Michael,
After many months of search we decided to develop an in-house solution
for such kind of needs. For a month our solution is in production and
does everything you mentioned below. Asterisk's built-in call queue does
not provide many of the features necessary for large organizations.
Idris
Yes - we even have a server at a clients site with 2 TE410P's in it as
an interim measure.
PaulH
On Wed, 2006-10-11 at 20:01 -1000, Thermal Wetland wrote:
> Does anyone know if you can have multiple TE110P cards in one chassis?
>
> -Thermal
> ___
> -
Does anyone know if you can have multiple TE110P cards in one chassis?-Thermal
___
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Jessee J Holmes wrote:
> As far 1.6.7 firmware supporting multiple presences (48 i think), maybe
> I was wrong on that; however, I remember reading the 2.0.1 firmware
> release notes and they mentioned that feature was fixed within the 2.0
> firmware. Maybe they fixed it before that and just never
Hello ALL!
1. eviroment:
1) PBX:[EMAIL PROTECTED](Asterisk 1.2.7.1)
2)Media GateWay:Grandstream HT 488(register to pbx using SIP account,such as
9001),connect to PSTN(phone no is 82820088) using its FXO.
3)Queue:set up a queue named [myqueue],dial 2020 can call queue
2.my question:
whe
Hello,
On Tue, 2006-08-29 at 15:52 +0300, Mindaugas Kezys wrote:
> Hello,
>
> Kolmisoft: http://www.kolmisoft.com released new versions of MCC and MOR -
> Billing solutions for Asterisk PBX
>
> MOR - is new product, it's MCC v2. Rewritten to support MySQL and is based
> on Ruby on Rails.
>
Wh
Analog routes (ie. copper telco lines) do not have DID information on them. Only digital lines (PRI, often VoIP DID) have this information sent alongside the call. What you need to do is change the context for each port you want redirected in
zapata.conf, like so:signalling=fxs_kscontext=from-pstn
Then you have something wrong. Inuse should always be 0 if the card is not in use. Each time a call is in process, the agi will change that +1 and when the agi is complete it will -1.
I don't think that FOP will work without a ton of modifications unless someone at A2Billing has a patch or someth
I am an asterisk newbie. I have successfully installed
asterisk on Freebsd. The problem I am having is when I try to route based upon
incoming DID. CALLERID(dnid) nor CDR(dst) have a number in them. Please
help.
Thanks
___
--Ban
Seems like we got another person that misses words written in plain English.
On 10/11/06, J. Oquendo <[EMAIL PROTECTED]> wrote:
C F wrote:
> OK, I'll agree with you that I'm looking at a point of view from
> Enterprise lever and not carrier level, BTW, NFAS for redundancy is in
> most cases a w
-= 464 extensions (2241 priorities) in 151 contexts. =-
Very small. 8 users, but 7 companies. 4 users work for one company, 1 user works for another, 3 users work for 3 companies, 2 users work for 1 company, and 1 user works for 1 company. Several users work for multiple companies if you're in
have you verified that dma is enabled on all of your harddisks?
hdparm -d /dev/xxx
Jason
The place where you made your stand never mattered,
only that you were there... and still on your feet
Boyd Goodin wrote:
> Hello everyone!
>
>I have a slight problem. Running * (trixbox) on a P4 3.0gh
# transfer is actually depreciated, replaced by default with #1, see
features.conf for more details.
On 10/10/06, MF <[EMAIL PROTECTED]> wrote:
Hi all
I have a call that comes in via a first E1 and goes out via a second
one, my problem is I need to catch a digit dialed by the second one,
duri
Hi guys,I've been installing Asterisk 1.4 with Asterisk addons, and i could notice that in /usr/lib/asterisk/modules/ doesn't have cdr_addon_mysql.so even after compiling Asterisk Addons!In fact the cdr_addon_mysql.c exists, but it doesn't seems to be compile when i run Asterisk-Addons: make && mak
I was using timeouts. The dial plan was altered and working by the time of the replies, so I'm sorry I can't show the original problem.It's possible I did something simple like not waiting the whole timeout.Thanks,
MarkOn 10/10/06, Dovid B <[EMAIL PROTECTED]> wrote:
Simple
Exten => 1234,1,D
C F wrote:
> OK, I'll agree with you that I'm looking at a point of view from
> Enterprise lever and not carrier level, BTW, NFAS for redundancy is in
> most cases a waste of money (again enterprise POV), since if one T1 is
> down usually all of them from the same provider will be down.
