On 1/12/07, Pierre du Plessis [EMAIL PROTECTED] wrote:
Thanks Eric, I'm using the asterisk DND
Is this really Asterisk, or is it Trixbox/FreePBX/[EMAIL PROTECTED]/etc?
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asterisk-users
Hello List
Just want to check if anybody else is having this problem.
Every time the PRI connections are disconnected, the card freezes, and I have
to reload the driver, to make it work again.
We are very seriously considering switching to Sangoma at this moment, due to
this and other
On Thu, Jan 18, 2007 at 11:17:12AM +0530, [EMAIL PROTECTED] wrote:
Hello friends,
I am trying to install asterisk 1.4.0 . I am configuring it as follows:-
./configure --prefix=/home/vivek/downloads/install/asterisk/
But still while running 'make install', it tries to install it in
CA == Colin Anderson [EMAIL PROTECTED] writes:
CA Sometimes it's asterisk, sometimes it's unknown sometimes,
CA it's Unknown so:
CA exten = asterisk,1,VoicemailMain(${CALLERIDNUM}) exten =
CA Unknown,1,VoicemailMain(${CALLERIDNUM}) exten =
CA unknown,1,VoicemailMain(${CALLERIDNUM})
CA This
In article [EMAIL PROTECTED],
Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello List
Just want to check if anybody else is having this problem.
Every time the PRI connections are disconnected, the card freezes, and I have
to reload the
driver, to make it work again.
We are very seriously
Hi,
I am using Asterisk 1.2.14 with zaptel 1.2.12 and libpri 1.2.4.
When we make a call to an unallocated number, the phone company will play a
recording saying it is unallocated and then send us the PRI cause 1. Is
there any way
to disable the recording and have the phone company send us
Return the card and ask for a new one. (i have seen this problem before
with a broken 411, a new card fixed it).
Zoa.
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello List
Just want to check if anybody else is having this problem.
Hi all,
I have very small issue for PAP2 registration issue. I hope some one already
faced this problem and solved.
I have more than 40 PAP2, first time it registered well and making call but
after some times like 5-10 mintes its not able to register on Asterisk till
customer
hello,
what you can do, is to activate NAT Keep Alive Msgs on LINE 1 in
admin/advanced mode,
Nat Settings:
Nat Mapping Enable: YES
Nat Keep Alive: YES
Nat Keep Alive Msg: $REGISTER
with this options, every 15 Seconds a Register Packet will be sent to
your Asterisk, an the Line should stay
Hello all,
Hoping someone can help me with an issue...I have i .call file which calls
out on a SIP channel and connects to an extension which dials another SIP
channel. (both via voip providers) - both to PSTN.
Problem is, hanging up the POTS phone doesn't release the channel (either
one -
Hi All,
Stupid and silly question - is there a way to limit the number of concurrent
calls an IAX client can make? something in the similar sense of incominglimit
and
outgoing limit on SIP?
Regards,
Nir S
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Hello, is there any way to reduce voicemailmain functionality without
recompiling .c file? Is there any external .conf file i can use to do it?
For example: i want to restrict password changing for users, etc.
Thank you for attention.
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Hello,
I am trying to install bristuff-0.3.0-PRE-1x.tar.gz on debian with kernel
2.6.19.2 and I've got some errors connected with XPP. I was wondering
if somebody managed to install bristuff with this kernel or any kind
of kernel 2.6.19. The bristuff mentioned above contains zaptel 1.2.10 not
Nir Simionovich wrote:
Stupid and silly question - is there a way to limit the number of concurrent
calls an IAX client can make? something in the similar sense of incominglimit
and
outgoing limit on SIP?
It can be done in the dial plan:
I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged
I can demonstrate this in practice by making a call from a handset, then
unplugging the handset from the power. The call remains active and
asterisk never seems to disconnect it.
More
Hi
i'm not very happy with TDM404B voice quality, low volume
Check the gain set in the zap config file. You can increase the in/out
gain quite a bit over standard.
Echo is frequently a symptom of wrong country settings, hence wrong
impedence settings. Also endpoints matter
Ed W
Hi
Echo cancel almost works, but the users
hear
what they describe as a 'crackle' coming back when they talk.
Just a thought, but check that your gain levels are not too high?
