Thanx alot for the tips.i'll try then out and let u know about the result
On 7/6/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:
Yep, forgot 'bout that.
Or you could use web-meetme, it has this feature.
On 7/6/06, Alexander Lopez <
[EMAIL PROTECTED]> wrote:
Snip, snip. Chop Chop.
__
The server name in the caller ID is a little
annoying
Cisco have removed that in the latest firmware
On 7/6/06, Ryan Amos <[EMAIL PROTECTED]> wrote:
The 8.2 firmware works just fine (asterisk 1.2.6). I have had exactly
zero problems with it, nothing even weird about it. Pretty trouble-free
IM
bram kortleven wrote:
We are currently looking for a way to easily configure a 'auto attendant'
system on our asterisk pbx.
More in detail, I'm looking for a webbased (or something similar)
configuration generator, that has a feature like asking me how many 'menu
levels' I want, what text to play
On Thu, Jul 06, 2006 at 03:32:04PM -0400, C F wrote:
> I have recently build 2 machines, one with an Intel Pentium Dual Core
> CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
> a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
> HDDs. Here are the show tra
We are currently looking for a way to easily configure a 'auto attendant'
system on our asterisk pbx.
More in detail, I'm looking for a webbased (or something similar)
configuration generator, that has a feature like asking me how many 'menu
levels' I want, what text to play, and in the first, how
C F wrote:
cat /proc/cpuinfo on amd:
cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu family : 15
model : 47
model name : AMD Athlon(tm) 64 Processor 3200+
stepping: 2
cpu MHz : 2000.000
cache size : 512 KB
fpu : yes
fpu
> Hello everyone,
> I'm trying to set up an Asterisk machine with a quad-port BRI
> Junghanns card, and I want to use the mISDN drivers.
>
> I'm having some trouble configuring it: do I need to use CAPI drivers?
> I haven't found good links, could you please provide some info?
At what point are y
I have problems posting ?!
m2f
Read this topic online here:
http://forum.globalvoicenet.com/viewtopic.php?p=114#114
m2f
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cat /proc/cpuinfo on amd:
cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu family : 15
model : 47
model name : AMD Athlon(tm) 64 Processor 3200+
stepping: 2
cpu MHz : 2000.000
cache size : 512 KB
fpu : yes
fpu_exception
On Fri, 2006-07-07 at 03:32 -0400, mike wrote:
> thank you very much !
> i've tryed the rtp range ports definition with iptables with effort,
> i'll try this asap !
For a quick reference try this site:
http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c6s5.html
You may need to tweak some
On Thu, 06 Jul 2006 19:34:01 -0400, Brian Capouch <[EMAIL PROTECTED]> wrote:
>Thomas Kenyon wrote:
>>
>> For some reason when I do this, It only works if I have callerID
>> switched off, otherwise I get authentication errors.
>>
>
>Do you know of anyway to bulk-save the contents of all the confi
Snom 360
On 7/6/06, Fabio <[EMAIL PROTECTED]> wrote:
Polycom 300, SPA 841/941 (841 is out of market...), pap2 (2 x fxs ata)
Fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Shaun
Enviado el: Jueves, 06 de Julio de 2006 04:45 p.m.
Para: asterisk-users@l
Hi Luca,
are you using SIP reinvite ?
post a bit mor information (sip.conf)
Fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Luca Corti
Enviado el: Jueves, 06 de Julio de 2006 01:59 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asu
Polycom 300, SPA 841/941 (841 is out of market...), pap2 (2 x fxs ata)
Fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Shaun
Enviado el: Jueves, 06 de Julio de 2006 04:45 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] for you guys se
SPA941's and 7960's
On Thu, 2006-07-06 at 12:44 -0700, Shaun wrote:
> What brand/model phones are you using.
>
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Hi CF,
please could you to include CPUs specs, thanks in advance.
Fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: Jueves, 06 de Julio de 2006 04:32 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asteris
Do you have the "mpg123" utility in your system? By default, Asterisk
uses "mpg123" to play the mp3 files for music on hold.
-kokmeng.
Julian Varanini wrote:
Hi,
I am running asterisk 1.1. When a client is placed on hold from the
x-lite or polycom phone, no hold music is heard. I have
m
thank you very much !
i've tryed the rtp range ports definition with iptables with effort,
i'll try this asap !
thank you very much for your time !
