just to let you know that i've started a mailing list on sourceforge
[EMAIL PROTECTED]
You can subscribe here
https://lists.sourceforge.net/lists/listinfo/ccmanager-users
Other news regarding ccmanager will be posted on this mailing list, i
invite interested people to subscribe.
Thanks
On
Hi Guys,
Im looking for the pricelist of big scale pbx like nortel and avaya.Because im
going to make a presentation of cost against cost of open source implementation
for voip provider.
Anyone there could help me?
Thanks so much.
eduard
eng'r.eduard
-
On Wed, 14 Mar 2007, shadowym wrote:
Hard to expect the business community to take Asterisk seriously when
this sort of stuff happens IMHO.
I think you hit the nail on the head with one word: community.
Asterisk is free, community supported, and the voip-info site has been
provided for free
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 believe. RAID
Hi All,
I'm having problem with MP3Player app. I want the caller to hear mp3 when he
is waiting until I answer my phone.
-- from extentions.conf --
exten = 200,1,Answer()
exten = 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3)
exten = 200,3,Dial(SIP/200|20|tTrR)
exten = 200,4,Hangup()
--
Hi,
anybody who has a complete list of variables used by meetme conferencing
application in asterisk, plz share.
--
Regards
Rizwan Hisham
Software Engineer
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To
Hi,
Please discribe me how we define busy/hang/answer detection with PRI E1
channels.
Since busydetect, callprogress, busycount giving falts hangup and call drops
what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
___
On Thu, 2007-03-15 at 10:15 +0100, Benny Amorsen wrote:
Not as slim as manufacturers want to make you believe. RAID drives
tend to be purchased at the same time, so they are often from the same
batch. They are then subjected to exactly the same load in exactly the
same environment. Is it any
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten =
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 mainframe
Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
PRI channels have call supervision and Asterisk will see the
If voip-info.us would allow an rsync of the database, I would gladly host a
mirror. Since they won't, I have setup the domain listed below. If the
community is worried enough/upset enough, please consider putting
information at voip-wiki.us. I have no problem with people rsyncing the
database
Gordon Henderson wrote:
The site is pingable, so I'd suggest it's either crashed in some awkward
way and just needs resetting, but you never know...
Voip-info.org is down due to a hardware failure.
Will be back soon.
Thanks for using voip-info.org!
[EMAIL PROTECTED]
--
Tomislav Parcina
Hi,
I have a Siemens Gigaset DECT base connected to a Sipura SPA3000.
The Message Waiting indicator on the handset works fine in this
configuration.
(I've used both the S100 and SL100 phones / bases, the operation is
identical.)
The illuminating Message button on the handset can also be
Hi, i had the same problem recently for sip. my scenario was that i
connected 2 asterisk servers and dialed from one asterisk to another. and
for sending the DNID i used the following comand:
exten= 1,1,Dial(SIP/[EMAIL PROTECTED])
here riz is the channel name. hope this works with zap also
On
Am Thu, 15 Mar 2007 14:32:28 +1100
schrieb Daniel Pittman [EMAIL PROTECTED]:
It seems to me that a direct SIP to DECT gateway could have
significant advantages in terms of supporting the MWI (voicemail)
indicator on the DECT phone directly -- there just isn't any way I
could trigger it on any
I can mirror too, if needed. I have lots of bandwidth, email me off list
On 3/15/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
Gordon Henderson wrote:
The site is pingable, so I'd suggest it's either crashed in some awkward
way and just needs resetting, but you never know...
Voip-info.org
On Thursday 15 March 2007 12:32 am, shadowym wrote:
Hard to expect the business community to take Asterisk seriously when this
sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives
could just suddenly fail simultaneously. There must be more too it. No
UPS? Someone spilled
Andrew Kohlsmith wrote:
On Thursday 15 March 2007 12:32 am, shadowym wrote:
Hard to expect the business community to take Asterisk seriously when this
sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives
could just suddenly fail simultaneously. There must be more too it.
Stephen Bosch wrote:
Patrick May wrote:
On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote:
Yikes.. you'd think a server would be running RAID.
At any rate.. Please feel free to visit http://www.voip-wiki.us
I have set this up to be able to hold information for the Asterisk
Hard to expect the business community to take Asterisk seriously when this
sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could
just suddenly fail simultaneously. There must be more too it. No UPS?
Someone spilled their coffee into it? Something!
Sure, there always
On 3/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
Obviously you didn't read Google's research paper on drive failures.
This one?
http://labs.google.com/papers/disk_failures.html
-HJC
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Just a heads up guys. I'm currently attempting to recover the website
through spidering the Google cache.