Ed
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Simon Tennant wrote:
I have noticed that Asterisk (version 1.2.13) is not hanging up a call
when the wifi handset moves out of range.
My setup is Nokia E61 connected to wifi access point (private IP range)
and then to server on internet (public IP).
I have been testing using the talking clock
Using a TDM400P in the UK nearly works fine, but I have a last remaining
problem in that if the incoming is ringing and then the caller hangs up,
asterisk takes another couple of rings before it spots the hangup.
This is annoying in that I sometimes get phantom calls late at night
(possibly
On Thu, Jan 18, 2007 at 11:46:01AM +0100, Andrew Nowrot wrote:
Hello,
I am trying to install bristuff-0.3.0-PRE-1x.tar.gz on debian with kernel
2.6.19.2 and I've got some errors connected with XPP. I was wondering
if somebody managed to install bristuff with this kernel or any kind
of
Looks like your trying to make both calls using the same channel. Try
using Zap/g1/number do dial since 'g' will take the first available
channel in the group 1(assuming your pri is setup as group=1 in
zapata.conf)
[]'s
MM
-Original Message-
From: Scott Keagy [EMAIL PROTECTED]
To:
Ed W wrote:
I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged
Have you tried the RTP timeout settings in sip.conf?
;--- RTP timers
; These timers are currently
On Thu, Jan 18, 2007 at 01:56:22PM +0200, Tzafrir Cohen wrote:
On Thu, Jan 18, 2007 at 11:46:01AM +0100, Andrew Nowrot wrote:
Hello,
I am trying to install bristuff-0.3.0-PRE-1x.tar.gz on debian with kernel
2.6.19.2 and I've got some errors connected with XPP. I was wondering
if
It is called exec.
http://www.google.com/search?hl=enq=asterisk+agi+exec
On 1/18/07, Thomas Hecker [EMAIL PROTECTED] wrote:
Hi everyone,
Is it possible to call an asterisk function out an AGI script? How do I do
this?
Thank you,
Thomas
___
On Wednesday 17 January 2007 4:14 pm, Gustavo Andrés Salazar Giraldo wrote:
Yes, i am using E1/PRI
Ok. What PRI switchtype is the Meridian set up for? You need to set the same
(or a similar-enough one) in zapata.conf, and specify the signaling
as pri_net so that Asterisk acts as if it were
Interesting, well if you're seeing the other selects in the mysql.log
then this update not showing up is bizarre. It would also mean that
permissions are irrelevant if doesn't even attempt to change the
password, as you'd rightly pointed out as well. I just tested it again
and this is what I
Thomas Hecker wrote:
Hi everyone,
Is it possible to call an asterisk function out an AGI script? How do I
do this?
Thank you,
Thomas
Yes, we have done this a few times, using PHP. You define an extension in the dialplan, from which
you call your AGI, then in in you have access to all
Note: you can also download and fine tune the php used to do it.
works like a charm.
On 1/17/07, Colin Anderson [EMAIL PROTECTED] wrote:
On the voip-info.org wiki there are good tips to get snoms to play nice on
lans. I personally have experienced wierdness using particular switches
(cheap
What is the best way to have 1 phone check multiple voicemail accounts. I am
using polycom 650 phones, and am wondering if mwi can work when checking
multiple accounts.
-Chris
Sent from my BlackBerry® wireless handheld ___
--Bandwidth and Colocation
Hey List -
I've been looking into the various options for small form factor customer
premise gear, and am wondering what your using and what your reccomendations
are.
I'd like to drop a unit at the customer premise to handle their internal
routing and trunk their outgoing calls back into my
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 16.01.2007, 12:01 -0800 schrieb Alejandro Duplat:
Hi,
Someone knows an Open Source solution that can handle Outbound IVR for
asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach
a Person and start making an
[EMAIL PROTECTED] wrote:
What is the best way to have 1 phone check multiple voicemail accounts. I am
using polycom 650 phones, and am wondering if mwi can work when checking
multiple accounts.
Try
[EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple
mailboxes
in
Hi Chris,
We have a customer who we set this up for on Polycom 501's.
