Had a similar issue with the same model a while back - happened out of the
blue.
Don't ask how we found this out - however we discovered that if the person
at the 7960 end screams into the phone, they can then hear the other end for
a few seconds (having it dialled in to MOH helps when testing
Hi,
Thanks for the answer.
Yes, I think both channels are built.
I see following messages at startup:
Parsing '/etc/asterisk/h323.conf': Found
Creating H.323 Endpoint
Parsing '/etc/asterisk/users.conf': Found
Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver)
H.323
Hi,
For timestamp a recording you can use this...
exten =
1XXX,1,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},GMT-5.5,%d%b%Y)}-${STRFTIME(${EPOCH},GMT-5.5,%H%M%S)})
exten =
1XXX,n,Monitor(wav,/home/recording${STRFTIME(${EPOCH},Asia/Calcutta,%Y%m%d)}/${CALLFILENAME},m)
Hi,
For recording your each phone call use this in your all dial-plan in
extension.conf
By these lines there will a time stamping in your all call, and call will be
saved in date directory. Choose your GMT and Time , accordingly.
exten =
Transfert authorization.
Le 14 juil. 07 à 22:20, bilal ghayyad a écrit :
Hi List;
[incoming]
include = parkedcalls
exten=103,1,Dial(SIP/Bob,,tT)
exten=104,1,Dial(SIP/Charlie,,tT)
When we use tT and when we use t alone or T alone, I
know this for call parking, but I do not know what the
tT
What output do you get from the CLI ?
- Original Message -
From: yonoko molomo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 17, 2007 9:59 AM
Subject: Re: [asterisk-users] asterisk 1.4 and gnugk with
Here is my piece of generosity of the work - and it's not even my work.
7xx to login, *7xx to logout.
Thanks for the code, it won't be usefull though (As I suspected it don't
perform what I wanted to do).
This piece of code may be usefull for someone else :-).
Am Montag, den 16.07.2007, 09:44 -0500 schrieb Jeremy Mann:
Does anyone know if X-Ten or SJPhone support multiple cordless
handsets for multiple lines? I have an office with multiple roaming
users(nurses) that are in and out. I’d like to provide them
telephones, and my idea is to have a PC
Hi,
I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
hi,
i fixed the problem.
as i thought it was a configuration problem, i was not defining the
asterisk users at ooh323.conf.
now it seems to work,
thanks
2007/7/17, Dovid B [EMAIL PROTECTED]:
What output do you get from the CLI ?
- Original Message -
From: yonoko molomo [EMAIL
Hi,
I am using asterisk 1.4.
I have confgured the musiconhold.conf file.
However, when i make a call and then hold the call it does nothing.
in the CLI i do not see the starting/stopping musiconhold messages.
i am making calls from sip to h323 using asterisk assip/h323 gateway
(with gnugk and
Hi,
Apologies if this has been asked before, but I don't seem to be able
to find any info on it anywhere.
Sometimes when placing a call on hold, the caller hears digitized/
robotic music on hold that gradually improves over the course of
about 20 - 30 seconds until it sounds pretty normal.
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as it could be (?). I'm wondering if there's active work on
that right now or if it's more of an issue about PSTN carrier
On Tue, Jul 17, 2007 at 11:56:35AM +0200, Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as it could be (?). I'm wondering if there's active
Hi to all,
till now I've used SER as sip registrar and Asterisk as its gateway(PSTN)
and for billing.
Now, I've received a request to setup a solution, for 5000 + o - users(this
is what they expext to have).
I was thinking to use only Asterisk with Freeradius, no SER.
Any suggestion/experience?
Dear Mojo;
Thanks a lot, yes I understood what you mean. You mean
that the tT was putted in the Dial as a feature or
setting to give them the chance to use the call
parking that was included in the first line.
But only when dialing bob or charlie. Only the second
line, the
'include' line, is
Hello,
I have two incoming lines connected to my Asterisk
([EMAIL PROTECTED]). One voip and one ISDN line. Both go into
incoming context.
I have a problem, that when I press Flash the client who calls does not
hear music on hold, but only on the ISDN line, on VOIP everything is ok.
[incoming]
hi there,
i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x
network, behind my main
computer, i cam make call, receive calls, all works fine, with all providers
except sipgate.de,
there i can receive call and make them, i can hear the other end but they can
not hear me,
hi there ppl :D
i'm a happy and very gratefull asterisk user, newest version, on a debian
machine (stable) i also
have tried asterisknow and freepbx etc, but i want to have a website, which
offer the following:
# Integrated Web Dialer (Click-to-Dial)
# Workgroup Answering Machine
# Monitor
Me again, another problem.