I'll let you know how it turns out.
On 3/15/07, Drew Gibson [EMAIL PROTECTED] wrote:
Stephen Bosch wrote:
Patrick May wrote:
On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote:
Hi,
After further research, it seems SIP MESSAGE rfc3428) and SIP INFO (rfc2976)
methods could be the more relevant for this feature.
I'm still wondering whether SIP hardphones or Asterisk implement these
methods in such a way you could make a welcome message, for example, appear
on you contact
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
___
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asterisk-users mailing
shadowym wrote:
. I can't understand how 3 of 4 hard drives could just suddenly fail
simultaneously. There must be more too it. No UPS? Someone spilled
their coffee into it? Something!
If you can't understand it, do some research before mouthing off (as
everyone on this list is
Hi all
message:
qozap: t4 timer expired for span 2
qozap: t4 timer expired for span 3
qozap: t3 timer expired for span 2
qozap t3 timer expired for span 3
wow -- what does this mean!? all of a sudden showing up on my server ... no
change after reboot .. Junghanns QuadBRI card in place
Hi,
I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to
show which devices are busy/not connected. The same phone worked with
Asterisk 1.2.9.1.
I would appreciate anyone who knows how to setup Asterisk 1.4.1 to
behave as 1.2.9.1.
TIA
Giorgio Incantalupo
Benny Amorsen wrote:
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
On Thu, Mar 15, 2007 at 08:08:57AM -0600, Joe Greco said:
Anyone who's been in the industry for any length of time will have
stories. Some of them even interesting. I remember a few years ago
when the roof/wall of an ATT data center was destroyed during a storm.
Yep.
Ashburn VA datacenter.
On Thu, Mar 15, 2007 at 10:30:23AM -0500, Chris Earle (CBL) wrote:
Hi all
message:
qozap: t4 timer expired for span 2
qozap: t4 timer expired for span 3
qozap: t3 timer expired for span 2
qozap t3 timer expired for span 3
Which version is it of bristuff?
wow -- what does this mean!?
On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:
Hi,
I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to
show which devices are busy/not connected. The same phone worked with
Asterisk 1.2.9.1.
I would appreciate anyone who knows how to setup Asterisk 1.4.1
OCOSA List Acct. wrote:
Hi All,
Personally all of you who are complaining you need to stop becoming part
of the problem and become part of the solution. Everyone makes mistakes
and if you all depend on James' site so much then you need to donate
some time or contact him about getting a
mitcheloc wrote:
Just a heads up guys. I'm currently attempting to recover the website
through spidering the Google cache.
I'll let you know how it turns out.
Great stuff! I'll be keen to hear how it goes.
-Stephen-
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Matt you are right it is voip-wiki.us I looked at my browser tab. LOL
sorry...but my POV still stands... good day.
Otis Surratt Jr. / [EMAIL PROTECTED]
OCOSA Communications, LLC
PH: 918.585.9882 x 205 Fax: 918.585.5857
Visit Tulsa's Best Internet
wrote:
*snipped
If I can't be confident enough in an important source of information like
this then I can't be confident enough to provide an Asterisk solution to
businesses. That's the way I see it. Yea, it's a wiki but it's the best
source of info out there.
*snipped
sorry to see you
Klaus Darilion wrote:
Benny Amorsen wrote:
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
Brandon,
What it sounds like you are looking at as far as having the phones
register to the system and then have users login to a phone should be
possible, I have not tried. I would suspect that you could build a
dial plan menu to prompt the caller for their credentials and then
take the phone's
I was expecting a response like this.
First of all. I do NOT rely on any one source of information seeing as how
so much of it is outdate and/or just plain wrong. I always try get at least
2 or 3 sources of info.
Hey, don't blame the messenger. EVERYTHING comes with a manual right? Why
do
A percentage of all my profits go back to the community.
What about you?
-Original Message-
From: Gordon Henderson [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 15, 2007 1:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] voip-info.org
75% failure at EXACTLY the same time? Come on! We all know better than
that.
Probably lost one drive at a time over weeks or months with no automated
warnings!
Amateur hour!
-Original Message-
From: Patrick May [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 14, 2007 8:10 PM
To:
lol yeh all will miss you :D . . its like stopping to use internet if google
is down sometime .
On 15/03/07, Richard Lyman [EMAIL PROTECTED] wrote:
wrote:
*snipped
If I can't be confident enough in an important source of information
like
this then I can't be confident enough to provide an
Matt wrote:
Community,
I have put up www.voip-wiki.us http://www.voip-wiki.us
My apologies to our fellow Asteristians outside the us... this was the
only easy domain available.
