For everyone that had inquired about the Find-Me/Follow-Me
application, it's now up in the bug tracker at
http://bugs.digium.com/view.php?id=5574.
It should compile cleanly against a 1.2b2 install.
On 10/14/05, BJ Weschke <[EMAIL PROTECTED]> wrote:
> CF -
>
> You'r
t; Regards
>
> Lee
>
>
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ
Weschke
> Sent: 02 November 2005 12:20
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Installing beta2
>
>
>
>
Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch.
On 11/2/05, Lee Archer <[EMAIL PROTECTED]> wrote:
Once built no matter whether I do make install or make clean I get the same output
[EMAIL PROTECTED] aste
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it.
On 11/2/05, Dave Morrow <[EMAIL PROTECTED]> wrote:
Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way callin
No sir. emailbody= may only be used within the [general] section of voicemail.conf
On 11/1/05, Kuniyoshi Murata <[EMAIL PROTECTED]> wrote:
Hello * users,I think I understand the basic configuration of /etc/voicemail.conf.What I would like to do is, to use different set of text for the voicemail t
This should work since you're not injecting any voip into the middle of the situation.
On 11/1/05, Neil Skowronek <[EMAIL PROTECTED]> wrote:
I'll bump this thread and add a link to a similar one.
http://lists.digium.com/pipermail/asterisk-users/2005-October/130359.htmlI'm trying to do the same th
You're looking for maxmsg= which can be set per mailbox and per context. In a mailbox setting, set it in the options field I believe.
On 11/1/05, Adam Moffett <[EMAIL PROTECTED]> wrote:
So maxmessage is a global setting for all voicemail boxes?BJ Weschke wrote:> maxmessage= in
v
maxmessage= in voicemail.conf Asterisk does not automatically delete voicemails after a period of time.On 11/1/05, Adam Moffett <
[EMAIL PROTECTED]> wrote:This is asterisk 1.2 beta.It seems a certain voicemail box wouldn't accept any more messages when
there were 100 messages in it...but I can't fi
The first post was received here in addition to your second one. While you can make an announcement to an agent prior to them being bridged to a call (to identify what queue a call is being delivered from, etc), I don't think that the current functionality of that parameter allows that announcemen
What does your extensions.conf look like now to try and implement this? Some prior examples that were based on jumping priorities may not work so well with a stock 1.2b install where priorityjumping=no in extensions.conf.
On 10/26/05, Ben Higley <[EMAIL PROTECTED]> wrote:
yes.. that is what i was
The number of spans. If you've got a quad card, you can actually configure 4 different wanpipe interfaces on that card.
On 10/25/05, Sharon <[EMAIL PROTECTED]> wrote:
sangoma tech's cldn't help with that error.Also it did work with asterisk 1.0.3 version wanpipe-beta9-2.3.3
now i'm using wanpipe-
Have you tried contacting Sangoma technical support? They are likely the best equipped to support the card and the alarm you're receiving.
On 10/25/05, Sharon <[EMAIL PROTECTED]> wrote:
I am using a A104 Sangoma card. We are runningasterisk cvs head on ourproduction box.After wanpipe configuratio
strace?
valgrind?
There aren't any profiling tools or the sort within the Asterisk suite that will deliver the information you're looking for about what each thread is doing.
On 10/20/05, Jason Walker <[EMAIL PROTECTED]> wrote:
When I run 'ps aux' I get this:
root 964 0.0
I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron <
[EMAIL PROTECTED]> wrote:It is set to rfc2833.Tom Vile wrote:
> maybe its not setting the DTMF tones properly. What do you have setup> for the phone and ex
Richard,
I'm sorry you and others feel the way you do. Businesses though don't
want an open source project that is a free for all when it comes to
contributions and discipline both in the code itself and
documentation.
