Hi,
I hava a ZAP device (a premicell), and it sends polarityswitches when the call
starts and when the
call ends.
in zapata.conf with
answeronpolarityswitch=yes
then when the phone starts to ring, you dont hear it ring, only when the
person answers the phone
do you start to hear him
Does anybody know what these are? Started getting them last night when
I upgraded from 1.2.6 (Zaptel 1.2.6) to 1.2.10 (Zaptel 1.2.7). Then my
E1 ISDN PRIs go down...
I've had to roll back to 1.2.6 :-(
Matt.
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Hello Marcus,
Check your gatekeeper.ini
**
TotalBandwidth=
**
Don't Forget a gatekeeper use RAS to manage the
bandwidth .
try to add bandwidth !
Harry
PS:
Do you want to test your h323 terminal with my
gnugk/asterisk ?
--- Marcus Carlson [EMAIL PROTECTED]
Hi
Please how can I get the user ip address and put it at cdr
,its too important
Thanks
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express
Download only first cd iso, you will download what is needed later
it will be much less than 2 DVDs
Got bad internet connection :-( - only 192 kbit
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On Wednesday 19 July 2006 21:59, Cosmin Prund wrote:
Got bad internet connection :-( - only 192 kbit
Wouldn't say it's bad, just slow :).
Mark.
pgp23TQx5U4Zh.pgp
Description: PGP signature
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Cosmin Prund schrieb:
I'm using a HFC-S ISDN card and a TDM400P zaptel card with 3xFXO and
1xFXS modules.
Okay, getting a HFC-S card, a single s0 card, working with bristuff
should be no problem at all. be aware, if you want to use 4s0 or 8s0
Cards other then from junghans bith bristuffed.
Thanks for that.. finally tweaked where it might be... my conference
extension was the following
[conference]
exten = 4000,1,Answer
exten = 4000,2,Wait(1)
exten = 4000,3,DigitTimeout,5
exten = 4000,4,ResponseTimeout,10
exten = 4000,5,Authenticate()
exten =
Hi,
as I read there should be the functionality in 1.2.10 to set call-limits. As I
understood this option can be set in sip.conf per user just typing
call-limit=1
to restrict the calls to 1 incoming or 1 outgoing line for this user.
I'm using mysql for sip peers, so I added the attribute
On 18 Jul 2006, at 15:28, Colin Anderson wrote:
We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9
server.
It works perfectly fine, except at peak periods, say, 10 AM and 3
PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to
the next
priority in the
Hi,
Does asterisk will reference to asterisk database in every action
(register, invite, ...)? Can I use ARA to replace asterisk database
totally? How?
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Hi,
Is there a way to toggle Do Not
Disturb on polycom 501's and 601's automatically from Asterisk?
Thank in advance
Phil.___
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Hi all,
We have two server with *.
With 2-2 PRI card (sangoma , digium TE110P, digium TE100P).
server 1:
- asterisk: SVN-branch-1.2-r36998M
- span 1 Digium Wildcard TE110P T1/E1
- span 2 Digium Wildcard TE110P T1/E1
zap show status
Description Alarms
On 7/19/06, Dan Brummer [EMAIL PROTECTED] wrote:
Hello,
Well I was having transfer issues in 1.2.9.1 so I downgraded to 1.2.7.1.
For testing I installed 1.2.10 on a test server and setup two Polycom SIP
phones. Tried the transfer on this configuration and had the same issues.
Here is a log from
Keyboardot ragadtam, hogy va'laszoljak Colin Anderson osszedobalt bytejaira:
I think, if you should receive network busy, or unreachable (or at least
something, you should handle). You can also try your cellphone, if it gives
better result. Before moving your adapter, you also can try, to buy a
Hi,
How can I set all call forward to another number? I have tried to
change the exten but it doesn't change.
original exten is 7654321
exten = s,1,Set(EXTEN=1234567)
exten = s,2,NoOp(${EXTEN})
The result is 7654321.
Is it possible to change the value of ${EXTEN}? Or does it have any
better
http://www.cyber-cottage.co.uk/wiki/index.php/Call_forward
Is it possible to change the value of ${EXTEN}? Or does it have any
better way to implement to the all call forward feature?
