Matt Riddell li...@venturevoip.com writes:
On 22/7/09 7:24 PM, Scott Gifford wrote:
[...]
In this case, I don't seem to have enough information to tell when the
call has failed and I should give up. I do get a Hangup event, but I
don't see a way to distinguish it from other hang-up events
Dear All,
i need help on Shared channel variable
can any body have example of SHARED function which implemented in 1.6
version
i can not find example
regars
Dhaval
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi all,
You may have heard yesterday that Adhearsion and Voxeo have created a
new baby, Voxeo Labs. From our (non-biz) point of view, I'd recommend
following the blogs: http://blogs.voxeo.com/ to see how what they do
might be of interest to you and your asterisk/voip activities,
commercial or
Hi,
I want to activate DND on ast 1.6.0.9 with asterisk-gui.
Is there special commands that i need to use during such script
or simply writing a code in extensions.conf that checks if the user has a
DND=yes value on ast. database and act according to that like forwarding
call to voicemail or
Hmm I was given the impression that the .call files were risky due to
locking issues... Is this no longer the case perhaps?
I also require knowledge of whether the originate was successful or
otherwise, with BUSY vs CONGESTION, etc.
From:
Hello!
I am looking for a way to test if a SIP device is still alive or not.
I want to add this functionality in an AGI or independent script in
order ensure all the SIP phones are properly connected to the system.
Thank you,
Elliot
___
-- Bandwidth
Ketema Harris wrote:
hello all...I have been trying to get a handle on CallerPres with an
analog handset. I have usecallingpres=yes in my chan_dahdi.conf file
member:file and when I dial *67 on my analog handset I see Disabling
Caller*ID on DAHDI/4-1 but when the call is then forwarded to
Maybe https://issues.asterisk.org/view.php?id=71 should be re-opened
because the north American vertical service codes are still hard-
coded in Zaptel/Dahdi.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer,
Asterisk can 'ping' the clients behind NAT with the qualify-option so
the NAT-tables and routes are kept open.
What happens when one resets the router (where the NAT-tables are
kept) ?? Do NAT-tables get flushed when a router is reset ??
Does the public IP-address needs to be a static
Elliot Murdock schrieb:
I am looking for a way to test if a SIP device is still alive or not.
What about qualify=yes in sip.conf?
I want to add this functionality in an AGI or independent script in
order ensure all the SIP phones are properly connected to the system.
Philipp Kempgen
--
I've just had this Static/Dynamic IP issue in the last couple of days
Any time the IP address changes the phone needs to be re-registered.
This normally isn't that much of a problem as most people only reboot
their routers about twice a year. However, here's a warning to anyone UK
based. BT
On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote:
Olivier wrote:
Hi,
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
If positive,
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
It certainly would seem possible and would be a great feature request.
There probably is no circuitry existing to do it, but I would assume that
ohms, volts, or something could be measured while sending a
Benny Amorsen wrote:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT
Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.
A call B, B press *2 and voice announce to digit internal and select
internal of C. CORRECT
A hear music on hold and B talks with C. CORRECT
If B press *0, the call return
Steve Totaro schrieb:
On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote:
Olivier wrote:
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
If
Have you monitored the call from CLI with verbose set up? What happens if
you use regular AGI instead of FastAGI?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Keiron Liddle
Sent: Wednesday, July 22, 2009
Why didn't you just do 1.4.26 or 1.4 SVN? What release did you have? Did
you (or the update) change your DTMF settings?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo
Sent: Thursday, July 23, 2009 7:55 AM
To:
Hello Philipp,
Thank you.
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could use?
Regards,
Elliot
On Thu, Jul 23, 2009 at 12:47 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Elliot Murdock schrieb:
I am looking for a way
Yes. I have sendrpid = yes in sip.conf. CallerPres works fine with
sip handsets.
On Jul 23, 2009, at 4:29 AM, Ishfaq Malik wrote:
Ketema Harris wrote:
hello all...I have been trying to get a handle on CallerPres with an
analog handset. I have usecallingpres=yes in my chan_dahdi.conf file
Elliot Murdock schrieb:
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could use?
You could have a script execute
asterisk -rx 'sip show peers'
and read the status for each peer.
On Thu, Jul 23, 2009 at 12:47 PM, Philipp
Hi
You can retrieve it in real time using the AMI from a script
http://www.voip-info.org/wiki/view/Asterisk+manager+API
Ish
Elliot Murdock wrote:
Hello Philipp,
Thank you.
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could
On Thu, Jul 23, 2009 at 7:50 AM, randulo spamsucks2...@gmail.com wrote:
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
It certainly would seem possible and would be a great feature request.
There probably is no circuitry existing to do it, but I
Hi,
Still I can manage to have good incoming calls from h323. Can someone give
me a hand?