What's wrong with voip-info.org ?
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Hi,
i am using the n-way-call dialplan solution found on voip-info. i have
added its entry in applicationmap of features.conf file. the problem
is..its not working. to activate the n-way call i dial *0 but nothing
happens. i have played around with dtmf and codec settings but no success.
the
I'm curious what you think that agenda might be?
If it is to push the perception of Asterisk as a solid alternative to
Traditional PBX's into the mainstream then I am guilty as charged!
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 15, 2007
I do not have any answer int he dialplan. what I mean is that when I
call any other SIP phone is does the answer in the CLI. Even if I put
and answer() in the dialplan still no ringing
Jason
Luki wrote:
shouldn't there be an answer in there somewhere?... like...
No... you can (and
A percentage of all my profits go back to the community.
What about you?
I think we've been contributing various resources to various online
Internet communities for about two decades, more if you go back into
the BBS era. We're still dedicating more than a quarter of a gigabit
of bandwidth
Nobody said anything about power supply problems did they?
Besides, this has NOTHING to do with one machine and what may or may not
have happened to it. It has EVERYTHING to do with the availability of the
information however that may be acomplished.
Half that info on the wiki is out of date
shadowym wrote:
A percentage of all my profits go back to the community.
What about you?
-Original Message-
From: Gordon Henderson [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 15, 2007 1:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
I completely agree with the cheap hosting commments - my company competes
against it all the time. Things go bad with the host in one way or another,
sites move, and the cycle repeats. Is that how someone reputable wants
to run a
business moving their site around every couple months when things
Quoting shadowym [EMAIL PROTECTED]:
75% failure at EXACTLY the same time? Come on! We all know better than
that.
Probably lost one drive at a time over weeks or months with no automated
warnings!
Amateur hour!
a power supply or backplane problem could easily physically damage the entire
If a wiki site about automobiles crashes, should I buy a horse?
shadowym wrote:
I'm curious what you think that agenda might be?
If it is to push the perception of Asterisk as a solid alternative to
Traditional PBX's into the mainstream then I am guilty as charged!
-Original Message-
What a bunch of whiny people! If you travel to the website now you'll
see the following note:
begin quote--
Voip-info.org is down due to a hardware failure.
Will be back soon.
Due to the kind offers of mirror services from many people, once the
site is back online, there will be a number of
From: Olivier [EMAIL PROTECTED]
Date: Thu, 15 Mar 2007 15:21:15 +0100
Hi,
After further research, it seems SIP MESSAGE rfc3428) and SIP INFO
(rfc2976)
methods could be the more relevant for this feature.
I'm still wondering whether SIP hardphones or Asterisk implement these
methods in such a
From: Mark Quitoriano [EMAIL PROTECTED]
Date: Thu, 15 Mar 2007 11:59:30 +0800
Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying
to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
I thought you'd get an error message about the syntax
#1 - It's down
#2 - The owner is prohibiting anyone from mirroring it.
On 3/15/07, Trevor Peirce [EMAIL PROTECTED] wrote:
Matt wrote:
Community,
I have put up www.voip-wiki.us http://www.voip-wiki.us
My apologies to our fellow Asteristians outside the us... this was the
only easy domain
I would suggest that we create a new wiki, make it solely for Asterisk
topics, as not to offend or replace voip-info. Build mirrors to
multiple sites and multiple domain names. This would give this
community a second resource with redundancy which is what I think ALL
of
us are looking for.
On 3/15/07, Matt [EMAIL PROTECTED] wrote:
#1 - It's down
#2 - The owner is prohibiting anyone from mirroring it.
Have you checked the message on voip-info.org recently?
http://voip-info.org/
Voip-info.org is down due to a hardware failure.
Will be back soon.
Due to the kind offers of mirror
Hi every body,
I've set up a Trixbox Server with TE110P,all things seem to work
fine(Thank You Malling lists irc Forums), but i need your help,
i ve 30 numbre from 60 to 89, i need to specify for each sip extension
a Zap number
for example to call the sales service the caller must call 555-4570
Excelent! Then once it comes up voip-wiki.us will be glad to provide a
read-only mirror.
On 3/15/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 3/15/07, Matt [EMAIL PROTECTED] wrote:
#1 - It's down
#2 - The owner is prohibiting anyone from mirroring it.
Have you checked the message on
Mark Quitoriano wrote:
Hi i have an asterisk pbx with E1 port connected to another PBX. Im
trying to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get the number 2 as it's DNIS when i change it to ZAP/1
or ZAP/g1
I may be able to get my hands on a few of these units as we are phasing them
out at a company. I could not find much in the way of connecting these to non
Intertel systems.