.mike
On Fri, 2006-07-07 at 09:49 +1000, Nikolai Lusan wrote:
> On Thu, 2006-07-06 at 16:55 -0400, mike wrote:
> > i'm having a strange issue with an
Did you do 'make mpg123' at the time of Asterisk installation? And do you have ztdummy installed if you don't have digium hardware?
If you didn't do these steps, let me know and I'll send you step by step instructions how to install Asterisk properly. It took me months to learn how to make everyt
Julian Varanini wrote:
Hi,
I am running asterisk 1.1. When a client is placed on hold from the
x-lite or polycom phone, no hold music is heard. I have
musicclass=default set up in sip.conf and default exists in
musiconhold.conf. Has anyone had a similar experience? Any help would
be appr
On Thu, 2006-07-06 at 16:55 -0400, mike wrote:
> i'm having a strange issue with an asterisk box behind a firewall
> i'm trying to answer a sip call made to an asterisk box with a public ip
> from another asterisk box behind a firewall
>
> on the natted box i've put
>
> externip=195.110.XXX.XXX
>
Hi,
I am running asterisk 1.1. When a client is placed on hold from the x-lite or polycom phone, no hold music is heard. I have musicclass=default set up in sip.conf and default exists in musiconhold.conf. Has anyone had a similar experience? Any help would be appreciated.
Thanks
Julian
You are correct, I did ask the owner of the Coral to find out if it can
act as a client as well, which would be perfect and save the hassle of
going TDM. Do I need to be considered about QSIG when doing PRI
crossover?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECT
Ciao Tzafrir,
> > I'm trying to set up an Asterisk machine with a quad-port BRI
> > Junghanns card, and I want to use the mISDN drivers.
>
> That card is said to be exactly the same as the bero card. They both
> look strikingly similar, though have different PCI IDs.
Yes, I know..
> > I'm havin
I'm having a heck of a time keeping my Astertisk box up an running:
Redhat 9
Asterisk 1.2.7.1
Digium TDM400 w/ 1 FXO + 2FXS
1 g729 codec
I have my Sip.conf set up to renew registrations every hour:
maxexpirey=3600
defaultexpirey=3600
When I look in the /var/log/asterisk/messages tho, I se
Thomas Kenyon wrote:
For some reason when I do this, It only works if I have callerID
switched off, otherwise I get authentication errors.
Do you know of anyway to bulk-save the contents of all the config
screens on that unit?
If so, I could scrub the passwords and send you the config for
Bill Gibbs wrote:
Goal – to get the CoralIP PBX long distance savings by sending it to
Asterisk (which then talks via SIP to other long distance voip providers)
The Coral IP supports MGCP and so does Asterisk. Has anyone tried
sending calls from the Coral PBX to Asterisk via MGCP? I will be
p
We are receiving a large amount of dropped calls on our asterisk system.
After debugging I find the following line at the same time the call is
dropped.
(DEBUG[8882] channel.c: Got a FRAME_CONTROL (5) frame on channel)
I was unable to find very much information on this message. Just a quick
back
I could have told you that. I have 4 handy tones wasting in my basement.
-- Original message -- From: "calvis" <[EMAIL PROTECTED]> > Polycom 501 > > Grandstreams are junk. (I have had bad experiences with them) > > > > -Original Message- > From: [EMAIL PROTECTED]
I've used the Intelltouch ITC-3200's
(http://www.intellitouch.com/products/phones/ITC3002_features.html )
Pretty simple business looking phones. They are working out well. They lay
kinda flat however.
I got mine cheaper than the advertised regular and sale price.
The speaker phone however is not
Brian Capouch wrote:
> Douglas Garstang wrote:
>> Somewhat off topic...
>>
>> I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port.
>> I'm trying to get all calls forwarded to Asterisk. However, (and this
>> is hard to believe), the docs say that 1-stage calling (I presume
>> that me
Shayne wrote:
I'm currently having the same issue. If anyone has a solution to this
it would be greatly appreciated.
I have a room with 70+ agents logged and and am not seeing this.. One
thing to keep in mind is if you do a reload it will reset the queue and
start over
Kyle
--
CONFIDENTIALIT
I'm currently having the same issue. If anyone has a solution to this it would be greatly appreciated. On 5/26/06, Matt <
[EMAIL PROTECTED]> wrote:Hi,I'm trying to use Round Robin Memory with my queues. It seems to work
fine... that being I call in.. first time agent 1 will get a call,second time
Douglas Garstang wrote:
Somewhat off topic...