Regards,
LS
Date: Thu, 16 Jul 2009 15:46:43 +0100
From: Luis Silva luis.si...@dreamware.pt
Subject: [asterisk-users] H323 situation
To: asterisk-users@lists.digium.com
Philipp Kempgen wrote:
Steve Totaro schrieb:
On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote:
Olivier wrote:
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an
Hi
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors - Music 2
Technical Support -
On Thu, Jul 23, 2009 at 9:35 AM, Juan C. Crespo R.jcre...@ifxnw.com.ve wrote:
Hi
Guys I wonder if its possible to set a different MoH based on groups, I
mean if one of the Admin group put on hold the call play music 1, if another
from Technical Support put on hold the call play music 3,
I am a Linux sysadmin who has been tasked with developing the phone
system for our nonprofit's new US headquarters building. We cannot
bring our legacy phone system with us, so I am building this completely
from scratch. I have already read Asterisk: The Future of Telephony
and done a fair
My .02 - IAX may not be an option and is probably not a good one if it is.
It requires a good bit of overhead to work reliably and well. You won't go
wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16 port
card and make sure you get the card away from any existing IRQ's,
Juan C. Crespo R. schrieb:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors
Depending on how your dialplan is set you can use the
SetMusicOnHold
application after creating classes in your musiconhold.conf
http://www.asteriskguru.com/tutorials/setmusiconhold.html
Ish
Juan C. Crespo R. wrote:
Hi
Guys I wonder if its possible to set a different MoH based on
On Thu, Jul 23, 2009 at 5:08 PM, Danny Nicholasda...@debsinc.com wrote:
My .02 - IAX may not be an option and is probably not a good one if it is.
It requires a good bit of overhead to work reliably and well. You won't go
wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16
Lyle Giese wrote:
Philipp Kempgen wrote:
Steve Totaro schrieb:
On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote:
Olivier wrote:
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to
I am running an AGI 1.4.26
A person answers the call, and presses a DIGIT really fast. Perhaps
while the AGI is still starting up.
Is there anyway to get that digit?
When doing wait for digit if my AGI is up and running I seem to get
the digit every time.
Is there a way/method to get ANY
Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone
know exactly what settings needed to reach the asterisk server on my
home network?
Internet -DSL transparent bridge -router -asterisk
-softphone
x-lite
Hello
I have 2 queues and I would like to send calls to queue_1 and queue_2
dynamically.
For example:
If I have 10 agents logged (2 in queue_1 and 8 in queue_2)
I want 20% of the calls to be sent to queue_1 and 80% to queue_2
Is this possible?
Is there a way I can see how many logged (or
Does your asterisk has a private or public IP?
Is your X-Lite in your LAN or outside? If X-Lite is outside the Lan, you
need to forward all traffic coming to your Lan in port 5060, to
asterisks private IP.
Activate SIP debug in asterisk CLI to check if the traffic is getting to
asterisk.
Joao
Joao Gomes Pereira wrote:
Does your asterisk has a private or public IP?
Is your X-Lite in your LAN or outside? If X-Lite is outside the Lan, you
need to forward all traffic coming to your Lan in port 5060, to
asterisks private IP.
Activate SIP debug in asterisk CLI to check if the traffic
I've got a couple of PRIs. When I call out on them from internal SIP phones,
I will get ringing if the dialed number is ringing, but if the dialed number
is busy I'll get dead air. Can anyone suggest ways to trouble shoot this?
Don't seem to having any other problems with the PRIs.
Thanks,
David
On Thursday 23 July 2009 07:24:46 Leif Madsen wrote:
Benny Amorsen wrote:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
You could do an AGI to get the queue information via AMI queue status, then
return variables to the dialplan and select the queue dynamically based on
that information.
[global]
CALLCOUNT=0
- exten = s,1,answer
- exten = s,2,AGI(questat.agi)
- exten = s,3,set(GLOBAL(CALLCOUNT)=[1 +
David Ruggles wrote:
is busy I'll get dead air. Can anyone suggest ways to trouble shoot this?
Don't seem to having any other problems with the PRIs.
It'd be nice to start with what version of Asterisk, what distro, who is
your service provider and snippets of your config.
On our PRIs we
Apologies. Didn't mean to omit key information, I doubt it's a problem with
* because everything else is working great so I was asking for help on
troubleshooting the PRI.
Anyway, here's the 411:
Asterisk 1.4.20, CentOS 5.2
Service Providers: Quest Deltacom, Local Loops provided by Embarq
What
David Ruggles wrote:
[channels]
I also have listed pridialplan=unknown
immediate=no
;Sangoma A102 port 1 [slot:8 bus:1 span:5] wanpipe5
I've got two sites running the Sangoma A101, what version of your
wanpipe drivers are you running (Mine are probably very outdated WANPIPE
On Thursday 23 July 2009 02:05:38 DHAVAL INDRODIYA wrote:
Dear All,
i need help on Shared channel variable
can any body have example of SHARED function which implemented in 1.6
version
It's actually fairly simple. On each channel, there is a space accessible for
other channels to write:
Anyone successully connected to nortel cs 1000 switch?
Care to share you switch settings?
I have asterisk 1.4.25, libpri 1.4.7, dahdi
We tried national and the verizon guy said that wasnt working...
We tried 5ess and we can get external calls - but internal calls we have
no audio.