As I said before, I have 2 lines going into incoming context.
When client calls, I press Flash, client hears music on hold (only on
voip line as said in previous post), when I get back and press Flash
again to get back to my client I cannon hear him, but he hears me
I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H
option in Dial() app. When press * during the call from caller side, Asterisk
does not disconnect the call. The * just pass through. Here is my test dial
plan:
exten = 8111001001,1,Answer()
exten =
On 7/16/07, The Asterisk Development Team [EMAIL PROTECTED] wrote:
fix various known issues. See the ChangeLog included in the releases
for a full list of changes. The ChangeLogs are also available
separately on the ftp site.
Is there any more information available on this change?
hi list,
I'm currently trying to get Asterisk running with an HFC-compatible ISDN
card (no-name product, but supposed to work with Asterisk according to the
packaging). the ISDN-card is connected to a alcatel ISDN-system where it
should act just like a normal ISDN-phone.
I went with
hi there :)
i want to have a website, which offer the
following:
# Integrated Web Dialer (Click-to-Dial)
Easy to make your own. The only question is integrate into what?
i don't know, maybe easy for you but not for me ;) just a webinterface in php
or twisted.web
maybe?
#
I also tried blind transfer with t option and it did not work. I added
following into my dial plan contest:
include = featuremap
exten = 8111001001,1,Answer()
exten = 8111001001,n,Wait(2)
exten = 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|tHL(12:61000:3))
exten = 8111001001,n,Hangup()
Anyone using any variation of Multiparking, Parking Valet or servicing Call
Parking with Multiple Tennants?
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
James FitzGibbon wrote:
Is there any more information available on this change?
2007-07-13 08:22 + [r2733-2736] Tzafrir Cohen
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
* Fix a digit mapping bug with hardware dtmf detection (r4357)
I assume those revision numbers (4xxx)
Hi,
Lately we've noticed that some Zap channels on one of our PRIs are
unavailable. We have 2 PRI lines with 60 channels in total. On the first
PRI there are currently 20 channels that are not being used for some
reason.
I tried googling around and found some similar problems but there really
I should have included using a multi parking feature with asterisk 1.4?
-Original Message-
From: Kevin Kiely [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 17, 2007 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Multiple Parking Lots
Hello Jan,
We have also been seeing this issue, and we are running Asterisk
1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI
provider that a 3rd party vendor has applied firmware to some hardware
along our path, and that it has an unfortunate bug of hanging B-channels
in the
Hi -
I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both
ports.
I need to be able to call one port from the other-- the idea is to have
two phones in two different locations that _can_ call each other.
So, in reading the Asterisk Wiki and other sites, the best
Dear all
I am going to install 2 port pri card on asterisk but i dont
know how to incomming call goes in to IVR and how to route call outside base on
pattern match means if some one call on mobile phone then use PRI 1 and if call
on landline phon call route through pri 2
Have you tried setting resetinterval=never in zapata.conf?
On Tue, 2007-07-17 at 15:43 +0200, [EMAIL PROTECTED] wrote:
Hi,
Lately we've noticed that some Zap channels on one of our PRIs are
unavailable. We have 2 PRI lines with 60 channels in total. On the first
PRI there are currently 20
Jody Gugelhupf wrote:
hi there :)
i want to have a website, which offer the
following:
# Integrated Web Dialer (Click-to-Dial)
Easy to make your own. The only question is integrate into what?
i don't know, maybe easy for you but not for me ;) just a webinterface in php
Hi
On Tue, Jul 17, 2007 at 08:58:26AM -0400, James FitzGibbon wrote:
On 7/16/07, The Asterisk Development Team [EMAIL PROTECTED] wrote:
fix various known issues. See the ChangeLog included in the releases
for a full list of changes. The ChangeLogs are also available
separately on the ftp
So what you actually want a web based phone?
Jody Gugelhupf wrote:
hi there :)
i want to have a website, which offer the
following:
# Integrated Web Dialer (Click-to-Dial)
Easy to make your own. The only question is "integrate into what?"
i
Anthony Francis wrote:
Doug Lytle wrote:
Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.
You have two servers in your MX records.
drdos.info. 60 IN MX 10 smtp.drdos.info.
Hi,
No I havn't tried that. That entry wasn't even in there so I'll try it.
I'll let you know if it helped.
