Analog, etc, I don't think there is a way to set your own
On Thu, 4 Nov 2004, Kurt Bauer wrote:
> > Is your timing source set correctly? If you are connecting to the pstn
> > the pstn connection should be the primary timing source.
>
> connection is to a Ericsson MD110 wich is set as network, * is set as CPE.
Have you set the span as the timing source
On Thu, 4 Nov 2004, Kurt Bauer wrote:
> Hi list,
>
> every now and then I get the following message in my * logs:
> chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
> D-channel of span 1
>
> As this is only a notice and voice worked quite well, despite the messages,
> I
On Wed, 3 Nov 2004, Michael Loftis wrote:
> --On Wednesday, November 03, 2004 15:40 -0700 Manuel Marin
> <[EMAIL PROTECTED]> wrote:
>
> > I would like to know if there is a way to change default ulaw for a T1
> > card. I see in the zap show channel X that is working as ulaw. How do I
> > change
On Wed, 3 Nov 2004, Paulo Adriano wrote:
> After trying that syntax I still have the same problem, one thing is
> very strange is the number that Asterisk reports as the incoming. My
> ESN*s numbers are 219898334 and 219898335 but on the console I see
> 219898334,1 and 219898335,1
Are you sure it
On Wed, 3 Nov 2004, Sergiu Dunca wrote:
> We have a E1/digital PBX in the office for ISDN-PRA telecom access and
> for digital phones connected to it. So the PBX is E1 capable.
>
> I wander if we can use a E1 interface in the PBX and a Digium PCI card
> to interconnect the PBX with Asterix. The
On Tue, 2 Nov 2004, TC wrote:
> > I have been thinking about adding inter-queue dependencies to ICD. It
> > seems better suited to more advanced queues than the built in queue
> > system.
> >
> > Another option in icd would be to keep all callers in one queue and then
> > pop the first customer wh
On Tue, 2 Nov 2004, steve szmidt wrote:
> > It is quite true for some classes of batteries. E.g. some Li-ion batteries
> > will explode if charged (or in the case of rechargeable batteries charged
> > with the wrong voltage / polarity). They pack quite a punch as well. The
> > normal household alk
On Tue, 2 Nov 2004, steve szmidt wrote:
> Our battery manufacturers did the same to stop people from recharging them.
> They came out with this fantastic story of how batteries might explode if
> charged. Most everyone believed it too.
It is quite true for some classes of batteries. E.g. some L
On Tue, 2 Nov 2004, Jonathan Moore wrote:
> Yes, I am aware of these settings, but unfortunately I know of no way to set a
> penalty flag for a dynamic agent. They have to be dynamic, because they agents
> log in and out to go on break, on/off shift and the command for adding them in
> this manne
On Mon, 1 Nov 2004, Jon Lawrence wrote:
> There isn't a digium solution to connect to POTS lines in the UK other than
> X100P's, and I for one can't live without callerID - I'm even considering
> going across to ISDN so that callerID continues to work with future *
> versions.
There are a lot
On Mon, 1 Nov 2004, Luís Palma wrote:
> I've been digging around /zaptel/zonedata.c file which has the
> different frequency tones per country, and I would like to know the
> purpose of the following fields in the struct data defined there.
>
> For example in US data we have:
> { 0, "us", "Unite
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