Anyone have an idea or success with this?
Thanks!
Bill
___
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Hi Younss,
You just need to setup Inbound Routes in FreePBX. The inbound routes allow
you to route calls based upon caller ID or DID. Since you want to route
based upon the number your caller dialed, you want to route based on DID.
For your example:
1. Create a new inbound route.
2. In the DID
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of nik600
Sent: Wednesday, March 14, 2007 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call center manager for Asterisk (Release
0.3)
Of course you should buy a horse. But then there are the questions
like. Do I get one like the Budweiser ones? Or just a mule (they can
be helpful). What about color? Maybe a spotted one? Will my horse be
able to talk to other horses using SIP? Or will it only be able to use
IAX? Man, so many
Hi,
Thanks for the pointer.
I will check previous threads (as I've not found yet any sendText compliant
hardphone).
Cheers
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You'll have to check the horse-wiki and pray it never goes down.
Alternatively, you could get a Cisco horse. While it may cost more, at
least you'll have a number you can call for tech support should your
horse throw a shoe.
The downside being, of course, if you want to modify your horse
Hey all, I have a question for those administrating/building out
systems with over 30 users on them. How often do you experience
the dropped call phenomena. Would you care to share your
experiences including what versions of * you were using, what
kind of connectivity was present (T1, Fractional
On Wed, 14 Mar 2007, Stephen Bosch wrote:
Gordon Henderson wrote:
On Wed, 14 Mar 2007, Jonathan k. Creasy wrote:
I would be willing to mirror it also?.
At the risk of sounding like an AOLer, Me Too ... (UK based mirror?)
The site is pingable, so I'd suggest it's either crashed in some
somebody can help me with this message
I don´t understand
*CLI stop now
Beginning asterisk shutdown
Executing last minute cleanups
== Destroying musiconhold processes
Yuck! Error in buffer handling...: Connection reset by peer
Yuck! Error in buffer handling...: Connection reset by peer
Nik,
This looks REALLY COOL!
thanks
Just an FYI in case you didn't know, there is also a callcenter asterisk
mailing list that you could post this to. I am not sure how many users
are subscribed but it is most certainly more of your target audience.
thanks, i'll subscribe on it.
At any
I've got 415 phones, mostly Cisco 7960's. The only time I see
dropped calls is when either end hangs up, or I restart asterisk. Using
all T1 PRI.
HW mainly: Dell 1750 w/2GB, Digium TE410 or TE412P's. Raid1 w/PERC.
I use Dell 1950's for the VM servers, but anything with a Digium card is
a
On 3/14/07, Steve Totaro [EMAIL PROTECTED] wrote:
Just an FYI in case you didn't know, there is also a callcenter asterisk
mailing list that you could post this to. I am not sure how many users
are subscribed but it is most certainly more of your target audience.
Where do you subscribe to
i haven't found any call center asterisk mailing list, but i've found this:
http://lists.digium.com/mailman/listinfo/asterisk-biz
On 3/15/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/14/07, Steve Totaro [EMAIL PROTECTED] wrote:
Just an FYI in case you didn't know, there is also a
Hi -
I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't
make it work- currently, asterisk will not startup because of a bad
module. Below are some log files/config files. If anyone has any
suggestions, I'd appreciate it.
I used Trixbox 2.0 and followed instructions on
Michael Collins wrote:
I would suggest that we create a new wiki, make it solely for Asterisk
topics, as not to offend or replace voip-info. Build mirrors to
multiple sites and multiple domain names. This would give this
community a second resource with redundancy which is what I think ALL
Does anyone know if I can get Incoming caller id name and number on a
sagnoma PRI.
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Rob Vinson wrote:
Does anyone know if I can get Incoming caller id name and number on a
sagnoma PRI.
Yes.
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That should be provided by your telco, if your referring to a PRI on a
Sangoma T-1 card.
On 3/15/07, Rob Vinson [EMAIL PROTECTED] wrote:
Does anyone know if I can get Incoming caller id name and number on a
sagnoma PRI.
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Will this do the intended thing?
This is in sip_nat.conf which is included in sip.conf:
externip=192.168.0.200
localnet=192.168.0.200/255.255.255.0
externip=64.168.237.110
localnet=192.168.1.2/255.255.255.0
I have Asterisk running on a box with two Ethernet interfaces and bound to
both. One
Sangoma gives excellent support
Suggest you try there first
They probably will want SSH access to the box.