Olle has contributed hundreds of hours of his own time over the time
he's b
If you're using AgentCallBackLogin it should be fairly easy to to do what you're looking for in step 'b'. On 10/18/05, Julian Lyndon-Smith <
[EMAIL PROTECTED]> wrote:I was wanting to use the new MuxMon application to record calls - it
seems to be a "nicer" way of recording than the Monitor applica
exten => s,1,SetAccount(internet) exten => s,2,Queue(internetq|t|||360)
Now an agent, say Sonia, (her SIP extension is 8012 too) calls 5000 andlogins as agent 8012.When she is on an ACD call, she does not receive another ACD call. Verygood.However when she makes an outgoing call, she
Can you open a bug please in bugs.digium.com and attach your sip.conf and then a full SIP debug/trace of a call attempt into the bug?
This will help us reproduce/diagnose the issue.
Thanks.
On 10/17/05, Sherwood McGowan <[EMAIL PROTECTED]> wrote:
Gents, this concerns a CVS-HEAD downloade
It is possible. I do it here and at many other client installs.
Please post your configuration so we can see why it's not working for you.
On 17 Oct 2005 15:31:46 -0400, J Thomas <[EMAIL PROTECTED]> wrote:
Given the current state of queues, it does not seem possible to stop ACDcalls coming t
You must specify to pqm and upqm what interface you're attempting to pause and unpause. It will not figure that out based on what interface the channel that called the app in the dial plan might have used.
On 10/17/05, Corey Frang <[EMAIL PROTECTED]> wrote:
So, I'm looking into using PauseQueueMemb
SIP requires RTP connections in addition to the signaling connection which normally happens on UDP 5060. The RTP connections vary in port usage (the range is configurable through rtp.conf) and are nearly impossible to get going without some "man in the middle" help when you have two Asterisk serve
search the list and you should find some examples on how todo it (I think there is even an example on the wiki try:http://www.voip-info.org/wiki-asterisk+cmd+dial
)On 10/14/05, BJ Weschke <[EMAIL PROTECTED]> wrote:> I have coded a new application in Asterisk called app_followme that wil
Should work that way. Right now it's coded so that if the person being called doesn't acknowledge with DTMF, it'll timeout and go on to the next caller.
If it doesn't reach anyone in the whole list, it then exits just as the Dial application does with DIALSTATUS populated and it's then your ch
Yes. They tell you what's acceptable to them inside the SIP messages you're trading back and forth on the call setup.
You can use "sip debug" within Asterisk to get a closer look at those messages.
On 10/14/05, Obelix <[EMAIL PROTECTED]> wrote:
Quoting Ray Van Dolson <[EMAIL PROTECTED]>:How c
I have coded a new application in Asterisk called app_followme that will do what you're looking for. The caller who made the call originally is also optionally put on hold music while the hunt is going on. There's also planned functionality for "blacklisting" certain callerIDs so a caller who is b
The quadspan card isn't a low profile card is it? I don't think it'll even physically fit in the net4801's footprint.
On 10/11/05, Craig Guy <[EMAIL PROTECTED]> wrote:
Has anyone on the list used a Soekris engineering PC as a TDM - Ethernetbridge? For example something like a net4801 with a TE11
There isn't a way to do it in agents.conf.
That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to s
The DS3 currently available from Sangoma doesn't do channelized voice, yet. You wouldn't be able to use it with Asterisk, yet.
On 10/8/05, Jason Walker <[EMAIL PROTECTED]> wrote:
Has anyone used the DS3 card from Sangoma with Asterisk?I have read many posts from users that the Sangoma cards have b
I've got a Polycom 501 that I run off a Wifi "Game Adapter" in my home. It works fine.
On 10/7/05, John Reynolds <[EMAIL PROTECTED]> wrote:
On 10/7/05, Will Glass-Husain <[EMAIL PROTECTED]> wrote:> Hi,
>> I'm provisioning an office with limited cabling. I'm looking for a desk> based wifi phone.
You might want to try Digium's support. They're probably the most qualified to support their card and help you with adjustments to manage through what the PSTN network is likely causing.