We set the first two lines buttons to be their own extension, and the last
one to be the general delivery mailbox. If either account has a message, the
MWI lights up. For transparency, you can have all the buttons say the
Nice! Both solutions works fine. I also have found third:
I have replaced the bristuff.../zaptel-1.2.10/xpp/xdefs.h with xdefs.h from
zaptel-1.2.12. It also works, but I am not sure if this won't damage
something else later.
Cheers
On 1/18/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu,
This might sound like an odd question but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does in fact see the callers ID. But...
our old phone system set
I guess I'm missing something else. 'ringinuse = no' doesn't change
anything. While on a call, the queue still sends another call and
proceeds to set the member paused after receiving 'Busy Here' back from
the SIP device.
My queues.conf is:
[general]
persistentmembers = no
On Wed, Jan 17, 2007 at 10:34:55AM -0600, JR Richardson wrote:
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally,
99% of the time that people have this problem, it's because they renamed
the column uniqueid in the vmdb.sql sample file, not realizing that
the name of the column MUST be uniqueid for password changing to
work.
--
Tilghman
Holly column name Batman! That worked. In my effort to be consistent
SOLVED.
I found that simply adding
senddtmf=yes
to my misdn.conf
solved the problem
so I can use only RFC2833 inside and all inside ivr, mailbox, route
password are OK
and I can access the remote ivr
senddtmf default to no, so specifing it to no or deleting that line from
misdn.conf rollback
Hi, this is a signalling question:
I have a 4port fxs-to-sip where i connect standard analog phones. I
want to connect this device to an avaya PBX and then the device talks
to asterisk via SIP.
What signalling do i need the avaya to provide? FXO signalling right, like this?
Hi all,
using latest asterisk-svn
I want to reflect an video call incoming via an PRI EuroISDN channel to
another outgoing PRI channel,
and I want the the outgoing channel to have the exact same bearer
capability
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability:
Hi,
Anyone has access to the LMC 10.0 software needed to configure the
Madge AccessSwitch 20 ? We bought one from ebay last year and now I can
not find the CD with the software...
Thanks,
Andre
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Hello everyone.
I need a BRI ISDN card that works in Romania. I already have one of the
Cologne HFC-S PCI cards and it doesn't work right, it's junk. I get
wy too much echo using it. I'm now shopping for a better card. Can
anyone recommend me a card that fits the following:
(a) Costs
Did u try this SetTransferCapability ?
Hi Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: giovedì 18 gennaio 2007 17.47
A: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Oggetto: [asterisk-users] Passing video
I finally have the solution, so thought I would post back to the list for
completeness.
It ended up being a series of changes. First, on the gateway, set Disconnect
on Broken Connection to false. Then, for the Polycom phones, set
voIpProt.SIP.allowTransferOnProceeding to 1 in the sip.cfg.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Fromm
Sent: Thursday, January 18, 2007 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
I guess I'm missing
Connect to the avaya line ports, not station ports.
On Jan 18, 2007, at 10:46 AM, Erick Perez wrote:
Hi, this is a signalling question:
I have a 4port fxs-to-sip where i connect standard analog phones. I
want to connect this device to an avaya PBX and then the device talks
to asterisk via
On Thu, 2007-01-18 at 19:00 +0200, Cosmin Prund wrote:
Hello everyone.
I need a BRI ISDN card that works in Romania. I already have one of the
Cologne HFC-S PCI cards and it doesn't work right, it's junk. I get
wy too much echo using it. I'm now shopping for a better card. Can
anyone
A follow up (late better than never)
Finally had time to sit down and look at this
sip.cfg
keys key.scrolling.timeout=1
key.IP_500.31.function.prim=BuddyStatus/
This turns the Services key which I never use on a 501 into the Buddy
Status. It even works while on a call. One
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On 18 Jan 2007, at 18:31, Patrick wrote:
I think http://www.melware.de carries the Eicon Server ISDN cards
which
have hardware echo cancellation. They are also the author of the
chan_capi driver for Asterisk. I use the Eicon Server BRI cards with
At 11:56 1/18/2007, Bill Gibbs wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary=_=_NextPart_001_01C73B2A.03C9AD84
A follow up (late better than never)
Finally had time to sit down and look at this
sip.cfg
keys
I do not care about fax. I just want a good VOICE card.