I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying
to get all calls forwarded to Asterisk. However, (and this is hard to believe),
the docs say that 1-stage calling (I presume that means no PIN is required) is
not poss
Goal – to get the CoralIP PBX long distance savings by
sending it to Asterisk (which then talks via SIP to other long distance voip
providers)
The Coral IP supports MGCP and so does Asterisk. Has anyone
tried sending calls from the Coral PBX to Asterisk via MGCP? I will be playing
aro
Hello,
I am using the TDM2400p and an outgoing call was joined
with an incoming call ONCE. Incoming should hit a Q and be
distributed. Is this bound to happen if the timing is
perfect with FXO lines? I am using kewlstart, no
busydetect settings.
I could change the outbound trunk from g to G, b
Ill do just that thanks
Warren (mailing lists) wrote:
> Pele Zico wrote:
>> Ive looked for cheaper mobile voip pstn calls in Nigeria and the cheapest
>> ive found is about 18p/min or 28c/min. Im looking at providing for
>> cpmanies here cheaper calls comparable to call cards. Can someone give
>
Did you try set autofallthrough=no. We have the same problem when using
1.2.9.1 (we are using A104d with IBM x306). So we downgraded to 1.2.6
and set autofallthrough=no. The call drop problem seems fixing. But we
have IVR DTMF recognition and queue not assign call to static agents
(Local channel) p
Do you have tetheral network analyser installed on server, that can be a good start, look at the analyses of the graphs. Also try pinging the CPE's and see if there is any latency. Do you also have the abilty to check the upstreams signals?
-- Original message -- From: "w
Somewhat off topic...
I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying
to get all calls forwarded to Asterisk. However, (and this is hard to believe),
the docs say that 1-stage calling (I presume that means no PIN is required) is
not possible with FXO-VOIP calls.
Sweet :)
Guess I should have looked harder, I didn't even know that app existed.
Thanks everyone!
On 06/07/06, Administrator TOOTAI <[EMAIL PROTECTED]> wrote:
Matt Gibson wrote:
> Hi,
>
> I'm experimenting with a little script here, and I'm tired of seeing
> my tests in the callerid logs.
>
>
try setting your dial plan in sip.conf using dtmf = rfc2833
-- Original message -- From: El Flynn <[EMAIL PROTECTED]> > Rizwan Hisham wrote: > > Hi, > > i need to set the dtmf mode on my quintum tenor a400 gateway. > > You might want to check the a400 manual on how to do
At 11:09 PM 7/5/2006 +0200, Jens wrote:
On 5 Jul 2006, at 19:00, John Kington wrote:
At 09:29 AM 7/5/2006 +0300, you wrote:
I have tollfree numbers with Nufone working OK.
But what I like most is the regular numbers with charge/month
but no charge/min on incoming calls...
Did your tollfree
thank you very much !
i'll try it asap
.mike
On Thu, 2006-07-06 at 15:37 -0400, Alexander Ginzburg wrote:
> in the rtp.conf you specify range of ports, this range should be
> forwarded on the firewall to the asterisk box.
>
> I have asterisk running on 192.168.0.250 ip and connect to broadvoice
Matt Gibson wrote:
Hi,
I'm experimenting with a little script here, and I'm tired of seeing
my tests in the callerid logs.
Is there a way to do something like the following:
exten => s,1,Answer
exten => s,n,DoNotLogCallData()
NoCDR()
--
Daniel
___
At 02:49 PM 7/5/2006 -0500, Rich wrote:
John Kington wrote:
At 09:29 AM 7/5/2006 +0300, you wrote:
I have tollfree numbers with Nufone working OK.
But what I like most is the regular numbers with charge/month
but no charge/min on incoming calls...
Did your tollfree number(s) with Nufone get c
How about this app:
NoCDR()
I.e.
exten => s,n,NoCDR()
-MC
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt Gibson
> Sent: Thursday, July 06, 2006 1:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [
Matt Gibson wrote:
Hi,
I'm experimenting with a little script here, and I'm tired of seeing
my tests in the callerid logs.