I see frame
In australia, I would usually suggest a mix of E1 and SIP for calls - it
doesn't cost any money to receive calls via E1, and redundancy is an
old, valuable friend of mine.
PaulH
Stephen Fierbaugh (PBT) wrote:
I am a Linux sysadmin who has been tasked with developing the phone
system for our
Can Asterisk be configured to hang up if another phone picks up?
I'm a bit lost as far as terminology goes, but here's my setup. At
home, I have asterisk answering calls from the pstn and sending them
through to a sip extension or voicemail. All that is working fine.
The box running Asterisk
On Thu, Jul 23, 2009 at 8:34 PM, Trevor Hammonds tre...@concipient.netwrote:
Bill Lovett wrote:
Can Asterisk be configured to hang up if another phone picks up?
I'm a bit lost as far as terminology goes, but here's my setup. At
home, I have asterisk answering calls from the pstn and
Marco Sambo a écrit :
Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.
[...]
Marco,
attented transfer are broken in 1.4.25, please upgrade to 1.4.26 (see
changelog).
--
Daniel
___
--
Jerry Geis wrote:
Anyone successully connected to nortel cs 1000 switch?
Care to share you switch settings?
I have asterisk 1.4.25, libpri 1.4.7, dahdi
We tried national and the verizon guy said that wasnt working...
We tried 5ess and we can get external calls - but internal calls we have
no
Short answer - no.
Leave the box on 24/7, and run the POTS phone through an ATA, or another
SIP phone.
If power consumption and wear and tear is a consideration, use AstLinux
on a thin client, and reduce your power consumption to under 30 Watts.
John Novack
Bill Lovett wrote:
Can Asterisk
An exclusion adapter is overkill. My Asterisk line card is the $10 Win
modem card that I got from ebay.
When you call my copper line, two devices see the inbound ringer:
1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the
calls
2. Asterisk with a win modem line card that:
Shorter answer is yes :-).
This is exactly how mine runs. The secret is that the copper interface
will ring a SIP extension but just exit from the dialplan on noanswer.
[main-copper]
exten = s,1,Dial(SIP/22,69)
and then nothing in my case.
Generally my wife answers using a cordless phone set
Bill Lovett wrote:
Can Asterisk be configured to hang up if another phone picks up?
I'm a bit lost as far as terminology goes, but here's my setup. At
home, I have asterisk answering calls from the pstn and sending them
through to a sip extension or voicemail. All that is working fine.
The
I get how everything is connected with your setup, but if you pick up
the cordless phone to answer a call does the sip extension just keep
ringing until it times out?
I like the exclusion adapter idea because it sounds like it would let
me keep my dialplan intact. But I do take John and
On Thu, Jul 23, 2009 at 8:58 PM, Tom Browning ttbrown...@gmail.com wrote:
An exclusion adapter is overkill. My Asterisk line card is the $10 Win
modem card that I got from ebay.
When you call my copper line, two devices see the inbound ringer:
1. The Uniden 5.8Ghz cordless phone base
If you don't have an objection to 24/7 then that is by far the best way,
just get some fxs ports and each POTS phone can have it's own extension if
you want.
Certainly the way to go if there is no reason stopping you.
On Thu, Jul 23, 2009 at 9:20 PM, Bill Lovett b...@ilovett.com wrote:
I get
The box running Asterisk isn't on 24/7 so I have a secondary phone
connected to the line as well. If Asterisk is not running, I can
answer an incoming call from that phone. If asterisk is running, I can
answer the call from a sip extension.
Can I have it both ways? Can Asterisk back off if
Just a little clarification for people refering to Asterisk as a PBX
and not an Answering Machine:
In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk
is a Telephony Toolkit. You can choose to use it as a PBX or an
Answering Machine or both or even in some case as a
Pascal Bruno wrote:
Just a little clarification for people refering to Asterisk as a PBX
and not an Answering Machine:
In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk
is a Telephony Toolkit. You can choose to use it as a PBX or an
Answering Machine or both or even in
I asked this question a while back before Dahdi and have been using
the X100P cards, but my understand is they will not have native
support under Dahdi. What is the best option for installs that are
pure SIP, but want to do reliable conferencing?
Thanks!
Yes I have monitored it on the CLI and everything appears to work
correctly but something is going wrong internally.
I tried it with a php agi and it does work properly, so I guess it could
be something to do with the fastagi. Even though the script is simple
(at the moment) I would prefer to
That's right, they say it is a PBX because it is mostly used as such, but
it is more than just a PBX. Some people use it as a VoiceMail tool or to
handle just conference, some use it to add functionalities to other legacy
PBX systems. Calling cards applications for example, a plain PBX wont be
Yeah, except in the OP he mentions that he wants or is at least using
Asterisk VM so your solution does not meet his needs.
Ah, yes. My config would not allow Asterisk to be a part time voicemail
destination. In my config, the POTS line has its own voicemail (it is
actually a Comcast line
I get how everything is connected with your setup, but if you pick up
the cordless phone to answer a call does the sip extension just keep
ringing until it times out?
Actually no, the SIP extension stops ringing and Asterisk takes no further
action.
I like the exclusion adapter idea
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