Can someone please give a price quote for this card, give or take 10%? I
just spent 5 minutes filling in a really long form on a shopping web
site to get a price quote, only to find my account needs to be manually
activated before I can
How about the Digium Wildcard B410P card? It seems to be Digium, it
has hardware echo cancel and I can buy this in Romania. Is this card any
good?
Cosmin Prund wrote:
Hello everyone.
I need a BRI ISDN card that works in Romania. I already have one of
the Cologne HFC-S PCI cards and it
Hi List,
I cant seem to find the setting that changes this! You put a person on hold,
they are on hold like normal, but after a few seconds the phone will then
start having dialtone coming from the speakerphone, really weird!! Anyone
know how to fix this? I see where it could be nice, but in
Jens Vagelpohl ha scritto:
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On 18 Jan 2007, at 18:31, Patrick wrote:
I think http://www.melware.de carries the Eicon Server ISDN cards which
have hardware echo cancellation. They are also the author of the
chan_capi driver for Asterisk. I use the
We are using Sirrix (http://www.sirrix.com).
4* BRI for app. € 550.
Works fine; has echo cancellation as well.
Cosmin Prund wrote:
Hello everyone.
I need a BRI ISDN card that works in Romania. I already have one of
the Cologne HFC-S PCI cards and it doesn't work right, it's junk. I
get
Cosmin Prund ha scritto:
I do not care about fax. I just want a good VOICE card.
Can someone please give a price quote for this card, give or take 10%?
I just spent 5 minutes filling in a really long form on a shopping web
site to get a price quote, only to find my account needs to be
On 1/18/07, Cosmin Prund [EMAIL PROTECTED] wrote:
How about the Digium Wildcard B410P card? It seems to be Digium, it
has hardware echo cancel and I can buy this in Romania. Is this card any
good?
...well, it's essentially a 4 port HFC based card with builtin
HW echo canceler.
I'd say if
On Thu, 2007-01-18 at 20:13 +0200, Cosmin Prund wrote:
How about the Digium Wildcard B410P card? It seems to be Digium, it
has hardware echo cancel and I can buy this in Romania. Is this card any
good?
Well I hope so, I'm installing my first one next week.
--
Dave Cotton [EMAIL
No, call-limit is not being used. Do you have ringinuse=no working?
Has anyone seen it work?
Each SIP device has a very minimal config in sip.conf. Here's a show
sip peer:
* Name : 3207
Secret : Set
MD5Secret: Not set
Context : outbound
Subscr.Cont. : Not set
Sinisa Djokic wrote:
hi..
does anybody knows can we use cisco voice router ( for example
2811-v/k9 ) with some free gatekeeper – for example GNU..
so, the point is as follows..
we register cisco ATA186 with h323 firmware on GNU gatekeeper..
but can we establish connection from some
I finally found a price tag for the darn thing, at around 500 euros I
can handle it.
Qustion: Do they behave properly if I've got an other Digium TDM400 card
in the system? How about installing two cards in the same server?
At the moment I've only got 1 ISDN line plus a few analog lines going
Ron McCarthy schrieb:
I cant seem to find the setting that changes this! You put a person on
hold,
they are on hold like normal, but after a few seconds the phone will then
start having dialtone coming from the speakerphone, really weird!! Anyone
know how to fix this? I see where it could be
On Thu, 18 Jan 2007, Alberto Pastore wrote:
I tested BRI-2M, 4BRI-8M, PRI-30M on several installations,
even older 1.0 version cards (PCI 5v only) just work great.
I use diva server drivers software source rpm from Eicon,
chan_capi from www.melware.org (0.7.1) on asterisk 1.2.14
(kernel
On Jan 17, 2007, at 11:00 AM, [EMAIL PROTECTED]
wrote:
Date: Wed, 17 Jan 2007 09:55:50 -0700
From: David Thomas [EMAIL PROTECTED]
Subject: Re: [asterisk-users] FW: Realtime Voicemail Password Change
Not Working
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks for your help, but I've already adjusted timers on the source
code. I found your document a week ago and read it.
Do you really think that is a matter of timers only?
Greets!
On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote:
Sometimes timers need to be adjusted on the mfcr2 source code.
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On 18 Jan 2007, at 19:49, Cosmin Prund wrote:
I finally found a price tag for the darn thing, at around 500 euros
I can handle it.