On 10/6/05, Ken Dresdell <[EMAIL PROTECTED]> wrote:
Hello,We are looking for a consultant able to remotely co
On 10/6/05, Mark Hatton <[EMAIL PROTECTED]> wrote:
Hi all,I have situation in that we are renting out a room in our office building toanother company, including the provision of phone lines via our Aterisk box.
To keep billing simple, we would like them to be billed separately for alltheir calls.
Upgrade Asterisk. Versions of HEAD post 8-29-05 have this functionality built in.
On 10/4/05, Mark Elkins <[EMAIL PROTECTED]> wrote:
I'm using a SNOM 360 with Ver 4.3 software.Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff +
Head)I've used the wiki info to set up some line
Your telco must provide remote disconnect supervision on that trunk in order for disconnects to be recognized by your card.
On 10/3/05, Leigh Fereday <[EMAIL PROTECTED]> wrote:
I realise I am not alone in this, but I don't seem to be finding manysolutions.I am running a CentOS box and *
1.09 wit
That's what I had thought originally too, but apparently there is/was an issue with the 3.1.5gw and below firmware where it was possible for AC noise coming from the power supply to be falsely identified as a ring. Sipura has apparently just released
3.1.7 to deal with this.
I've never had t
Yes. It's gone.
On 10/3/05, Dinesh Nair <[EMAIL PROTECTED]> wrote:
On 09/30/05 03:12 Verlin Henderson said the following:> Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
> large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or> TE410P cards and implement
ne these settings in "International Control" in the Advanced/Admin and PSTN Line tab section of the Spa3k config, but you've got to know what and for how long you're receiving something first before you know what to tune.
On 10/3/05, Paul Dugas <[EMAIL PROTECTED]> wrote:
On Mon, Oct
e line voltage that is coming through the FXO port.
On 10/3/05, Paul Dugas <[EMAIL PROTECTED]> wrote:
On Mon, October 3, 2005 9:33 am, BJ Weschke wrote:> Do you have the SIP acct it's interacting with enabled for MWI? There's a
> setting in the SPA3k where it will ring the
Do you have the SIP acct it's interacting with enabled for MWI? There's a setting in the SPA3k where it will ring the phone periodically for one ring in addition to the stutter tone for MWI.
On 10/3/05, Paul Dugas <[EMAIL PROTECTED]> wrote:
This is a wierd one. Can't figure it out. I have an SPA
stion about discover line protocol between pbx&ivr in any way?Do you have confience with AG-NMS Card ?I appreciate your interest.Regards
--- BJ Weschke <[EMAIL PROTECTED]> wrote:From: BJ Weschke <[EMAIL PROTECTED]>Date: Tue, 27 Sep 2005 08:03:43 -0400
To: [EMAIL PROTECTED], Asteri
What is the line protocol you're using on this legacy PBX? Is it E&M Wink? If so, then you'd just configure the Digium card for wink, plug in a T1 crossover cable and you should be ready to start testing.
On 9/27/05, Exciting <[EMAIL PROTECTED]> wrote:
I want to replace a custom PBX, that is infro
Is there a functional reason why you'd use MeetMe here? I think probably the easiest way to accomplish this is to use an DeadAGI script which can be invoked via the 'h' extension in the context that would then perform the functionality you're looking for and if they get through it should just brid
The snom360 phones along with the current CVS-HEAD of Asterisk can presently do this. You'll want to do a wiki search on "Hint" in the dialplan for implementation details.
Polycom has also just released a DSS sidecar to go with their 601 model phones, but the firmware to support more than 8 app
For your outbound calling problem, if you're operating with CVS-HEAD you can PauseQueueMember and then UnpauseQueueMember as part of the dial-plan for your outbound calls for those agents.
