I am having the same issues when asterisk gets a call and then sends it to
an Avaya system. Anyone have an idea as to what would be causing it ?
On Nov 12, 2007 3:03 PM, Mark Bell [EMAIL PROTECTED] wrote:
Hey Guys,
I have something that just started happening. When my users call each
other
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Tuesday, November 20, 2007 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ACD functionality , Skills for agents
On Nov 20, 2007
On Tue, Nov 20, 2007 at 11:47:01PM -0800, Douglas Garstang wrote:
Does anyone know where I can get an rpm spec file for zaptel 1.4.x?
Please provide feedback to bug http://bugs.digium.com/10950
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
Hi All;
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
1) As SIP or H323 client, with the ability to add
button functionalities (call pickup, call transfer,
...) so if there is a wireless network, then it can
use it to
hi,
i'm added another patch to chan_ss7
it's from Denis Smirnov http://download.seiros.ru/SeirosPBX/chan_ss7/
New in version 0.10.1 (community version)
- support for more than 256 channels
- zap style addressing
http://download.seiros.ru/SeirosPBX/chan_ss7/
bilal ghayyad wrote:
Hi All;
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
1) As SIP or H323 client, with the ability to add
button functionalities (call pickup, call transfer,
...) so if there is a wireless
Anthony Chapellier a écrit :
Robert Lister a écrit :
On Fri, Nov 16, 2007 at 02:28:45PM +0100, Anthony Chapellier wrote:
Hi all,
I know Asterisk is able to send a waiting message (audio) to people
trying to call a busy user agent using a queue. However, I'd like to
change
Here's one sip softphone for mobiles you can give a try:
http://www.minisip.org/
Regards,
Ricardo Carvalho.
___
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On Wed, 21 Nov 2007, bilal ghayyad wrote:
Hi All;
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
1) As SIP or H323 client, with the ability to add
button functionalities (call pickup, call transfer,
...) so if there
try to use http://www.fring.com/download/
On Nov 21, 2007 3:28 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Here's one sip softphone for mobiles you can give a try:
http://www.minisip.org/
Regards,
Ricardo Carvalho.
___
--Bandwidth and
On Mon, 2007-11-19 at 09:14 -0500, Brian J. Murrell wrote:
I am using asterisk 1.4.10 and seem to be having a problem with blind
transfer. This could very well be a pebkac problem but I'm not sure.
...
Called phone keys in '#':
-- Started music on hold, class 'default', on channel
Lacy Brian,
Could you please set verbosity to 10, then place your calls/holds/transfers and
post the output?
Both where it works and where it doesn't.
Otherwise, helping you troubleshoot this will be difficult.
Tony Plack
On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote:
I
Guys can someone answer how the ACD works when it needs to decide which call
to take next from queues with equal weights? Does it take the call with the
longest period of watiting or does it work randomly?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have often wondered the same thing.
It seems to me to be random, or it works it out some way I am not familiar
with. I have seen calls with wait time of 30 seconds get answered before
calls with 30 minutes wait time from queues with equal weight.
It would be great if someone who actually knows
Hi
here is my setup :
1. USER - PSTN - Asterisk A - IAX2 Trunk - Asterisk B - SER -
Asterisk C (Accepting DTMF)
All Asterisk box has dtmfmode = inband, when user pressed DTMF able to
receive and working fine.
2. Asterisk C --- Dial Customer
Customer input DTMF and its not taking any dtmf but
Kyriakos wrote:
Guys can someone answer how the ACD works when it needs to decide which call
to take next from queues with equal weights? Does it take the call with the
longest period of watiting or does it work randomly?
Whichever thread from the queue that does its processing first is
I have never tried doing this myself, but we use Bugzilla as a
well-working bug tracking tool, and it has an import script called
importxml.pl that can be used to import bugs using its own XML format. So
you would likely need some kind of AGI to create the XML and then run
importxml.pl
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Cheers Andrea
I have installed Asterisk with FreeTDS many times before (this same Asterisk
and same TDS version)... but today when I did the make it gave me this
error:
ake[1]: Entering directory `/home/matth/asterisk126/asterisk-1.2.6/cdr'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
It would be nice to add an option of choosing to answer the call with the
longest waiting time, or answer randomly, or round robin, etc...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Wednesday, November 21, 2007 4:35 PM
To: Asterisk
On Wed, Nov 21, 2007 at 11:13:38AM +0200, Tzafrir Cohen wrote:
On Tue, Nov 20, 2007 at 11:47:01PM -0800, Douglas Garstang wrote:
Does anyone know where I can get an rpm spec file for zaptel 1.4.x?