You must
On 10/11/06, Jason Aarons (US) <[EMAIL PROTECTED]> wrote:
I thought XO was reselling Level 3s (old Genuity assets) network/voipjust like Qwest ?
We're on XO in Houston, and it is definitely not Level 3. This is an XO network here.
I also have a Covad SDSL account, it hits Level 3 pretty quick
On 10/11/06, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
On which topic: do *you* know who to call and what to tell them to getyour lead DID forwarded to your cell phone when your span (or switch)
goes down?
Actually, Jay, sure don't. I don't know what the answer to that would be. Wish I did, a
What are your T.38 plans with this?
On 10/11/06, Olle E Johansson <[EMAIL PROTECTED]> wrote:
Friends in the Asterisk community,
I've been talking for years about the new version of the SIP channel.
I've been trying to get funding
and get going. Well, the funding part remains to be handled, but
Are you using mpeg123 for format_mp3? Does it happen just with the
queue, how about with WaitMusicOnHold
On 10/11/06, Boyd Goodin <[EMAIL PROTECTED]> wrote:
Hello everyone!
I have a slight problem. Running * (trixbox) on a P4 3.0ghz pc with
1.5gig of ram, pbxhardware.com T200P Tor2 dual t-
Please read these with care:
Causes of Echo http://www.voip-info.org/wiki/view/Causes+of+Echo
Asterisk Echo Avoidance: http://www.voip-info.org/wiki-Asterisk+Echo+Avoidance
Until you understand echo it will be impossible to eliminated it.
Notice that you have a digital connection and your SIP p
After working with NEC systems for more than 10 years, both as a
technician and as an end user, I can say with confidence that their
stuff just doesn't break. Period. You can kill it by installing it in
an unventilated phone closet, outside and exposed to 110F degree Fresno
summers, but even then
Your profile shows that you are running 5ESS custom PRI. Is that what your service provider told you to run? The reason I ask is that the 5ESS can run "custom" or NI-2. I mention this is that the lack of RESTART or RESTART ACK messages isn't telling the service provider's switch that the 24th B cha
No I meant to say agi, because I haven't seen from plain dialplan
using a stable version, that asterisk should *just die*. But then
again, I only have one system that uses agi, and that one hasn't
crashed yet, in fact it's been on and up since I installed it - System
uptime: 9 weeks, 2 days, 4 hou
Hi Moises, Ignore my last reply about the presence of the DTMF in 1.4.
Do you have the source for patching the DTMF event?
There is no link to it on the bug6082 page, and I am not quite sure
how it can be obtained from SVN.
Regards
Richard
On 10/12/06, Frank Church <[EMAIL PROTECTED]> wrote:
On Wed, Oct 11, 2006 at 04:26:45PM -0500, Aaron Daniel wrote:
> That was kinda spiteful of you.
Well, I thought it was a bit rough, but I'm not sure it was spite...
> Not everyone has the same needs as you in their systems, especially
> those with only 50-60 users. Your view in the telecom world
I thought XO was reselling Level 3s (old Genuity assets) network/voip
just like Qwest ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 3:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
Hi Moises,
does the you mentioned earlier at
http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch include the DTMF
event, or is it for PlayDTMF and SendDTMF?
Looking through the actions on bug6082 it is hard to tell whether
the DTMF event patch is still in there when I last compiled that
branch t
On Wed, Oct 11, 2006 at 05:08:32PM -0500, Lacy Moore - Aspendora wrote:
>As a carrier, I would expect you to have an abundance of
>redundancy, but not an SMB. SMB's don't have the money to cover
>everything. That's what cellphones are for :-)
On which topic: do *you* know who to call a
The monthly Melbourne Asterisk get together is happening tonight!
Where: Pint on Punt, 42 Punt Road Windsor.
Time: Tonight 7:00pm.
As usual we'll be talking Asterisk and VOIP in general, and a special
discussion on our upcoming summer schedule.
later,
PaulH
___
On Wed, 11 Oct 2006, C F wrote:
I agree that if an asterisk box dies (I don't know how such a thing in
a well controlled stable system will happen, but I guess with a bug in
an agi it could happen, then that will be another reason not to use
AGIs for me) you need another one to take over, but ag
On Tue, Oct 10, 2006 at 10:57:54PM -0600, Douglas Garstang wrote:
> I see some awefully large dialplans here. Are people putting all this
> on one box or clustering it amongst a number of boxes? I think any
> business is going to be pretty annoyed if they suddenly lost access
> to 16,000+ extension
On Tue, Oct 10, 2006 at 03:09:11PM -0600, Steve Murphy wrote:
> > 400 extensions for a home system, that is ... extreme! :-)
>
> Not really. I have only 4 zap extensions, and two FXO lines.