Send them an e-mail
John Novack
Todd H wrote:
Hi -
I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't
make it work- currently, asterisk will not startup because of a
Patrick wrote:
Thanks for the info Trevor. Was your proof of concept also with Windows
PCs or *nix PCs? I haven't played with realtime yet so I might be in for
a bit of a learning curve.
This was just on Linux user stations with a simple bash script that send
a request to a web server.
Rob Vinson wrote:
Does anyone know if I can get Incoming caller id name and number on a
sagnoma PRI
The bigger question is if your telco is sending it to you. asterisk
generally takes care of everything automatically, provided it's
available and you've configured your PRI properly. Number
On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Thanks for the responses.
iptables on the * box has no rules and all tables default to 'accept.'
I have not got to the point of placing calls out across the internet
yet. The issue here is no audio back from the * box when running
through
On 3/15/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Ok. I have not been able to setup the box to call outside, however,
watching the packet traffic I see plenty of data flowing from the
xlite client to the * server, but never any packets from the server to
the client. (That is, during the
I would take an alternative stance and say that an Asterisk only solution is
needed.
This is a wildly growing product with nearly limitless possibilities.
Trying to cram too much on a site just causes confusion.
KISS (no I am not calling anybody in particular stupid.)
On 3/15/07, Davis
Ok so I read the Linux 2.6 related README and finally
compiled propperly, I thought but at the end I notice
that lscpi does report the cards, but I cant modprobe
wcfxo nor zaptel and I do have wcfxo.ko in the
/lib/modules/2.6.8/extra/ directory, so what gives?
This is a Debian Sarge, thanks!
Exactly, Sangoma support is THE BEST! :)
Best Regards
Josué
2007/3/15, John Novack [EMAIL PROTECTED]:
Sangoma gives excellent support
Suggest you try there first
They probably will want SSH access to the box.
Send them an e-mail
John Novack
Todd H wrote:
Hi -
I just got an A200 card
I couldn't agree more.
-Original Message-
From: Josué Conti [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Cc:
Sent: Thu, 15 Mar 2007 20:31:53 -0300
Delivered: Thu, 15 Mar 2007 20:20:22
Looks like the site is back up. Don't all hit it at once, it might go down
again ;-)
Sean
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That's awesome, we were nearly done with the spider too!
On 3/15/07, Sean Bright [EMAIL PROTECTED] wrote:
Looks like the site is back up. Don't all hit it at once, it might go down
again ;-)
Sean
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On 3/14/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
The third field, in my case Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]
is the channel to announce the parked call slot to. In my case,
extensions beginning with 1xx are the phones themselves, and extensions
4xx are the same phones
I figured it out by examining the log files. The following works but not
as hypothesized above.
1. the second externip overrides the first, so externip can only be
specified once
2. the first localnet and the second localnet are BOTH understood and used.
3. asterisk tests the destination IP
Sean Bright wrote:
Looks like the site is back up. Don't all hit it at once, it might go
down again ;-)
...and now...
mirrormirrormirrormirrormirrormirrormirrormirrormirrormirrormirror
-stephen-
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Hello,
We have an asterisk setup at our client's site using a TE205P. The line to
telco is a 23 channels T1 PRI, however the line has random echo problems
(about 5-10% of the calls)!
Can anybody tell me if echo cancellation is really needed even at a T1 PRI
to the telco? Because people keep
On Thu, Mar 15, 2007 at 03:38:20PM -0700, pedro noticioso wrote:
Ok so I read the Linux 2.6 related README and finally
compiled propperly, I thought but at the end I notice
that lscpi does report the cards, but I cant modprobe
wcfxo nor zaptel and I do have wcfxo.ko in the
We had 2 of 3 SCSI drives fail in a RAID a couple of weeks ago - its
hard to explain that to a customer!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of cb
Sent: Thursday, March 15, 2007 12:00 AM
To: Asterisk Users Mailing List - Non-Commercial
channels?
--
Thanks Regards,
Vidura B. Senadeera.
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Message: 16
Date: Thu, 15 Mar 2007 10:35:16
On 3/14/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
There is no need for any 'map'; any Asterisk 1.2.x release should be
usable with any Zaptel 1.2.x release, but of course we'd suggest using
the latest releases of both. There are no API changes or feature
additions (generally) in release
Hi Vincent -
Can anybody tell me if echo cancellation is really needed even at a T1 PRI
to the telco? Because people keep saying when they deploy voip solution in
Hong Kong using T1 PRI, there is no need of echo cancellation. (even the
local Digium distributor)
I have to do echo cancellation
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