The odd thing is that this problem started yesterday. And our asterisk
has been running for +1 year without these kind of problems.
So either our telco has changed something OR it's
Okay, I've got an update on the resetinterval=never... same thing even though i
added the line to zapata.conf and restarted the server.
Now the load wasn't even high, maybe 6-7 calls. I think I just might call my
telco, feels like it's their issue, but if anyone has any other suggestions let
Philipp Kempgen wrote:
Anthony Francis wrote:
Doug Lytle wrote:
Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.
You have two servers in your MX records.
drdos.info.
Hi Satish -
I am going to install 2 port pri card on asterisk but i dont
know how to incomming call goes in to IVR and how to route call outside base
on pattern match means if some one call on mobile phone then use PRI 1 and
if call on landline phon call route through pri 2
Hello,
I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
in my ata the the GW return a media not acceptable error.
but If i add the g729 codec the all is ok.
I see in the config of the cisco where to define codec for imcoming call but
not for outgoing
*Jul 17
Laurent,
You should be able to set it with the 'codec' subcommand on the outgoing
dial peer as well. 'codec g711ulaw' or similar.
-- Alex
On Tue, 17 Jul 2007, laurent schweizer wrote:
Hello,
I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
in my ata the the
On 7/17/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I am going to install 2 port pri card on asterisk but i
dont know how to incomming call goes in to IVR and how to route call outside
base on pattern match means if some one call on mobile phone then use PRI 1
and if
I did a quick test. What happens is Congestion() answers the channel and
leaves it open. IE do a 'show channels' and you will see the channel is
still open on your end.
Sorry I don't have further suggestions.
On 7/17/07, Arun Kumar [EMAIL PROTECTED] wrote:
Hi,
I've an PRI coming to my
On 7/16/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?
My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could
I have already setup a list of prefered codec , but it's only for incoming
call, not outgoing
Laurent
2007/7/17, Alex Balashov [EMAIL PROTECTED]:
Laurent,
You should be able to set it with the 'codec' subcommand on the outgoing
dial peer as well. 'codec g711ulaw' or similar.
-- Alex
On
I'm using Queue app with Asterisk 1.4.6
It was working 5 days without problems and then it crash.
When I did #gdb asterisk core.xxx
I see...
#0 ast_senddigit_end (chan=0x0, digit=54 '6', duration=0) at channel.c:2691
#1 0xb780c7d5 in agent_answer (ast=0x925cb78) at chan_agent.c:398
#2
On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote:
I did a quick test. What happens is Congestion() answers the channel
and leaves it open. IE do a 'show channels' and you will see the
channel is still open on your end.
What happens in you pass a timeout to the Congestion()
In general section of sip.conf you can bind sip service to multiple ip
addresses. If you setup routing successfully you can send the call
received one of ip address through other ip addresses of asterisk. All
you have to do is to setup routing the right way. In this conf asterisk
can be used both
I have a customer that is complaining that any call coming in from
Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel
1.4.3 and all the MFC/R2 patches and libraries. All other calls go out
and come in, just Nextel seems to have this problem. The phone company
(catching up while my adsl is offline)
David L. West wrote:
I want callers to go into the queue(s) and just hear ringing instead
of MOH. Is this possible?
If everything else fails, you can generate a file with ringing tones, and
use that for moh.
Leif
(While my adsl is down, I'm reading old posts.)
Tom Lanyon wrote:
Hi list,
Does anyone have any advice on the following:
Incoming calls to our office come in on a SIP trunk. Since all our
offices/desks are in close proximity, we would like just a single
phone to ring when a call comes in
Dnia 2007-07-15, o godz. 14:49:27
sean [EMAIL PROTECTED] napisał(a):
asterisk-1.4.7, Fedora 7, intel emt64 - nocona:
modprobe snd-pcm-oss
--
.: Jakub Głazik,
.: email jabber: zytekatnuxi.pl
___
--Bandwidth and Colocation Provided by
El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba:
I have a customer that is complaining that any call coming in from
Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel
1.4.3 and all the MFC/R2 patches and libraries. All other calls go out
and come
Hey folks,
So I'm trying to get Festival() working on 1.2.17. I'm trying to use
app_festival:
Here's the show dialplan output from that extension:
'3378' = 1. Answer()
[pbx_config]
2. Festival(Hello Asterisk caller. How is your day?)
[pbx_config]
The Asterisk development team has released Asterisk versions 1.2.22 and
1.4.8.
These releases contain fixes for four critical security vulnerabilities.