>
>
> The extra dialplan logic does things like recording CID in a database,
> playing stuff over the spea
On Wed, Oct 11, 2006 at 09:13:26AM -0500, Jessee J Holmes wrote:
>I believe there is some RFC for presence out there that some people
>consider the "standard"; although, I'm not sure what this is...
XMPP; RFC 3920, et al.
Cheers,
-- jra
--
Jay R. Ashworth
I don't like following up with my own post but I figured out the
problem.
The server is running FC2, I did a 'yum -y update' and after it was
complete I rebuilt zaptel (make clean ; make ; make install) and
everything works fine now. Musta been something with the compiler ??
-Matt
O
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm having a hard time tracing calls that go through the parking lot. I
see the call placed from the original extension, and then it looks as if
it is hung up, then I see another extension grabbing a call from the
parking lot, but there are no fields i
Thanks guys!
I was hoping to let them leave a voicemail, then transfer to cell - in
case the user doesn't answer at least they get to leave a message.
On 10/10/06, Chris Ramsey <[EMAIL PROTECTED]> wrote:
I don't think you would need a macro for this. After Asterisk determines
that their first e
Mike best to do it yourself so you can make moves and changes as you
see fit "on demand" rather than relying on a carrier to be available
in a mid afternoon rush etc.
Also means that you can deliver calls to your system from pstn or maybe
a 1800 voip service etc or even multiple locatio
On 10/11/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, October 11, 2006 10:59 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How big is *your* dialplan??
>
>
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Colin Anderson wrote:
> what happens when you drop your gains? use /etc/asterisk/zaptel.conf and
> fiddle with tx and rx values. Works, most of the time.
>
> -Original Message-
> From: John McEntee [mailto:[EMAIL PROTECTED]
> Sent: Wednesday,
Diego, this is an english mailing list, there is no need to post in
spanish the same message.
Your error is due to missmatching versions between libmfcr2 and
spandsp. Downgrading spandsp will fix the problem.
Regards
On 10/11/06, DiegoF <[EMAIL PROTECTED]> wrote:
hola a todos de nuevo, tengo e
On Wed, Oct 11, 2006 at 02:52:26PM -0500, DiegoF wrote:
>
> they span is E1, I have still not connected it, although I will connect to a
> PBX panasonic
> this is what leaves in /proc/zaptel/
>
> Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/
>
> 1 WCT1/0/1 CAS
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 11 Oct 2006, at 13:19, Stefan Tichy wrote:
On Wed, Oct 11, 2006 at 11:32:57AM -0400, Jens Vagelpohl wrote:
The call file created by the outgoing script "file2fax.py" specifies
3 retries in case of failure.
Fax may fail even if the phone call
On Wed, Oct 11, 2006 at 07:51:46PM +0100, David Bath wrote:
> As further info, here's the tail of the verbose logging (as enabled in
> logger.conf). I have the complete log (but there are lots of irrelevant
> SIP transactions for other phones/providers) which I can send if it
> becomes helpful.
>
> No one's system is redundant? :O
Was the Lucent Merlin Legend system I replaced redundant? I don't think so. What about any other proprietary system for SMBs? I don't think so. I'd guess a Dell Server based on Linux and Asterisk is a lot more redundant and easily replaceable. I can get it
Local government office with approximately 100 sets (going to 600):
593 extensions (1241 priorities) in 88 contexts
CP
On 10-Oct-06, at 1:16 PM, Steve Murphy wrote:
Hello!
In my relentless quest for knowledge, I pose this question: who's got
the biggest
dialplans, and how big are these monst
I am trying to write a script to attempt to make a call on a
Zap channel, and if it fails, send an alarm. I can generate the call, but
because the Zap channel accepts the call, even though the other end never
answers, it sees it as a successful call, which it isn’t.
Anyone have any ide
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, October 11, 2006 2:00 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] How big is *your* dialplan??
>
>
> On Wednesday 11 October 2006 13:23, Douglas Garstang wrote:
> > No o
That was kinda spiteful of you.
Not everyone has the same needs as you in their systems, especially
those with only 50-60 users. Your view in the telecom world is going to
be WAY different than those that only run a smaller system.
Some people don't view their voice traffic as being as important
Hi,
I've been asked to put together a quote for a system which basically
will be a virtual PBX based on Asterisk with some IVR's and a whole
bunch of GotoIfTime's.