Please provide feedback to bug http://bugs.digium.com/10950
I also forgot: while the userspace
I dont know any non-linux guys who use Cygwin.
Drew Gibson wrote:
but ... why?
Zoa wrote:
Cool, i'll help out a bit with the windows port, i will start right
away with a new project on asteriskguru making nightly executable builds
and installers - will post the links in -users
I also intermittently get the Short write error followed by a cascade of Zap
alarms, the funny part is I always get an alarm clear exactly 5 seconds after
the first red alarm. The carrier always notes no drop. This happens on a
variety of digium hardware, all connected to T1 PRI, some in NFAS.
I have a Linksys sipura phone which does not dial ext 26 only, every
other ext works. When I dial ext 26 it show to:0 instead. Does anybody
know how to fix this?
Thanks in advance.
Jarga Jallow
image001.jpg___
--Bandwidth and Colocation
Hello
Some of our customers bought a bunch of phone numbers whose prefix is
the same, eg. 555-12xx - 555-1200, 555-1201, etc. There's a telco
name for this, but I forgot what it's called (think it's DID in ISDN.)
To avoid having to input all those numbers in the DB in the cidname
group, is there
On Wed, 21 Nov 2007 01:29:24 -0800 (PST), bilal ghayyad
[EMAIL PROTECTED] wrote:
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
I guess you're really looking for a (smart)phone that supports wifi in
addition to GSM, and to
Don't know if they are related, look for 26 on this page:
http://www.freepbx.org/support/documentation/howtos/howto-resolve-freepbx-and-sipura-linksys-feature-code-conflicts
--
On Nov 21, 2007 10:45 AM, Jarga Jallow wrote:
I have a Linksys sipura phone which does not dial ext 26 only,
Kyriakos wrote:
It would be nice to add an option of choosing to answer the call with the
longest waiting time, or answer randomly, or round robin, etc...
Agreed, but, understand that each queue defined in app_queue is separate. The
way the weights work is only by instructing a thread
Dear Bruno,
I had the experience of using the Vcidial with the boards of Digivoice.
It worked very well!
Leonardo Silva
Does Vicidial work together with Unicall/mfcr2 ?
Best Regards
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Wow I can't believe I missed this, and I can't believe no one else saw it!
Look at the word FROM in both the script, and the way it is called. 'From'
and 'from'... that doesn't work.
On Nov 14, 2007 8:59 AM, Matt [EMAIL PROTECTED] wrote:
I have asterisk 1.2.18 running on a new system we just
There's an application server that sits between asterisk and the gprs network
that can switch calls real time between wifi, your office pabx extensions and
the gsm network.
I've forgotten the name of it but I remember it costs $US6,000 for 10 licenses.
If you want me to find out more I can
Divitas Networks http://divitas.com/ has an asterisk based solution that
allows seamless roaming between the Wi-Fi and GSM network.
An appliance connects to or is the PBX on the office LAN and a client runs on
the smartphone. The appliance and client then coordinate which network to use
based
On Wednesday 21 November 2007 09:09:13 Matt wrote:
I have installed Asterisk with FreeTDS many times before (this same
Asterisk and same TDS version)... but today when I did the make it gave me
this error:
ake[1]: Entering directory `/home/matth/asterisk126/asterisk-1.2.6/cdr'
We don't
On Wed, Nov 21, 2007 at 11:35:42AM -0500, Dean Collins wrote:
There's an application server that sits between asterisk and the gprs network
that can switch calls real time between wifi, your office pabx extensions and
the gsm network.
I've forgotten the name of it but I remember it costs
Thanks Tzafrir, I took the stuff out of visudo - it turns out the only
way I could get this working was to create a symbolic link -
/usr/bin/asterisk to point to /home/asterisk .asterisk - using
the link created in /usr/sbin/ would not work for 'asterisk -r'
It seems that all commands in
I just bought an Aastra 480i CT for a client who needed cordless
capabilities in their office. I'm trying to set up the base station and
cordless handset in my office first. I'm able to connect the phone to
my Asterisk box and make outgoing calls from either the base station or
the handset -
Hello All,
I am hoping someone out there can enlighten me on this issue. I am using
asterisk 1.4.11. We have a call queue setup, and our agents log into the
queue. As long as no one is on the phone the queue works properly.