Is there a way to do something like the following:
exten => s,1,Answer
exten => s,n,DoNotLogCallData()
...
exten => s,1,Answer
exten => s,n,NoCDR
Doug
-- Ben Franklin
Hi,
I'm experimenting with a little script here, and I'm tired of seeing
my tests in the callerid logs.
Is there a way to do something like the following:
exten => s,1,Answer
exten => s,n,DoNotLogCallData()
...
I basically don't want anything inserted to mysql or master.csv
whenever this parti
Kevin Savoy wrote:
#
#AT&T
#
dynamic=eth,eth0/00:0C:42:03:63:0F/0,24,1
e&m=1-24
dynamic=eth,eth0/00:0C:42:03:63:0F/1,24,2
e&m=25-48
dynamic=eth,eth0/00:0C:42:03:63:0F/2,24,3
e&m=49-72
Totally alien to me. I was expecting to see something like:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
defa
No I have not using bristuff.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of stoffell
Sent: Thursday, July 06, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Channel not hanging up on Telco side
Good point. I am using Asterisk 1.2.9.1, Zaptel 1.2.6. libpri 1.2.3. I had
the same issue with Asterisk 1.2.7.1, Zaptel 1.2.5 and libpri 1.2.2.
My Zapata.conf looks like this:
[channels]
context=default
musiconhold=default
resetinterval=60
;AT&T T1's
group=1
switchtype=national
signalling=em_w
c
On 7/6/06, Kevin Savoy <[EMAIL PROTECTED]> wrote:
I'm having an issue where Asterisk hangs up a call (either party hangs up) but
the telco side
of the T1, both the local company and AT&T, does not receive the hangup signal
from
Asterisk. Therefore Asterisk thinks the channel is available but it
Kevin Savoy wrote:
I’m having an issue where Asterisk hangs up a call (either party hangs
up) but the telco side of the T1, both the local company and AT&T,
does not receive the hangup signal from Asterisk. Therefore Asterisk
thinks the channel is available but it’s still off-hook on the telc
Polycom 601. The screen on the 501 is too small. You can't even fix a full 7
digit number on the screen. What was Polycom thinking when they did that? They
also don't have the microbrowser.
> -Original Message-
> From: calvis [mailto:[EMAIL PROTECTED]
> Sent: Thursday, July 06, 2006 2:01
calvis wrote:
Grandstreams are junk. (I have had bad experiences with them)
The former doesn't necessarily derive from the latter :-)
Others of us have found them to be an excellent low-cost solution that
puts VoIP in places it otherwise would not be economical to deploy.
Everyone's mile
Polycom 501
Grandstreams are junk. (I have had bad experiences with them)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun
Sent: Thursday, July 06, 2006 12:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] for you guys setting up
I’m
having an issue where Asterisk hangs up a call (either party hangs up) but the
telco side of the T1, both the local company and AT&T, does not receive the
hangup signal from Asterisk. Therefore Asterisk thinks the channel is available
but it’s still off-hook on the telco side. I have co
On Thu, 2006-07-06 at 12:44 -0700, Shaun wrote:
> What brand/model phones are you using.
>
Aastra all models
--
Dave Cotton <[EMAIL PROTECTED]>
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Polycom 301/501/601, and SPA9xx
On 7/6/06, Shaun <[EMAIL PROTECTED]> wrote:
What brand/model phones are you using.
--
~Shaun
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What brand/model phones are you using.
--
~Shaun
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in the rtp.conf you specify range of ports, this range should be
forwarded on the firewall to the asterisk box.
I have asterisk running on 192.168.0.250 ip and connect to broadvoice
server. here is my iptables rule:
-A PREROUTING -i eth0 -s 147.135.0.128 -p udp -m udp --dport 10010:10013
-j DNAT
I have recently build 2 machines, one with an Intel Pentium Dual Core
CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
HDDs. Here are the show translations from both:
Intel Dual Core machine:
pbx*CLI> show
Colin,
Very good points, and you are right, I need to start tracking what has
been done.
A bit of history - this server was very unstable when running Digium
hardware - every day or two, it would kernel panic and lock up,
requiring a manual reboot. The other servers had issues as well, and
ALL
Can I please have some orientation as to how to plot in a graph the
data presented by fxotune -d parameter?
Thanks,
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http:/
Hi,
I'm getting a lot of these messages in my asterisk 1.2.9. The clients
are all Xlite Xten softphones. Isn't this message supposed to be part
of hardphones?