There will be one incoming DID via SIP, user gets dropped in to an IVR
and then depending on the option they choose, *and* the time o
OK, you are all correct, but it's still a viable option for use with
Asterisk.
Michiel van Baak wrote:
On Oct 11, 2006, at 7:13 PM, Joe Dennick wrote:
The X-Ten is probably the most know "free" soft-phone availible. You
can find it at
http://www.xten.com/index.php?menu=Products&smenu=xli
You could use heartbeat http://www.linux-ha.org (or ultramonkey
http://ultramonkey.org). With this you set up a director that shares
the load to multiple servers. You can even set it to have consistent
connections so a originating IP will return the the same server. I
have hearbeat running on t
Never tried it but would not reccomend installing
asterisk on a windows box for many reasons. Just google windows
sux.
- Original Message -
From:
Naidu, Vijay
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 11, 2006 6:42
PM
Subject: [asterisk-users]
Depends on your system hardware.
- Original Message -
From:
Don
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 11, 2006 7:17
PM
Subject: [asterisk-users] max users
Whats the max headcount you can have in a
conference bridge using ztdummy...si
Hello everyone!
I have a slight problem. Running * (trixbox) on a P4 3.0ghz pc with
1.5gig of ram, pbxhardware.com T200P Tor2 dual t-1 card, everything on it's
own irq, and about 22 Sip phones (either Grandstream ATA or Budgettone 102
units). IF I have 5 or less calls in queue, MOH and annouc
what happens when you drop your gains? use /etc/asterisk/zaptel.conf and
fiddle with tx and rx values. Works, most of the time.
-Original Message-
From: John McEntee [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discu
Steve Totaro wrote:
It did but I was getting no audio on those channels so I removed them in
hopes that the telco would not try to send call to those channels as a
temporary fix while I track down the cause of the problem. How can I
just busyout those channels (48,72,96) so that calls are no
> > Sometimes the internet connection is dropped and asterisk doesn't do a dns
> > lookup and provider re-rest quickly enough so all calls are going out via
> > expensive ISDN.
>
> So detect a connection change and then restart, by the way of 'asterisk
> -rx restart now' (or 'restart when convi
On Wednesday 11 October 2006 13:23, Douglas Garstang wrote:
> No one's system is redundant? :O
Is your Norstar MICS redundant? How about an NEC Electra?
I'd put good money on the VAST majority of SMB's phone systems NOT being
redundant, and maybe only 60% of them being on any kind UPS, with m
el span es E1, todavia no la he conectado, aunque la idea es conectarla a una pbx panasonicesto es lo que sale en el /proc/zaptel/they span is E1, I have still not connected it, although I will
connect to a PBX panasonic this is what leaves in /proc/zaptel/Span 1: WCT1/0 "Digium Wildcard TE110P T1/
I am little confused on load balancing, when asterisk server is also a sip
client.
Based on these,
XO Communications one of the largest US DID Provider, now offer SIP
Orignation Services for wholesale.
Verizon Communications One of the largest US Teleco, now offer SIP
Orignation Services.
That mea
I think at best its a bug with the safe_asterisk script and at worst
could be a bug with asterisk itself. I can't see how this is a
configuration issue with freepbx.
---
Shidan
On 10/11/06, Dinesh Nair <[EMAIL PROTECTED]> wrote:
On 10/11/06 21:15 Joseph said the following:
> I quits on my as
I am using a2billing as billing software ,and I make an 800
call service which means that the destination extension should be build
I put this code at extensions.conf
exten => 99909994,1,SetAccount(2704714849)
exten => 99909994,2,Wait,2
exten => 99909994,3,DeadAGI(a2billingp.php)
ext
On Wed, Oct 11, 2006 at 09:21:38AM -0800, Mojo with Horan & Company, LLC wrote:
> Conceivably, if only one SIP UA were in use behind a NAT router, then
> when it constructed a call and needed to receive RTP streams, it would
> configure port mappings in the router via the UPnP protocol, so extern
> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, October 11, 2006 10:59 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How big is *your* dialplan??
>
>
> Douglas, it seems to me that you don't understand how th
On Oct 11, 2006, at 7:13 PM, Joe Dennick wrote:
The X-Ten is probably the most know "free" soft-phone availible.
You can find it at
http://www.xten.com/index.php?menu=Products&smenu=xlite
Free != GPL
xlite is still a closed product that you can use for free.