Qustion: Do they behave properly if I've got an other Digium TDM400
card in the system? How about installing two
On Thu, 18 Jan 2007, Cosmin Prund wrote:
I finally found a price tag for the darn thing, at around 500 euros I can
handle it.
Qustion: Do they behave properly if I've got an other Digium TDM400 card in
the system? How about installing two cards in the same server?
At the moment I've only got
Hi Ron,
You can change this setting through the web interface Advanced/Audio/Dialtone
during Hold.
Hope that helps!
Regards,
Usman.
-
Usman Tahir
snom technology AG
Gradestraße 46
www.snom.com
This e-mail may contain
I have over 12 sip trunks that are very similiar, like extension 412 below.
Is there any way to have a single variable in [general] that contains
the 7 lines after 'username=412'? This would reduce clutter in sip.conf.
I don't want to put these common lines in [general].
What I have now
Check without the echocan module (remove it) if any 'crackle is listen
again.
If yes, the echocan is not faulty.
If yes, check another echocan module temporary.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ed
Hello All,
I have basic Asterisk 1.2 and Zaptel 1.2 running with Digium TE405P
T1/E1 card.
I downloaded the chan_ss7 (ver 0.9 and 0.8.4) from
http://www.sifira.dk/chan-ss7/
But I am failing to install the chan_ss7 drivers... Has anyone installed
chan_ss7 drivers successfully?
Regards,
Hi all,
I have configured the queue below, but when I go into the queue,
asterisk does not announce hold time:
[support]
musiconhold=default
strategy=ringall
context=check_time
timeout=20
wrapuptime=1
maxlen=3
announce-frequency=5
announce-holdtime=yes
joinempty=no
leavewhenempty=yes
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I think what you're looking for are configuration templates:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+template
Then you would have something like:
[grandstream]
context=default
type=friend
qualify=yes
insecure=very
host=dynamic
What does unmonitored mean in the below reference?
Ref:
CLI meetme list
User #: 01 5665 zzz Channel: SIP/5665-9f8038a0
(unmonitored)
User #: 02 5664 no nameChannel: SIP/5664-0096b660
(unmonitored)
2 users in that conference.
Also, is
Philipp Kempgen wrote:
Ed W wrote:
I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged
Have you tried the RTP timeout settings in sip.conf?
Sounds exactly like what I need! Thanks
Is there no default set then??
Cheers
Ed W
Ed W schrieb:
Philipp Kempgen wrote:
Ed W wrote:
I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged
Have you tried the RTP timeout settings in sip.conf?
Is there no default set then??
Not sure about that. I guess these timeouts
Hello all,
I have an Asterisk PBX with the Queue Log Analyzer installed
[http://www.micpc.com/qloganalyzer].
On the main menu, there's an option of CALLS COMPLETED [ALL] where
I can see the completed calls that entered any of the queues and my
question is:
There's a column that states
On 15:48, Thu 18 Jan 07, Lee Jenkins wrote:
Hi all,
I have configured the queue below, but when I go into the queue,
asterisk does not announce hold time:
[support]
musiconhold=default
strategy=ringall
context=check_time
timeout=20
wrapuptime=1
maxlen=3
announce-frequency=5
Bernardo Vieira wrote:
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I think what you're looking for are configuration templates:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+template
Then you would have something like:
[grandstream]
context=default
type=friend
qualify=yes
On Wed, 17 Jan 2007, Wireless wrote:
should have sent this to the list, Gordon how are you getting on with BT?
Badly.
However out of 2 separate systems for that particular client with 9
incominf BT lines over 3 boxes, only 1 line has this really bad problem,
and we've swapped out boards,
Michiel van Baak wrote:
What's the line in extensions.conf to go into the queue ?
I found out that if you use the r flag there (provide
ringtone) the announcements wont work.
Odd. I *did* originally have it set to use MOH:
exten=999,1,Queue(support,t|||60)
but it didn't work (for whatever
Well now that I look into it, if you disable the call waiting the
response is 486 busy here. If you use the DND it is the same response,
so there's no way to do phone-side DND and correctly report the
voicemail state.