However, when there are agents on the phone, the queue will
I'm having this problem. Here is my output with verbosity on 10:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/2524-099012b0, SIP/2523|15)
in new stack
-- Called 2523
-- SIP/2523-09905220 is ringing
-- SIP/2523-09905220 answered SIP/2524-099012b0
-- Packet2Packet bridging
On Nov 21, 2007 11:45 AM, Tilghman Lesher [EMAIL PROTECTED]
wrote:
On Wednesday 21 November 2007 09:09:13 Matt wrote:
I have installed Asterisk with FreeTDS many times before (this same
Asterisk and same TDS version)... but today when I did the make it gave
me
this error:
ake[1]:
Am Mittwoch, den 21.11.2007, 13:47 -0500 schrieb [EMAIL PROTECTED]:
I just bought an Aastra 480i CT for a client who needed cordless
capabilities in their office. I'm trying to set up the base station and
cordless handset in my office first. I'm able to connect the phone to
my Asterisk
We just installed an Asterisk 1.4 system and have a Polycom 501 phone we
are using to test it. We have a PRI installed as well and it works well.
The problem
When a call is incoming, the caller id says:
99
sip:[EMAIL PROTECTED]
how do you get it to just say 99 and remove all
On Wed, 2007-11-21 at 08:17 -0600, Tony Plack wrote:
Lacy Brian,
Could you please set verbosity to 10, then place your calls/holds/transfers
and post the output?
I had mine set to 100 in fact and did paste the portion of asterisk
output where the transfer was happening.
But as I said in a
I have an asterisk 1.4 setup with a PRI installed and working. We are
using a Polycom 501 to test the setup..
Inbound calls work great as do phone to phone calls.
However in all cases, the caller id is a bit odd. It shows:
99
sip:[EMAIL PROTECTED]
what cause's this? How do I get just
On Nov 21, 2007 12:37 PM, Robert McNaught [EMAIL PROTECTED] wrote:
Thanks Tzafrir, I took the stuff out of visudo - it turns out the only
way I could get this working was to create a symbolic link -
/usr/bin/asterisk to point to /home/asterisk .asterisk - using
the link created in
Hi,
Is it possible to failover from Outbound SIP proxyA to Outbound SIP
ProxyB on the event that Outbound Sip Proxy A became unavailable - using
the qualify option for sip peers - it should be possible to monitor the
ping/back time, which would give us a good indication of whether a host
is up
I figured it out. Unlike the Linksys SPA942, the Web GUI interface for
configuring the phone requires Proxy Server as well as the Registrar
Server fields be populated with the IP address of the Asterisk server.
[EMAIL PROTECTED] wrote:
I just bought an Aastra 480i CT for a client who needed
On Tue, 20 Nov 2007 23:27:34 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
use dialplan function STAT()
Thanks for the tip, but it doesn't seem to work:
==
exten = 888,1,Playback(/root/asterisk_sound_files/leave_msg)
exten = 888,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
On Wed, Nov 21, 2007 at 09:37:50AM -0800, Robert McNaught wrote:
Thanks Tzafrir, I took the stuff out of visudo - it turns out the only
way I could get this working was to create a symbolic link -
/usr/bin/asterisk to point to /home/asterisk .asterisk - using
the link created in
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
handle the presentation, but you could have Asterisk rewrite the CID
name/number on the fly.
Hi, my name is Cristian, i am Argentina.
I Have asterisk 1.4.11 with libs and patchs for unicall from
http://www.moythreads.com/astunicall/. I work with mfcr2 and my configuration
is:
zaptel.conf
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
loadzone=us
defaultzone=us
unicall.conf
On Wed, Nov 21, 2007 at 01:53:58PM -0500, Kristian Kielhofner wrote:
(as root)
chown -R asterisk:asterisk /etc/asterisk
chmod -R 770 /etc/asterisk
Nitpeeking:
Now you made everything there executable.
chmod -R o= /etc/asterisk
chmod -R ug+rwX /etc/asterisk
(Any shorter way?)