One of these vulnerabilities is a remotely exploitable stack buffer
overflow, which could allow an attacker to execute arbitrary code on the
Asterisk Project Security Advisory - ASA-2007-014
++
| Product| Asterisk|
Asterisk Project Security Advisory - ASA-2007-015
++
| Product | Asterisk |
Asterisk Project Security Advisory - ASA-2007-016
++
| Product | Asterisk |
Asterisk Project Security Advisory - ASA-2007-017
++
| Product | Asterisk |
Hi,
compile and load of modules works fine.
After ztcfg I can see
.
.
Changing signalling on channel 1 from Unused to Clear channel
Changing signalling on channel 2 from Unused to Clear channel
Changing signalling on channel 3 from Unused to HDLC with FCS check
and then the board is frozen.
WRONG is the abbreviated answer, right? :)
If Doug is looking for a USB interface that will interface to the PSTN, he
just needs to call it a channel bank instead of a modem.
Wouldn't a Xorcom solution work for him? Are there others?
http://www.xorcom.com/products
--Don
Don Kelly
PCF Corp
On Tue, 2007-07-17 at 15:39 -0500, Victor Toofic wrote:
El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba:
I have a customer that is complaining that any call coming in from
Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel
1.4.3 and all the MFC/R2
This message arrived today 18 July NZ time
Full headers below but most of my mail is like this - the offending bit seems
to be: INXS.digium.internal which took 4 days to
deliver it
Cheers Duncan
Return-path: [EMAIL PROTECTED]
Envelope-to: [EMAIL PROTECTED]
Delivery-date: Wed, 18 Jul 2007
David L. West wrote:
I want callers to go into the queue(s) and just hear ringing instead
of MOH. Is this possible?
...use option 'r' for the Queue application. For more options,
use 'show application queue' at the CLI.
Cheers,
--
exvito
Asterisk is loading the chan_misdn and lists mISDN when issueing show
channeltypes - however it indicates Devicestate - No. when I look for
misdn show stacks, it lists the single port of the ISDN-card, however
indicates L2Link DOWN, L1LinkDOWN. so I guess theres something wrong,
On 7/17/07, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote:
I did a quick test. What happens is Congestion() answers the channel
and leaves it open. IE do a 'show channels' and you will see the
channel is still open on your end.
What happens
I thought initially it was a pretty poor generalization about postgrey and our
capabilities until I realized that this was sent a
few weeks ago when this probably wasn't an as obvious issue. But it clearly is
an issue now.
I have checked my mail servers for failures, implicitly greylisting is
In order to help you I need testcall traces, with max level of
logging, of incoming Nextel calls.
Regards,
On 7/17/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Tue, 2007-07-17 at 15:39 -0500, Victor Toofic wrote:
El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba:
I
A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot
of interoperability issues, a common troubleshooting issue was to make sure
all endpoints where using the latest version of Asterisk. I have not seen
these issues in a while.
However I've been working with a customer of
On Tue, 2007-07-17 at 19:30 -0500, Moises Silva wrote:
In order to help you I need testcall traces, with max level of
logging, of incoming Nextel calls.
Here is the log file from a couple of calls from a Nextel phone:
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
-
Hi list,
I'm trying to use an AudioCodec Mp114, 4 FXO Media gateway.
I went trough what i could find in wiki and also trixbox forum and so far no
good results.
i had this in trixbox frorum :
http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup
any
Andrew,
I don't know about your first question ... but my experience with IPcomms was
not that good ...
I was trying their service ... (DIDs) and I got a lot of dead spots in the
voice calls ... One guy from support was very friendly, trying to resolve the
issue, but I cancelled the service
Good morning, it now works, failure was due to a misconfigured/misunderstood
Class of Restriction Group Assignment for the SIP Trunk Routes on the
3300ICP.
Now Asterisk can call the world through the Mitel and incoming calls (DID,
operator transfers etc) to Asterisk via the 3300ICP, all work.
Thanks Gordon for your response,
It helped me a lot ...
I should have done this already, but the QoS issue was holding me back ...
Actually, for now ... I'll start with just a backup box and test how it goes ...
I was looking for a kind of dedicated server hosting with a MPLS network that
could
On 7/18/07, Ex Vito [EMAIL PROTECTED] wrote:
Asterisk is loading the chan_misdn and lists mISDN when issueing show
channeltypes - however it indicates Devicestate - No. when I look
for
misdn show stacks, it lists the single port of the ISDN-card, however
indicates L2Link DOWN, L1LinkDOWN.
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