But you cannot get the sources
Well, IF you had stated that in your
OP, you might not have gotten this reply! How does this reply help
anyone?
Your mother wouldn't be proud of your attitude.
John Novack
Steve Totaro wrote:
This
reply helps me how?
Of course I am pursuing the issue through their support channel.
I am running a server with a Digium TE410P, about 40 grandstream gxp-2000's, 10 Polycom 500's, running svn branch 1.2 rev 44144, and FreePBX. The server only has one PRI at the moment, and at times all 23 channels are full, the phones are all on the local network with the server. Every day or two a
I am little confused on load balancing, when asterisk server is also a sip
client.
Based on these,
XO Communications one of the largest US DID Provider, now offer SIP
Orignation Services for wholesale.
Verizon Communications One of the largest US Teleco, now offer SIP
Orignation Services.
That mea
[EMAIL PROTECTED] wrote:
Hello all,
we want to use asterisk queues for a call center application. Depending on
the average waiting time in a queue, we want to make a decision to either
enqueue a call or transfer it to another site.
Are the applications available to query the average waiting tim
As further info, here's the tail of the verbose logging (as enabled in
logger.conf). I have the complete log (but there are lots of irrelevant
SIP transactions for other phones/providers) which I can send if it
becomes helpful.
NB. The mysql server was down for maintenance at the time, so the cdr
Kristian Kielhofner wrote:
Steve Totaro wrote:
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on
channels 48,72, and 96 have no audio. I tried removing these
channels fr
On 10/11/06, Dinesh Nair <[EMAIL PROTECTED]> wrote:
On 10/11/06 21:15 Joseph said the following:
> I quits on my as well, when I try to make a second call.
> There is a bug report on it:
> http://bugs.digium.com/view.php?id=7972
this seems like a configuration error within FreePBX and isnt rea
Xlite is not GPL!
Joe Dennick wrote:
The X-Ten is probably the most know "free" soft-phone availible. You
can find it at
http://www.xten.com/index.php?menu=Products&smenu=xlite
Gregory Duchatelet wrote:
Hi,
I’m searching for GPLed softphones. I found WengoPhone but actually
not available
Lol - use a real PC maybe :P
Cheers,
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tim Panton
> Sent: Wednesday, 11 October 2006 1:02 PM
> To: asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-u
-= 1967 extensions (2838 priorities) in 285 contexts. =-
Shared services PBX with a dozen or so customers.
-ejay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy
Sent: Tuesday, October 10, 2006 3:17 PM
To: asterisk-users@lists.digium.com
Subj
OK I have been battling with echo problems with asterisk on ISDN for a
few weeks now, and still can't solve it (although I think I have tried
everything I can find.)
I will try a post everything I think is possibly relevant that I can
remember with the hope someone can point me in the right di
Asterisk can only be the proxy/server for MGCP, you connect other
devices to it. Asterisk can not be a user agent connecting to other
MGCP server.
On 10/11/06, Paul Ianas <[EMAIL PROTECTED]> wrote:
Hello everybody!
I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol.
What
> Hi,
>
>
>
> I'm searching for GPLed softphones. I found WengoPhone but actually not
> available for Asterisk PBX, only for Wengo network. I found Kiax but only
> for IAX protocol.
>
>
>
> Did you know a good GPLed softphones which works on Windows ?
>
>
>
> Thanks
>
> Greg
>
>
Apparently (from
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote:
> I lost my internet connection today for a short time.
> During that time 1.2.12.1 stopped talking to my phones.
> Asterisk was still working as I got 2 voicemails. I have TDM analog
> cards for incoming calls.
>
> Anyway my cisco phones had X
hola, este lo copie de internethello, this it copies it of Internetspan=1,1,0,cas,hdb3cas=1-15:1101dchan=16cas=17-31:1101
loadzone = usdefaultzone=usthanksOn 10/11/06, Giorgio Incantalupo <[EMAIL PROTECTED]
> wrote:Hi DiegoF,I had a similar problem, it was a zaptel.conf misconfiguration. Maybe
for
Hello,
I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I
installed the following
-rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0-
beta2.tar.gz
-rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0-
beta1.tar.gz
-rw-r--r-- 1 root root80019
Title: Re: [asterisk-users] How big is *your* dialplan??
No
one's system is redundant? :O
-Original Message-From: Douglas Garstang
[mailto:[EMAIL PROTECTED]On Behalf Of Douglas
GarstangSent: Tuesday, October 10, 2006 10:58 PMTo:
Asterisk Users Mailing List - Non-Com
H,
hugolivude wrote:
For various reasons, I'm not too partial to UPnP, but maybe there needs
to be a SIP UA that uses UPnP to configure a NAT router for it, when an
RTP stream is begun?