On 9/17/05, Rajkumar S <[EMAIL PROTECTED]> wrote:
Hi,I have a small callcenter with 3 agents who login usingA
IAX will use less than individual SIP calls when trunked, yes. I'm not sure it's a significant savings with the number of streams we may be talking about for your particular scenario, but for larger carrier trunking scenarios, it could be quite significant.
On your original question, if you wa
That is an issue with Vonage not providing remote disconnect supervision through the ATA. The way I got around that was not to use Comedian mail with my home system, but instead, just use an analog answering machine I already had around. Not ideal, I know, but the answering machine hangs up the FX
The CID with the Cisco isn't a "Cisco issue". It's actually an issue based on the way Vonage passes CID through the Cisco. It doesn't follow the same standard that LECs and others use.
I tried to get this going with an SPA3000 at first as well and never really could get it to go right without h
What version of spandsp are you attempting to compile in to the 1.0.9 tree?
On 9/14/05, David Sampson <[EMAIL PROTECTED]> wrote:
Anyone know how to fix this?gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff
In file included from app_rxfax.c:14:
/usr/include/asterisk/lock.h: In
What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static IP that these calls will be coming from and only accept calls from that IP, but that's still not quite as secure as digest authentication that would be available
When your landlord switched the phone service, he more than likely put on new service that doesn't supply remote disconnect supervision which was what was causing disconnects to be detected correctly before. They will need to activate this for you again in order for things to begin working again a
Linksys and Netgear switches now also do private VLANs for far less than 6k. They will not provide the features/functionality/management that your 6k Catalyst will provide, but it doesn't sound like you're looking for anything more than making sure traffic from room to room is secure.
While y
b is possible. See res_features.conf for more information on transferring via DTMF.
c is not yet possible. This would require shared call appearances which isn't yet implemented.
On 9/8/05, Arnar Birgisson <[EMAIL PROTECTED]> wrote:
Hello,Avaya has a nice feature that allows you toa) ring bot
app_rxfax and txfax will not compile unless they see spandsp.h in /usr/include/ or /usr/local/include as per the Makefile in asterisk/apps
On 9/6/05, Arne Morten Johansen <[EMAIL PROTECTED]> wrote:
Hello.I just emerged spanDSP with all the packages needed. After a bit ofconfiguration i was read t
Yes. You can use the openh323 toolkit along with the ast-oh323 channel to do this or you can also try and use the more recent native h323 channel driver provided by Digium.
My experience with doing this comes from doing it with the openh323 toolkit with the ast-oh323 channel and it does work w
Yes. SER is a good way to do load balancing. The way you'd do it here would be to have SER as the edge device talking to your voip provider and then have it balance calls back to your asterisk instances using DNS SRV or a similar technology for the balancing.
As far as limiting inbound calls,
Inbound calls?
Outbound calls?
Zap channels (TDM) ? SIP channels? IAX channels?
There's a good number of questions that need to be answered before one can get to an answer on your original question.
On 9/6/05, Matt King <[EMAIL PROTECTED]> wrote:
Hello, I'm going to need to take up to
There is a patch that Frank Sautter has been working on to get call pickup and the record button on the Snom's working. It's 5014 in Mantis. The current patch available there will likely need manual intervention to get it to merge with the current CVS-HEAD now that 3644 has been committed.
Fra
Yes. You could send a sip notify via asterisk and set that up via cron to happen once per day.
eg.
"asterisk -rx "sip notify reboot-snom "
Make sure that the reboot-snom clause is setup in sip_notify.conf before attempting this.
On 9/1/05, altus <[EMAIL PROTECTED]> wrote:
IS there a way
Issue #3644 has recently been committed to CVS-HEAD which allows for full device state notification via subscriptions for Snom 360 and other supporting phones w/o the need for additional patches.
On 9/1/05, Alessio Focardi <[EMAIL PROTECTED]> wrote:
Hello Paul,Thursday, September 1, 2005, 4:38:42
On 8/26/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
> Broadvoice sent out a notice and threatened to disconnect me if I did
> not respond. If I disagreed with their stand they would disconnect me too.