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Hi,
Need to get the following working:
1. User calls ext 750.
2. If no answer or busy go elsewhere.
3. If answered and press 1 accept call.
4. If answered and not pressed 1 or timed out then
send call to be redirected to the busy or no answer
option.
The issue is that the call gets accepted if
Hi all,
I had some doubt about meetme application, hope that some
one can tell me what to do?
(No GUI and CLI in this
case)
Say 5 user in the conference room 1000. Ofcoz, each user is
holding userID in the conference room.
Say user 1 SIP/200
Say user 2 SIP/201
3 SIP/202
4
Hi,
I'm not sure if this is the right list for this question...
I have written a small application which looks up a number in a
database and return the name for the number if available.
But I have now the problem that I cannot read the variable CALLERIDNUM
from my script.
I tried it with:
Hey guys
The module reload is not really reloading user data's that I entered into extension.conf and sip.conf whileasterisk is still running.
Does this mean that I have to shutdown and restart asterisk everytimeI want to enter a user?
I've also tried extensions reload and sip reload, they too
Steven wrote:
If he modifies the local dialplan on the SIP device, the 3-way issue should
go away because he will no longer need to dial a #.
Pablo, Look for something like
(0T|011x.T|101x.T|x.#|9x.T|*x.T|#xx|393*1x.T|8|5xxx|x11) in the SIP
devices config.
You would want to change
On 7/20/06, TWV [EMAIL PROTECTED] wrote:
Hi,
We have a few PSTN Lines connected to our Asterisk server (via SPA-3000 FXO
interfaces), and everything works great except for 1 major annoyance.
When PSTN callers get to a user's voicemail and they just hangup, a few (4)
PSTN disconnect tones are
Tried the syslog debug, but it reports the exact same thing as the sip
debug shown below. It includes the INVITE, 100 TRYING, AND 486 BUSY
HERE. There are no hints as to why the Busy Here message is returned.
I was kind of guessing that something in the sip header was not as
expected for the
How did you managed to et info who pressed the button for feature request?
CHANNEL and CALLERID(NUM) variables are pointing to wrong direction.
Regards/Pagarbiai,
Mindaugas Kezys
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wayne P. HIll
Sent:
Hey guys
I'm on a task of creating a user\client software, for asterisk. My research led me to asterisk.org/doxygen,really highlights the functions to be used, but how to use them is what I'm interested in.
If there's anyone out there whoknows and can offer a hint, documentation, insight,
Actually it is a non-commercial solution he needs and you offered a
commercial one in a non commercial group. ;-)
Cheers
Gonzalo
there is a small chance of that, seeing that an address validation script is
used for just that address validation and is implemented in customer service
applications
On Thu, 2006-07-20 at 13:05 +0200, Matthias Fechner wrote:
But I have now the problem that I cannot read the variable CALLERIDNUM
from my script.
I tried it with:
value = pbx_builtin_getvar_helper(chan, key);
where chan is the value given from the initial function call (struct
Hi all,
I have a Voismart GSM card. I have calls through going fine. But in the cdrs, all the calls have
disposiotion of NO ANSWER and the billsecs are 0.
I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2
can anyone help?
--
thanks,
yusuf
--
This message has been
I didn't do anything except upgrade. We were using 1.2.7 because of
problems with transfer using Cisco 7960s over Sergio's chan_sccp
driver in later versions.After much frustration we dropped 1.2.10
on last night, and it just worked. Didn't make any changes at all to
the dialplan.
I would really want to know the interfaces between asterisk and softphone, how they plug in together.
The insight will be hight valued
Regards
Kevin
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Dear
I want to make a billing recharge through receiving digits from IVR
through dtmf which will
be inserted at mysql a2billing database after validation the digits entered ,
How can todo
that ?
Regards
*
No employee or agent
Hi List,
I need to put an Asterisk server in a remote office where
only ADSL is available. Maximum of 8meg downstream 646k upstream.
I need to handle 20 concurrent calls over IAX preferably
uLaw, so 64k per channel.