But the Aastra phones do support the selection, which IMO should be
603
What you have been told is correct, seems like there is something strange
in your setup then, with * not logging correctly. I'd try running a couple
more calls to see if the problem persists.
l.
On Thu, 18 Jan 2007 23:03:28 +0100, Danny Lan M. - Telegroup®
[EMAIL PROTECTED] wrote:
Is it not coming in as CallerID(RDIS)? The specifications for the
service don't seem too different from any other PRI.
On 1/16/07, Gary Mensenares [EMAIL PROTECTED] wrote:
Hi everyone!
I'm wondering if anyone on the list had the opportunity to work with an NTT
INS1500 ISDN PRI service before.
Most of the Cisco phones sold cheap are UNLICENSED (global spare)
thus you would not be able to purchase (or at least aren't supposed
to) the smartnet contracts, you need to buy the license ($100+) and
the contract ($10 or so)
On 1/10/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
David Thomas
Yehavi Bourvine +972-8-9489444 wrote:
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
That's a very nice little script.
--
Warm Regards,
Does there exist any workaround? In the iax.conf file I need to configure a
peer with a FQDN instead of the IP because the IP of this domain changes once
in a while.
Kevin P. Fleming [EMAIL PROTECTED] wrote: housi mueller wrote:
The dnsmgr in Aterisk 1.4.0 seems not to work. I enabled DNS
- Original Message -
From: Benny Amorsen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 17, 2007 12:10 AM
Subject: [asterisk-users] Re: How to detect long calls
KS == Savoy, Kevin - Williston, ND [EMAIL PROTECTED] writes:
KS We have been running an
Thanks Jerry. Are the avaya station ports a special type ?
On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote:
Connect to the avaya line ports, not station ports.
On Jan 18, 2007, at 10:46 AM, Erick Perez wrote:
Hi, this is a signalling question:
I have a 4port fxs-to-sip where i connect
Hi folks,
Moving on to a new install, I'm jumping straight to v1.4
Without using Priority jumping I'm wondering what the 'standard' way
to indicate to the calling party that the number the dialed is busy or
unavailable. So,if I have an entry in extensions.conf like this:
[outbound]
exten =
Russell Horn wrote:
Hi folks,
Moving on to a new install, I'm jumping straight to v1.4
Without using Priority jumping I'm wondering what the 'standard' way
to indicate to the calling party that the number the dialed is busy or
unavailable. So,if I have an entry in extensions.conf like this:
On 1/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Looks at macro-stdexten in extensions.conf.sample. Also see show
application dial
Ah, that's exactly what I was looking for - thanks.
Russell
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I know that NAT is something no one really likes to talk about, but does
anyone know how work with it elegantly? There are many providers which deal
with it on a daily basis in fact they cater to it, is this possible to do
with asterisk or does it require other exotic setups? I even know of a
Andrew Joakimsen wrote:
Most of the Cisco phones sold cheap are UNLICENSED (global spare)
thus you would not be able to purchase (or at least aren't supposed
to) the smartnet contracts, you need to buy the license ($100+) and
the contract ($10 or so)
I'm always surprised by by the number of
Voip Asterisk wrote:
I know that NAT is something no one really likes to talk about, but
does anyone know how work with it elegantly? There are many providers
which deal with it on a daily basis in fact they cater to it, is this
possible to do with asterisk or does it require other exotic
What about open sip stack:
http://www.opensipstack.org/
?
Use a far end nat traversal appliance. Acmepacket , kagoor and Jasomi
are some examples.
Leo
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asterisk-users mailing list
To
Howdy,
How do you populate the queuestats from asteriskguru.com
outgoing stats? Does it take an outbound dialer like
Vicidialer?
For good manager reports, should the user be tied to a
queue and have all calls count as work?
Inbound stats works well, but outgoing stats do not work
and I'm not
Is is possible to detect open SIP channels in the dial plan and use that
detection to perform logic?
For instance, say you have a multi line device such as a poly 301, depending
on the path of the incoming call you want to be able to route the call to
the phone if 1 line is already in use or
You can use the group function to determine this
On 1/19/07, Voip Asterisk [EMAIL PROTECTED] wrote:
Is is possible to detect open SIP channels in the dial plan and use that
detection to perform logic?
For instance, say you have a multi line device such as a poly 301, depending
on the path of
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