>
> I think they said something like we don't have it and we ain't getting
> it. Click here to a
It shows ringing via a blinking LED and some other device states for
devices like the Snom phones and eyebeam.
On 8/26/05, Kevin Ragsdale <[EMAIL PROTECTED]> wrote:
> Pardon my ignorance, but could someone explain to me what the benefits
> of this patch? We use 1.0.9, and have our Polycoms showi
I've been thinking about how one would accomplish the same thing.
I've got a CTI enabled GUI that tells the agent that they're logged in
with the call centers that I've deployed thus far, but it's not quite
the same as the agent just being able to look at the phone as well and
know that they're lo
> Fra: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] På vegne av Eddie
> Sendt: 24. august 2005 08:42
> Til: asterisk-users@lists.digium.com
> Emne: [Asterisk-Users] Fax to email using mime-contruct
>
> I have followed and succesfully receive incoming faxes to email with
> Scott Laird's faxing wit
You will need to do this either in the code itself or within an AGI script.
On 8/24/05, Jeremy Salmon <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I need to exec and app during on a channel during a dial.
>
> For exemple I have to exec every 30s app_reverse during a dial command.
>
> Thanks.
> Jeremy
I'm using their dca proxy and have not had any problems at all today
with them.
I've got 201 and 212 DID's with them and both have completed incoming
calls throughout the day today.
On 8/19/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
> OK, now I know of 5 peeps that suddenly are having this pr
Your line must provide disconnect supervision, which it sound like
you say that it does, and you must configure that Zap trunk in
asterisk for "kewl start" signaling (fxoks) which makes use of the
disconnect supervision.
On 8/18/05, James Fogg <[EMAIL PROTECTED]> wrote:
> I'm trying to use Asteri
Yes. It does work. I had it working with a Varion card into an S8700
system with a TN464 DS1 circuit pack sitting in a IP 600 cabinet
earlier this year as a proof of concept.
Double check that your *ANI*DNIS* settings in the Definity setup
match what you're expecting on the Asterisk side of thin
I would go with an ATA like the SPA-1001 and an analog set that mets
your requirements. That's more than likely going to be your best bet.
On 8/16/05, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 16:01, Tue 16 Aug 05, Colin Stefani wrote:
> > I have a call center where we're looking at conver
I believe that this is a solid state device based trying to follow
the principle that the things that would normally go wrong with a
server (fans, spinning hard disks, etc) aren't present here and
therefore should hang in there longer than a server with a PCI card
plugged in.
I don't disagree wi
You're right. Sounds like there's definitely something more going on
that just a missing library. Where did you get this version of
asterisk from? The CVS server or a binary/RPM installed from
somewhere? Is this the first install on this machine or has there been
other installs prior to this?
On
ogg_vorbis is a media format, like MP3/WAV/etc (most notably used for
music on hold in the case of Asterisk).
You are probably getting this error because you're missing required
libraries on your linux install in order for the module in asterisk to
be able to read ogg_vorbis files.
If you have
Check out the Asterisk @ Home project. I believe they've done some
similar integration with the XML capabilities for the Cisco phones.
I'm not sure about Polycom.
The XML directory Polycom is talking about is an XML file that it can
pull down from it's "boot server" on boot time from the phone.
No. It's not that. I know that was a problem previously, but I've had
the same problem as the user mentioned and those emails aren't in my
Spam folder. It's like they've completely disappeared.
I guess that's why they call it Beta. :-)
On 8/1/05, Time Bandit <[EMAIL PROTECTED]> wrote:
> > I ha
It does work. Your registration line looks correct except for the <
and > characters. Those shouldn't be there. Pull them out and the
registration should work for you after a "sip reload".
On 7/19/05, Bernie Courtney <[EMAIL PROTECTED]> wrote:
> does anyboy have a how-to online on how to do this?