Is it possible to somehow have multiple NICs in the server
each
Hello Russell,
* Russell Bryant [EMAIL PROTECTED] [20-07-06 08:12]:
ast_verbose(The channel cid num is: %s\n, chan-cid.cid_num);
thx a lot!
Everything is working perfectly now. :)
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
build bigger and
Okay, I think i have missed something:
When i use AgentCallbackLogin*(||*007) the agent is logged in, fine.
But how do i log OUT.
okay there is a timout,
autologoff=time
but how can an agent explicit log off?
regards
Kai
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Chris Blunt wrote:
Hi List,
I need to put an Asterisk server in a remote office where only ADSL is
available. Maximum of 8meg downstream 646k upstream.
I need to handle 20 concurrent calls over IAX preferably uLaw, so 64k
per channel.
Is it possible to somehow have multiple NICs in the
Oops! :P nope, i didn't!
It was the problem :$
Thank you!!
Santiago
On 7/20/06, C F [EMAIL PROTECTED] wrote:
You using the t or T options in the dial app?
On 7/19/06, Delca [EMAIL PROTECTED] wrote:
Hi, Now that i fixed the problem with roundrobin, now i can't get
Blind Transfer to work. I
Chris,
Take in account that 64k on the RTP layer is about 108 kbit on DSL when sending
50 packets/s (20 ms samples)
ERik
Chris Blunt wrote:
Hi List,
I need to put an Asterisk server in a remote office where only ADSL is
available. Maximum of 8meg downstream 646k upstream.
I need
Rich,I had the same problem and the solution was to take out a 'malformed' callerid value from my sip.conf entry.TomOn 7/20/06, Rich Adamson
[EMAIL PROTECTED] wrote:Tried the syslog debug, but it reports the exact same thing as the sip
debug shown below. It includes the INVITE, 100 TRYING, AND
On Thursday 20 July 2006 15:10, Chris Blunt wrote:
I need to handle 20 concurrent calls over IAX preferably uLaw,
so 64k per channel.
Now that Thomas has handled your current bandwidth limitations,
I'll take a look at what you are trying to achieve...
Is it possible to somehow have multiple
It was a simple solution. Bruce Reeves was correct. Sangoma had and
issue with their drivers and re-syncing.
I upgraded the drivers and all is good.
Thanks to those that offered help. It was much appreciated.
Greg
I had similar problems with a Sangoma card in this configuration. I
recently
You need to create
another callbacklogin extension using # as the extension to log into. Logging
into extension # logs the agent id out.
Jordan Novak
Communications Technician
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Asterisk does not have softphone interfaces. You can write a softphone
to support some VoIP protocol supported by Asterisk, and voila, you
can connect to Asterisk. Supported and common protocols are IAX2, SIP
and H323. For IAX you have a library called iaxclient, so you are not
required to make
from the comments in mfcr2.c
/*
There also appear to be R2 variants for at least the following:
Australia
Belgium
Costa Rica
Eastern Europe
Ecuador (ITU)
Ecuador (IME)
Finland
Greece
Guatemala
Israel
New Zealand
Paraguay
Peru
South Africa
Uruguay
*/
So if the far end does not unblock then the telco has not finished
activating the line?
Right, :)
If the other end (the telco) has activated your line correctly, here
in Mexico you should be receiving in ABCD bits 1001, you can confirm
that by using zttool command and selecting the
I have a Voismart GSM card. I have calls through going fine. But in the
cdrs, all the calls have disposiotion of NO ANSWER and the billsecs are 0.
I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2
that's call received via vgsm interface
,+3620xxx6626,s,gsm417,
Hi All
I'm trying to get a 7970 working with SIP 8.0.3S and the latest build of
Asterisk (doing this as a new build before a replacement of my existing
system).
So far I've managed to get the phone upgraded successfully.
I can dial the phone from the console successfully.
However whenever I
We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9
server.
It works perfectly fine, except at peak periods, say, 10 AM and 3 PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to the
next
priority in the dialplan. Our interpretation of this is that the
I did a yum update queuemetrics and am now locked out of my box and my
unlimited license now shows as trial.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lenz
Sent: Wednesday, July 19, 2006 10:39 AM
To: Asterisk Users Mailing
Hi,
I'm wondering how the calls waiting announcement works when you have
several queues? We have different people answering different kind of
calls and we have three queues setup because of this...