Not following this part...
While I could probably never bring myself to enjoy (Microsoft's?)
Universal Plug
On Wed, Oct 11, 2006 at 11:32:57AM -0400, Jens Vagelpohl wrote:
> The call file created by the outgoing script "file2fax.py" specifies
> 3 retries in case of failure.
Fax may fail even if the phone call was successfull.
> This just retries it within Asterisk, I
> don't know if I could have c
Whats the max headcount you can have in a
conference bridge using ztdummy...since it is all sip based
incomming?
Don
___
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To UNSUBSCRIBE or update options visit:
On Wed, Oct 11, 2006 at 10:29:44AM -0500, DiegoF wrote:
>
> Hello to all, I have a question. I am installing te110p, when I give ztcfg
> him - v leaves the following error to me
>
> ZT_CHANCONFIG failed on channel 25: No such dev
The X-Ten is probably the most know "free" soft-phone availible. You can
find it at
http://www.xten.com/index.php?menu=Products&smenu=xlite
Gregory Duchatelet wrote:
Hi,
I’m searching for GPLed softphones. I found WengoPhone but actually
not available for Asterisk PBX, only for Wengo networ
On Wed, Oct 11, 2006 at 06:23:48PM +0200, Gregory Duchatelet wrote:
> Hi,
>
>
>
> I'm searching for GPLed softphones. I found WengoPhone but actually not
> available for Asterisk PBX, only for Wengo network.
Have you actually tried it? Were you actually able to build it?
> I found Kiax but o
On Wed, Oct 11, 2006 at 11:25:08AM +0200, Remco Barendse wrote:
> On Wed, 11 Oct 2006, Tzafrir Cohen wrote:
>
> > On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote:
> > > Hi list!
> > >
> > > I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
> > > something to
Has anyone built and run asterisk 1.4 beta2 on an intel mac?
Did it work?
I've got it building ok (once I installed Xcode, wget and bison)
However Asterisk hangs on startup (halfway through loading the modules).
I have not (yet) had time to debug it, but I wondered if anyone else had
done this
Douglas, it seems to me that you don't understand how the extensions
of an asterisk dialplan relate to real life. As an example:
-= 135 extensions (657 priorities) in 31 contexts. =-
This from a box (yes one box) that has just 10 phones, and 6 lines.
Every s extension is considered an extension. W
hola a todos de nuevo, tengo el siguiente error cuando compilo el libunicall despues de compilar spandsp y libsupertone. esto es en fedora 5hello to all, I have the following error again when I compile
libunicall after compiling spandsp and libsupertone. this is in
fedora 5testcall.o: In function `
Dean,Tough call ... I haven't played with an IP 500 in a long time now and all that I know is Polycom officially doesn't support them.I'm sure the 2.0.1 firmware wasn't designed to ever work with bootroms 2.xx. I'm sure the problem lies with either the phone not supporting it or the bootrom not acc
Issac Simchayof wrote:
Polycom 601 with Sip 2.01
Anyone using Sip 2.01? I have upgraded my phones and now presence no longer
functions.
Buddy list shows all phones online but status does not change when someone
is on a call. Also blf does not function.
I am using trixbox, 1.67 was working fine
Dear Issac,Makes sense.We got asked about moving back to firmware 1.6.7 as well and the official answer from Polycom is "not a problem"! Put the firmware on your server and remove the 2.0 firmware from this server and when the phone reboots it will grab the 1.6.7 firmware and load it on the phone.
Hi,
I had a question. I am installing Asterisk on a windows
machine – Astwind. I was wondering if it works with Dialogic card or if
it needed only digium card. Is there anyway Asterisk can work with a Dialogic
card or a Pika board?
Thanks in advance.
Vijay Naidu
"Never Interrup
Steve Totaro wrote:
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on
channels 48,72, and 96 have no audio. I tried removing these
channels from zapata.conf with hopes th
Hello all,
we want to use asterisk queues for a call center application. Depending on
the average waiting time in a queue, we want to make a decision to either
enqueue a call or transfer it to another site.
Are the applications available to query the average waiting time of a queue,
if possible f
Hi C.,
Check out the "pickupgroup" and "callgroup" options in sip.conf -- these
should accomplish what you're looking for:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
More about this feature is defined here:
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+p
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