Got SUBSCRIBE for extensions without hint. Please add hint to 1001 in
context xlitephones
extensions.conf
exten => 1000,1,Macro(call-sip-
>Also, when I connect to the server locally (the server is in the room
>next to me, in other words, and i have 1 Gbit of bandwidth all the way
>to the back of the server, I still get call dropouts.
>However, this IS the only server (of 8 total, all in the same rack and
>connected to the telco vi
Hello List,
I work for VoiceIP Solutions in Seattle(Asterisk Provider) and I am willing
to help set the group up also. Please email me and we should get a core
group together and figure out how we want to handle it.
Thanks,
-Lauren M.
- Original Message -
From: "Josh Reineke" <[E
Thanks for the quick responses everyone.
To answer some of the questions posed:
The main traffic going over this pipe is voice, with a small amount of
web traffic as well. There are 60 total users, 5 of which access
anything other than what is on their LAN up there. In any case, we
are not sat
The mail system somewhere seems to have eaten some of the digetst versions of this list that are sent to me (jumped from 24 to 29). So - in case this didn't make it out, just expressing my interest. Were there many others around here who responded that I must have missed?philippePhilippe Lindheimer
Pele Zico wrote:
> Ive looked for cheaper mobile voip pstn calls in Nigeria and the cheapest
> ive found is about 18p/min or 28c/min. Im looking at providing for
> cpmanies here cheaper calls comparable to call cards. Can someone give me
> some ideas as to how i can do this. Can you negotiate pr
So you need a "divide and conquer" strategy here:
1. Is it Asterisk or the WAN? This should be easy enough to test for. Do
call dropouts happen in your datacentre? If not, your Asterisk install is
good. My money's on the 10mbit WAN pipe, and that's what I would be
focussing on.
2. If it's the WAN
Have completed a rough write up and I am having a linux/asterisk guru look it over. Once I submit it I would like some feedback on any stuff I left out or need to change.
Thanks
Julian
> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: RE: [asterisk-users] Got Mediat
Your problem is intermittent. It is probably Network related as if you
reboot that problems may or may not comeback.
In addition to the lspci stuff requested. Have you checked your
fiberlink. Is it possible that something or someone is saturating the
link with Virus/Spy/PtP Ware???
SIP doesn't ha
Here ya go:
lspci -vv
---
00:00.0 Host bridge: Intel Corporation E7520 Memory Controller Hub (rev 09)
Subsystem: Dell: Unknown device 016d
Control: I/O- Mem+ BusMaster+ SpecCycl
Server load is averaging around 20%, plenty of memory, disk space, and
bandwidth available. No QOS running on network. ulaw is the primary
codec. Server is stable, and there are no extraneous services
running, save mysql and httpd. Even running a processor intensive
query doesn't trigger the dro
The 8.2 firmware works just fine (asterisk 1.2.6). I have had exactly
zero problems with it, nothing even weird about it. Pretty trouble-free
IMO.
I believe the phone that doesn't work quite right with the 8.2 SIP image
is the 7970. I have probably 20 7940s/7960s, all running the 8.2 SIP
image wit
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Don
> Sent: Wednesday, July 05, 2006 11:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] intel vs amd motherboards
>
> >If you want t
> -Original Message-
> Thanks for reading,
>
> Wes
> ___
Please reply with the output of the following:
lspci -vv
lspci -vv | grep IRQ
lspci
cat /proc/interrupts
Thank you.
Andrew
___
--Bandwidth
On Jul 6, 2006, at 6:01 AM, Peder @ NetworkOblivion wrote:
Is there a "buddies" feature on the Cisco phones, like there is on
the Polycom? If not, how are people getting around the issue where
a receptionist wants to see who is on the phone? Or are they just
living with the limitation?
Hello everyone,
I've set up an asterisk box with basic PBX features (DiD, MoH, MoT,
Blind and Attended Call Transfer, PickUp, ecc.) for 10 Cisco 79xx (7912
and 7960) with SIP image (8.0). PSTN gateway is done using a Cisco
AS5350 with two ISDN PRIs connected to Asterisk via SIP. Between the
phones
Does anyone know if Asterisk Home (2.7) runs OK on a 64 bit architecture? Thanks.
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aster
Where may I find the 7.4 firmware for 7940? I was only able to find the 8.2
at cisco's website.