If I where to use queue-callswaiting - how would it behave? Would it
only prompt the caller the
HiI have two phone numbers from my SIP provider sippro.com, say and . I use two sip.conf entries to register this phone numbers:register =
:[EMAIL PROTECTED]/register = :[EMAIL PROTECTED]/[]type=friendusername=secret=passinsecure=veryhost=
Title: Message
I'm
not exactly sure on the /how/ * mathes items from the sip.conf (I suspect it
goes to the latter for whichever provider), but the way configured my
extenions.conf to handle multiple incoming accounts from sipgate is like this
(obviously much simplified for ease of
It's a side effect of the implementation of agents and queues. Pausing is done
by the queue application itself. So if you don't use agents you still have the
pause ability. However the concept of logging in/out is done at the agent
channel level. So it is possible for an agent to be logged out
I fear the licence must be reinstalled after an update. My fault for not
adding it to the documentation.
For the locked out thing, there is no reason, unless you dropped and
rebuilt the database.
Please let us know in more detail so we can help you. Anybody else
experienced this?
l.
On
Hi.
I'm developing an application that dynamically adds agents to a queue.
First, the queue is created via RealTime, and then the agents are added
vía Manager, using QueueMemberAdd action.
However, for some reason, most of the time (NOT ALWAYS, and this is the
strange behavior), Asterisk
Hi All
I have an Asterisk box with 2 PRI's going into a Samsung DCS 500. After
a day or so of usage, the PRI channels are all used up and users cannot
make or receive calls over the PRI's. I get many of the following
messages in the logs:
Ring requested on channel 0/1 already in use on
I am trying to figure out a way to give agents the ability to do
attended transfers without having to touch their little black boxes
Plain cold transfers are easy with the manager Redirect action.
Can anyone think of a way to do this with the manager interface or local
channel somehow?
On 7/20/06, Mat Stace [EMAIL PROTECTED] wrote:
I'm not exactly sure on the /how/ * mathes items from the sip.conf (I
suspect it goes to the latter for whichever provider), but the way
configured my extenions.conf to handle multiple incoming accounts from
sipgate is like this (obviously much
Setting canreinvite=no on all the sip peers and gateway made the warm
transfer work. I'm still noticing ZOMBIE SIP entries, is this an
issue?
== Spawn extension (ANC, 1691, 2) exited non-zero on
'SIP/1691-09766938ZOMBIE'
Thank you,
Dan
-Original Message-
From: [EMAIL PROTECTED]
Hi List,
1. Has anyone thought of another way to show a user their login status? I am hesitant to start using a branch of code that has not been officially released yet. I am using 1.2.10 and the latest Polycom firmware.
2. How do I configure a soft button in the polycom phone for login and
On 7/20/06, Dan Brummer [EMAIL PROTECTED] wrote:
Setting canreinvite=no on all the sip peers and gateway made the warm
transfer work. I'm still noticing ZOMBIE SIP entries, is this an
issue?
== Spawn extension (ANC, 1691, 2) exited non-zero on
'SIP/1691-09766938ZOMBIE'
I think the ZOMBIE
Thank you Steve, I greatly appreciate the assistance.
-Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Davies
Sent: Thursday, July 20, 2006 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Warm
Not documented anywhere that I can see. What are the W:, C:, A:, SL: and
'within' fields showing?
Is holdtime AVERAGE hold time?
hestia*CLI show queues
oe_techsupp has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime),
W:0, C:52, A:11, SL:0.0% within 0s
Members:
Has anyone had any success creating a redundant ethernet connection from
their Asterisk server? What I would like it to do is use both ethernet
controllers on my motherboard so that if one fails the other one takes over.
I don't see anyway to make it work seamlessly with 2 IP addresses it would
Folks,Just as an update on this, DUNDI is working prefectly and regexten is working fine for peers defined in sip/iax.conf. However, for peers defined in Realtime the regexten does not appear to be created although the console reports that it is.