[EMAIL PROTECTED]/1234
> register => user:[EMAIL PROTECTED]/2345
>
> And then create a dialplan for extensions 1234, 2345, etc?
>
> > -Original Message-
> > From: BJ Weschke [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, July 19, 2005 8:11 AM
> > To: Nate K
The way I fixed this with another provider that had similar behavior
was to patch chan_sip.c so that it pulled out the DNIS value from the
to: tag in the SIP header and then threw that it into the DNID channel
variable. Then, I took the common extension ('s' in your case) and did
a Goto with the v
That's coming from a function that's being re-used from the voicemail
application. You will need to do some custom C coding to remove it.
On 7/12/05, Robert Goodyear <[EMAIL PROTECTED]> wrote:
> Anyone know how to bypass the CONFIRMATION of the user announcement
> recording in MeetMe?
>
> While
I have used it. The biggest difficulty of the footprint is that the
chip doesn't do IO-APIC-edge interrupt handling in linux so you end up
having a real difficult time if you try to add additional devices (eg
- zaptel cards, more nic cards, etc) and don't want to end up sharing
interrupts with all
Yes. I configured it for a former employer.
We had an S8700 talking to * via h.323 with no problems.
oh323 did need to have it's rtp frame size adjusted initially for
some sound quality issues, and we needed to dbl check that oh323
wasn't trying to negotiate for codecs that * didn't want to
It is likely possible. It's going to depend on getting * and your
modem bank to play nice together. If your modem bank is collecting ANI
or any kind of other carrier signaling info for normal operation, you
might have an easier time doing E&M wink between * and the modem bank
if your modem bank su
Send me your current app_meetme.c and I will patch it for you so that
you can do it, and then you can provide that patch back to the
community.
On 5/23/05, Michael Blood <[EMAIL PROTECTED]> wrote:
> I have been trying to figure a way to SendDTMF into a MeetMe room using
> the Manager API.
>
> I
just became GA.
On 5/19/05, M O <[EMAIL PROTECTED]> wrote:
> BJ,
>
> >BJ Weschke <[EMAIL PROTECTED]>
> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
> >SIP termination vs. DS3
> >To: Asterisk Users Mailing List - Non-Commercial
> >Discussi
Did I miss pricing/availability announcements from Digium on that DS3
card somewhere? I wasn't aware they were going to be GA in less than 3
weeks from now.
On 5/19/05, M O <[EMAIL PROTECTED]> wrote:
> Message: 16
> Date: Thu, 19 May 2005 00:16:34 -0600
> Michael,
>
> Do both!
>
> As for Sip Te
I have found them to be very reliable to work with.
On 5/17/05, Manjit Riat <[EMAIL PROTECTED]> wrote:
>
>
> I am going to buy some IP phones from them but I sent them an email couple
> of weeks ago and got no reply. Has anyone ordered anything from them? Any
> other places that I can buy from?
This isn't a switch. It's just mid-line injector for POE, but you can
interchange for Cisco and Polycom.
We deployed them in the network here because we had a number of ports
on the switch that had no need for PoE and we didn't want to pay
Extreme Networks for PoE on ports we knew we weren't goi
You need to post your extensions.conf and oh323.conf for further assistance.
It sounds like though that the h.323 endpoints are sending a call to
you and since you didn't define a default extension/context for them
to go to, they are trying to go to extension 's' in the default
context, but this
It is indeed much higher. I'm using it here in production w/o media
running through it and it is supporting 400 connections with virtually
no load on it on a 1.8Ghz machine.
On 5/17/05, Alistair Cunningham <[EMAIL PROTECTED]> wrote:
> Michael Manousos wrote:
> > Alistair Cunningham wrote:
> >> Mi
It plugs directly into the back of the Digium card. You will not need
that CSU.