F
On 7/6/06 11:05 AM, "Aaron Daniel" <[EMAIL PROTECTED]> wrote:
> On Thu, 2006-07-06 at 10:55 -0500, Peder @ NetworkOblivion wrote:
>> What is the current recommended version of firmware for SIP on
>
Hate to drag this one back up, butit's happening again.
Overview of architecture:
Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel
1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the
sangoma a104d with onboard echo can.
Server is located in our data center
Yep, forgot ‘bout that.
Or you could use
web-meetme, it has this feature.
On 7/6/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:
Snip, snip.
Chop Chop.
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asterisk-users
Or you could use web-meetme, it has this feature.
On 7/6/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:
I would like to walk you through it but I have much on my plate right now that requires my attention.
I will point you in the right direction.
Look at the menu options in MeetMe, the ex
Steve Davies wrote:
On 7/6/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for a
Peder @ NetworkOblivion wrote:
What is the current recommended version of firmware for SIP on
7960/7940's. I was reading through some of the stuff on voip-info and
it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
Stay with the v7.x version until the newer stuff becom
Rizwan Hisham wrote:
Hi,
i need to set the dtmf mode on my quintum tenor a400 gateway.
You might want to check the a400 manual on how to do that.
i cant dial
any extension thru my normal digital phone which is connected to asterisk
thru the quintum gateway. it always falls to 's' extension.
On Thu, 2006-07-06 at 10:55 -0500, Peder @ NetworkOblivion wrote:
> What is the current recommended version of firmware for SIP on
> 7960/7940's. I was reading through some of the stuff on voip-info and
> it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
>
> PA
We stic
What is the current recommended version of firmware for SIP on
7960/7940's. I was reading through some of the stuff on voip-info and
it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
PA
___
--Bandwidth and Colocation provid
Doug,
Cheer up! There's some great beer brewed in Montana! Have a Moose Drool and get down to some "creative resume re-inventing."
On 7/6/06, Kevin Savoy <[EMAIL PROTECTED]> wrote:
I'm in Williston, North Dakota and we have an office in Billings, MT. He'sright. We are 500 miles form civilizat
On 7/6/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for about 5 seconds, then
This was written for use with AAH but should work for you as well.
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#722Polycom
On 7/6/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
Hi Friends,
I am using Polycom IP 501 hard phone. Now, I want to confiugre my Polycom IP
501
The 79X1 phones don't use the same configuration setups as the 79X0's.
They're the upgraded versions, using the SEP.cnf.xml files instead
of the SIP.cnf files.
On Thu, 2006-07-06 at 10:44 -0400, Doug Lytle wrote:
> Per Møller wrote:
> >>
> >
> > Hey Doug,
> >
> > Yes my 7940 and 7960 using the 7.4
Hi all !
i'm having a strange issue with an asterisk box behind a firewall
i'm trying to answer a sip call made to an asterisk box with a public ip
from another asterisk box behind a firewall
on the natted box i've put
externip=195.110.XXX.XXX
localnet=10.1.1.0/255.255.255.0
and on the phone co
Hi,
i need to set the dtmf mode on my quintum tenor a400 gateway. i cant
dial any extension thru my normal digital phone which is connected to
asterisk thru the quintum gateway. it always falls to 's' extension. So
plz help
___
--Bandwidth and Colocation
Per Møller wrote:
Hey Doug,
Yes my 7940 and 7960 using the 7.4 or 7.5 SIP firmware works fine and does
not use xml style config files.
I was looking for:
Asterisk side:
sip.conf
Cisco side:
SIPDefault.cnf
SIPMacaddress.cnf
Doug
-- Ben Franklin quote: "Those who would giv
On Thu, Jul 06, 2006 at 03:19:13PM +0200, Andrea Spadaccini wrote:
> Hello everyone,
> I'm trying to set up an Asterisk machine with a quad-port BRI
> Junghanns card, and I want to use the mISDN drivers.
That card is said to be exactly the same as the bero card. They both
look strikingly similar,
On Thu, Jul 06, 2006 at 09:52:57AM +0100, Marco Mouta wrote:
> Tzafrir Cohen,
>
> I'm not a linux expert, i just wanted to share what happened to me...
> Only after new detection of Sound Board, i got the audio from Asterisk
> services like MOH, Voicemail busy message ... working fine.
>
> Probab
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