If anyone knows of any gotchas with regexten and
Yes, use the bonding driver. That way you only have one IP address and
both connections are viewed as one logical.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Thursday, July 20, 2006 12:45 PM
To:
You can try bonding the interfaces into one:
http://www.cyberciti.biz/nixcraft/vivek/blogger/2006/04/linux-bond-or-te
am-multiple-network.php
We have this in our Asterisk setup but recently got recommended to
remove it to combat intermittent issues.
-Dan
-Original Message-
From:
On 7/20/06, shadowym [EMAIL PROTECTED] wrote:
their Asterisk server? What I would like it to do is use both ethernet
controllers on my motherboard so that if one fails the other one takes over.
I don't see anyway to make it work seamlessly with 2 IP addresses it would
here are some url's to
Have a google for 'interface bonding'. You bond your two cards together to appear as a single one and then bind Asterisk to an IP address on it. The cards work in loadbalance or failover mode as you specify.
On 7/20/06, shadowym [EMAIL PROTECTED] wrote:
Has anyone had any success creating a
We're using OSPF...
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 20, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Redundant Ethernet
Yes, use the bonding driver. That way you
Guys,
I am trying to set up a SIP to PSTN gateway, and I have a client sending in
SIP traffic, once on the asterisk machine, I do something like this.
[outbound-sip]
exten = _XXX.,1,NoOP(${CDR(accountcode)})
exten = _XXX.,2,Dial(ZAP/g1/${CDR(accountcode)}${EXTEN})
This works, except that the
Unless my brain did melt down under the
high heat here, there are a few things I would share :
-
1 channel u-law uses at least 80k because of encapsulation
So,
think about compression
-
You can never use 100% of a bandwidth. (80% is a good max)
-
The values given are maximum values,
on 1.2.4 and 1.2.7, we have to set the 'type=peer' for call-limits to
work effectively.
type=friend doesn't seem to enforce call limits at all.
if you haven't tried type=peer, try that first.
good luck.
On 7/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
as I read there should be the
I'm a
little confused about Asterisk codec negotiation. Hopefully someone can
help.
I have
two phones, one on a slow link where I'd like to use G729, and one on a fast
link where I'd like to use ulaw.
My
sip.conf has:
[general]
allow=ulawallow=g729
...
[slow-phone]
allow=g729
allow=ulaw
On Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote: I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use G729, and one on a fast link where I'd like to use ulaw. My sip.conf has: [general]
I saw a similar question, but the solution didn't help me. I have my 9133i
setup behind a Linksys NAT/Firewall connected to an Asterisk server open to
the internet.
I can make all the calls I like, but the Asterisk server says I'm
unreachable.
I have contifure the NAT settings for the phone, and
Have you considered SDSL for higher bandwidth? Also, a dual wan router
would help with your load balancing problem? Just google'd and spotted one
that seems to be voip friendly. For incoming calls you could ask your ISP
to bind your ADSL/SDSL feeds.
Hypothetical offcourse :-)
Marty,
Ahhh I wasn't thinking about the fact that it would be keyed of the
callers settings, rather than the callee's.
However, setting the slow-link phone to g729 isn't a very workable
solution. We want to have ulaw as a backup, in case all of our g729 licenses are
in use. Having the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, July 20, 2006 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Redundant Ethernet
We're using OSPF...
Is That?
Oh
Shit
Hi All,
Using AAH 2.8.
I have configured a group to handle a common number for a remote office.
All phones in the office is in the group and they are ringing with a
seperate ringtone. All this is very well.
However all phones other than the one how answered the call is recording a
missed call.
Hi Harry,
Tried setting the bandwidth to different sizes (from 128 up to 1024000)
and both tried my normal account and the special SwyxGate account. Same
result.
I would very much appreciate if I could try against your H323 server.
Marcus
If you want realtime chat you could reach me at
W - Waiting
C - Completed
A - Abandoned
SL - Service level(defined in queues.conf servicelevel value). Percentage of
calls answered within the time frame.
These numbers reset on reload or restart.