In most cases
Robert McNaught wrote:
Thanks Tzafrir, I took the stuff out of visudo - it turns out the only
way I could get this working was to create a symbolic link -
/usr/bin/asterisk to point to /home/asterisk .asterisk - using
the link created in /usr/sbin/ would not work for 'asterisk -r'
It
I have the same issue and I cant fix it :(
On Nov 21, 2007 9:56 PM, Vincent [EMAIL PROTECTED] wrote:
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
Are you calling the other phones by URL or through asterisk? if your
phone is registered to asterisk, and you ask to dial a number, it will
connect through asterisk to another registered phone. If you ask to
dial a url from the polycoms, i.e. sip:[EMAIL PROTECTED], then it will connect
Tzafrir Cohen wrote:
On Wed, Nov 21, 2007 at 01:53:58PM -0500, Kristian Kielhofner wrote:
(as root)
chown -R asterisk:asterisk /etc/asterisk
chmod -R 770 /etc/asterisk
Nitpeeking:
Now you made everything there executable.
chmod -R o= /etc/asterisk
chmod -R ug+rwX /etc/asterisk
Disable URI dialing on your phones.
CP
Rob Schall wrote:
I have an asterisk 1.4 setup with a PRI installed and working. We are
using a Polycom 501 to test the setup..
Inbound calls work great as do phone to phone calls.
However in all cases, the caller id is a bit odd. It shows:
On Nov 21, 2007 2:51 PM, Vincent wrote:
Thanks for the tip, but it doesn't seem to work:
==
[...]
==
Looks like Record() always creates the file, even
if the user hung up without leaving a message.
Any other idea?
STAT() and record() are doing exactly what they are
Try playing with the options.
protocolvariant=ar,10,4,7
And please post debug output of unicall.
unicall.conf
loglevel=255
- Moy
On Nov 21, 2007 2:04 PM, [EMAIL PROTECTED] wrote:
Hi, my name is Cristian, i am Argentina.
I Have asterisk 1.4.11 with libs and patchs for unicall from
On 11/21/07, Arun Kumar [EMAIL PROTECTED] wrote:
try to use http://www.fring.com/download/
I installed out of curiosity today, and guess what? You can do SIP
over 3G (and probably wifi if you got it), plus the most unbelievable
thing - you can talk and chat over Skype.. Even on Symbian S60.. (i
I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on
a CentOS 5 server. The server has a single TE110 card connected to a
provider called Alestra in Monterrey, Mexico. Since we installed
everything we have been having problems dialing certain numbers, those
numbers
It seems that Spandsp has everything in it (when you include rxfax and
txfax) to be a T.38 termination when used with Asterisk 1.4?
And if so, what version of Spandsp?
What version of IAXModem (so I don't have to also deal with T38Modem)?
___
Started music on hold, class 'default', on SIP/2524-099012b0 --
Please post your [default] section of musiconhold.conf
Also need to know what version of Asterisk, version of kernel. Do you have
ztdummy loaded (lsmod)?
___
--Bandwidth and Colocation
There is a bug in the 480 firmware where if the callerid of the incoming
call is malformed (or basically the Aastra doesn't like, for example have a
# sign in the number), the phone won't ring.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Carlos Chavez wrote:
I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on
a CentOS 5 server. The server has a single TE110 card connected to a
provider called Alestra in Monterrey, Mexico. Since we installed
everything we have been having problems dialing certain
Asterisk version 1.4.13
Also when I listened in on a transfer it sounds like the moh is trying
to start but then immediately stop and tries to start again.
Below is my musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/moh
random=no
-Original Message-
From: [EMAIL
You need a T38 gateway of sorts, sort of like the app_t38gateway of CallWeaver.
However digium refuses to include such a program with Asterisk.
On Nov 21, 2007 6:13 PM, Robert Moskowitz [EMAIL PROTECTED] wrote:
It seems that Spandsp has everything in it (when you include rxfax and
txfax) to be
Hi Bruno,
actually vicidial is working on top of asterisk, vicidial doesn't know what
asterisk using in layer 2. SS7, ISDN stack, Unicall/mfcr2 is working with
asterisk. vicidial uses asterisk application to deliver call center
functionalities.
Regards,
Vidura.
Dear Bruno,
On Wednesday 21 November 2007, [EMAIL PROTECTED] wrote:
You need a T38 gateway of sorts, sort of like the app_t38gateway of
CallWeaver.
However digium refuses to include such a program with Asterisk.
It's not a matter of refusal; it's a matter of licensing. We don't have any
package that is
Dear all
anybody have idea about asterisk support V5.2 protocal ??
PGP Signature--
Satish Patel
mobile:- +91-9818875535
http://www.linuxbug.org
-
Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See
Hello All,
I am using ODBC storage for voicemail on my asterisk box. I want to have
a common voicemail box for different extensions.
I know how to do that, but the question troubling me is how and where do
I store the the extension name for which a particular voicemail was left.
e.g. extensions
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