On 5/13/05, Oswaldo Arratia <[EMAIL PROTECTED]> wrote:
> Hi,
> A customer has a Avaya PBX and is looking to migrate to Asterisk, they have
> a T1 from the telco going into a CSU and then from the CSU to the Avaya PBX
huh? That's a TDM/RBS type question.
I've not seen most implementations of SIP interconnections doing
things like that?
On 5/12/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> Hello,
>
> Being totally fed up with the lack of quality and reliability from both
> VoicePulse and BroadVoice,
> We are
I ordered a 973-XXX- and 585-XXX- DID from them on 2/3 and 2/7
of this year respectively.
Their customer service portal still lists these orders as "pending"
though they told me back when I ordered them that provisioning would
happen "within 1 business day".
On 5/11/05, Wiley Siler <[EMA
et an WARNINGs or any other kind of logging info when you
reset the password? Looking at the code, it's supposed to issue
warnings if it cannot open the old file for read and/or open the new
file for write.
On 5/11/05, BJ Weschke <[EMAIL PROTECTED]> wrote:
> I see what you're say
Are you running a 2.4 kernel or a 2.6 kernel? Use a 2.6 kernel and
you won't have these problems. If that's not an option for you, then
you may be stuck. :-/
You may want to look at app_conference instead as it doesn't require
ztdummy for boxes that don't have any zaptel hardware in them.
On 5/
Only if you're on a 2.4 kernel. A 2.6 kernel doesn't require USB for
it's timing source.
On 5/11/05, Chris <[EMAIL PROTECTED]> wrote:
> Edit the Makefile for the zaptel drivers. You will see two commented
> lines that say ztdummy. Uncomment them and rebuild.
> Once you install the rebuild
stop asterisk
modprobe zaptel
modprobe ztdummy
start asterisk
Try now.
On 5/11/05, Daniel Salama <[EMAIL PROTECTED]> wrote:
> I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm
> getting the following problem:
>
> -- Executing MeetMe("SIP/3210-38a9", "0224|qM") in new
t;[EMAIL PROTECTED]> wrote:
> On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
> > voicemail.conf
> >
> > edit that file and issue a reload to change them.
>
> I tried this, but I still can't get access to voicemail from one of the
> phones.
>
> This is
voicemail.conf
edit that file and issue a reload to change them.
On 5/10/05, Jeff Heath <[EMAIL PROTECTED]> wrote:
> Where are user's voicemail passwords stored and how does the asterisk
> administrator change them?
>
> TIA,
>
> Jeff Heath
>
> ___
Not yet. Though Sangoma and Digium both have channelized DS3 adapters
that are now "in the works".
On 5/10/05, Kyle Hagan <[EMAIL PROTECTED]> wrote:
> Is there a DS# (T3) card that will work with asterisk? OR a card that
> supports more that 4 T1's per card?
>
> Kyle
> __
I was going to recommend the same to him last night, but then I
started digging into the code there and realized they were transcoding
back to LINEAR at their core as well. Now they're not passing that
back through a ZapTel psuedo channel like app_meetme does, but I'd be
interested to see if that
You are getting "all 1's" in the bitstream from your carrier. This is
a carrier issue with your circuit you have plugged into this card.
On 5/3/05, Chris A. Icide <[EMAIL PROTECTED]> wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
>
> Using zttool with a Sangoma A104 card, I am seeing a BLU/RE
I don't know about channel banks, but when you go T1 to T1 device
with a cable, you need the RX/TX pairs cross connected. Do you have a
T1 crossover cable in play or a straight through?
On 4/20/05, Dan Goscomb <[EMAIL PROTECTED]> wrote:
> Hi
>
> I have just purchased a Rhino Channel Bank and am
Sure. You need to decide how you will interconnect to * vm from CM.
H323? SIP? MGCP?
Then, you'll set your dialplan so that when calls come into *,
instead of going to a station first, it goes immediately to the
Voicemail app.
MWI is probably the biggest unknown. I'm not sure if anyone has
fig
401 - 500 of 514 matches
Mail list logo