--johann
Douglas Garstang wrote:
Not documented anywhere that I can see. What are the W:, C:,
On Jul 20, 2006, at 11:00 AM, Douglas Garstang wrote:Subject: Re: [asterisk-users] Codec NegotiationOn Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote:I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use
LOL
found this:
http://fcp.homelinux.org/modules/smartfaq/faq.php?faqid=549
hope it helps...
Guido
We're using OSPF...
Is That?
Oh
Shit
PBX
Failed?
SNIP
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Thanks Johann. Yes, I wish they wouldn't reset on a restart. :(
-Original Message-
From: Johann [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 20, 2006 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue Stats
W - Waiting
C -
Sorry
for the top posting. My email client is misbehaving.
Can't
use gsm. The polycom phones only support g711/ulaw and g729.
No, we
aren't intending to check for available g729 codecs that's why we wanted to
have ulaw as a backup when no g729 codecs where available.
-Original
Douglas Garstang
Not documented anywhere that I can see. What are the W:, C:, A:, SL: and
'within' fields showing?
Is holdtime AVERAGE hold time?
hestia*CLI show queues
oe_techsupp has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime),
W:0, C:52, A:11, SL:0.0% within 0s
Is it possible to make a 7960 speed dial automatically send DTMF digits
some specific number of seconds after dialing? I'd like to automate
dialing into a PBX.
-Dan
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asterisk-users mailing
Have you checked out pfSense? It's based on the BSD firewall m0n0wall, supports multiple WAN links and load balancing between them. Seems like it might do the trick for you. Philip Mullis will be giving a talk regarding how to get it all setup at the next Toronto Asterisk User's Group meeting, if
I used to build interactive voice response systems for a university. I
have someone who would like a small system built. Where would I find
information on * IVR capabilities and syntax?
Bob Rawlinson
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Warning: This message is a valid question, and is also kind of a [RANT]
at the end... but I'm high on caffeine and I had fun writing it.
The ranting part more or less reflect the state of the Asterisk
ecosystem until the end of 2005, which has been getting a little better
but a lot of the
Hello,
I have installed Asterisk, and configured the phone. I can hear the dial
tone,
But When I try to make a call I hear the message that
That number is not in speed dial, Please try again.
Can any please suggest what I am missing.
Below are my very basic configurations..
[EMAIL PROTECTED]
Hello,
I have installed Asterisk, and configured the phone. I can hear the dial
tone,
But When I try to make a call I hear the message that
That number is not in speed dial, Please try again.
Can any please suggest what I am missing.
Below are my very basic configurations..
[EMAIL PROTECTED]
On Jul 20, 2006, at 11:41 AM, Douglas Garstang wrote: Sorry for the top posting. My email client is misbehaving. Can't use gsm. The polycom phones only support g711/ulaw and g729. No, we aren't intending to check for available g729 codecs that's why we wanted to have ulaw as a backup when
snip
# If you live in Chiapas or northen Kazajkstan change it to .
Some guy used 0001 and his dog died.
I'll shut up now.
Funny stuff, thanks for the giggles.
I don't use E1, but see your points all to clearly.
Marty
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Hi Thomas,
That setting is controlled by line. Maybe you could setup two seperate lines
on the phones and direct the two different call types accordingly.
Franklin Webb
Assistant IT Project Leader
Inter Medi@ Marketing Solutions
610-701-9670
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- Original Message -
Hi, I'm using a Linksys PAP-2 to test my next PBX with asterisk.
The problem I'm haivng is that when I dial the extension, I've to end
it with # and then it starts calling is there any way to override that
# so with just dialing the 3-digit extension I'll be able to call?
This actually works
*snipped
Patrick, yes, this is a literal portion. I have no reason to believe that
spsaces between the priority, and the command cause problems, so I haven't
tried that yet. Just trying to make the horrible assembler-like Asterisk
dialplan language more readable.
*snipped
this doesn't
Hello all. I have a TE110P Digium board. Could I setup this board to
generate the clock to another equipment ?
Thanks
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Have you checked out pfSense? It's based on the BSD firewall m0n0wall,
supports multiple WAN links and load balancing between them.
Has anyone gotten this to work with Asterisk? I've got a couple of clients
who've tried it but they couldn't get it to work, and I've not been given the
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