Hi the following is message,Any advice appreciated, thank you.
(gdb) bt
#0 0x00429410 in __kernel_vsyscall ()
#1 0x00bead80 in raise () from /lib/libc.so.6
#2 0x00bec691 in abort () from /lib/libc.so.6
#3 0x00c2324b in __libc_message () from /lib/libc.so.6
#4 0x00c2b883 in _int_malloc ()
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
Thanks Jamie, but isn't there a
Hi friends,
I want to implement following scenario using Asterisk. Please suggest me
whether it is possible or
not.
This is bit off Asterisk and more on SIP side.
An Asterisk box with one Station(SIP channel) and PRI.
Agent dials a PSTN number of customer from station through Asterisk PRI.
Hi All,
We have a Tandberg VCS System for Video conferencing and a customer running
AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing.
Problem Statement:
How do we integrate the 2 systems such that Audio SIP calls are seamlessly
passed between the two. Sorry we're just starting up so
Hi,
for dahdi-calls I can see the current calls with dahdi show channels.
But where can I see the current call-duration or the call-start-time?
dahdi show channel n does not show this info.
-Thorsten-
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Le 17/12/2010 07:45, Gilles a écrit :
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to
reply please
On 12/17/2010 10:03 AM, Nikhil wrote:
Hi
Does anyone knows how to find out a call in a asterisk is
external incoming ,external out going or internal
Thanks
Nikhil
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Hi
Does anyone ported Asterisk to Android OS .please give details
Thanks
Nikhil
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New to Asterisk? Join us for a live introductory webinar every
Experiencing a problem when users attempt to forward a voicemail from within
VoiceMailMain(Option 8) I see the console message:
Couldn't not find mailbox XXX in context default
As why are running in a multi-tenant environment voicemail.conf has been
separated into individual contexts. The
Guys,
Why is such contradiction between 2 AMI actions QueueSummary and
Queuestatus?
Look at LoggedIn of QueueSummary and Event: QueueMember.
Also LongestHoldTime of QueueSummary does not give correct value.
-
Action: QueueSummary
Queue: retailBanking
Hi,
What is currently missing in Asterisk ecosystem to get 2 servers
active-active redundancy such as when server 1 is failing (in some
circumstances), its ongoing calls (or most of them) are kept alive and
handed over to server 2 ?
I remember that a couple of years ago, Avaya claimed it could
Asterisk Version: 1.8.0
Members are added through AddQueueMember in realtime Queues
On Fri, Dec 17, 2010 at 4:52 PM, Asterisk Man theasterisk...@gmail.comwrote:
Guys,
Why is such contradiction between 2 AMI actions QueueSummary and
Queuestatus?
Look at LoggedIn of QueueSummary and Event:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
You probably want "core show channels verbose".Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brazil+55 54 2104-7000Hi,for
- Original Message -
reply please
On 12/17/2010 10:03 AM, Nikhil wrote:
Hi
Does anyone knows how to find out a call in a asterisk is
external incoming ,external out going or internal
Thanks
Nikhil
Perhaps if you were clearer in the question you are asking ?
--
On Thu, 16 Dec 2010 11:54:31 +0100, Gilles codecompl...@free.fr
wrote:
Now, I'd like to be able to call any number on the Net that is
advertised as sip:u...@domain.com, such as those:
I mean: Do I really have to first create a section in sip.conf each
time a user needs to call a number on a new
There's no such concept in Asterisk. Everything is a call, doesn't matter its direction.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo
Le 17/12/2010 12:48, Gilles a écrit :
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone.
On 10-12-17 06:48 AM, Gilles wrote:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone.
I HATE OUTLOOK
_
reply please
On 12/17/2010 10:03 AM, Nikhil wrote:
Hi
Does anyone knows how to find out a call in a asterisk is
external incoming ,external out going or internal
Thanks
Nikhil
_
From: asterisk-users-boun...@lists.digium.com
Hi,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of
HI,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of
Has anyone seen the following in 1.4 (1.4.17)
We have istances when the number of sip channels in use multiples up
(eg: we have 40 channels in use, and then it will jump to 80, then 100+
and it will keep going upwards) and in doing this, all the channels
which are in use at that time are simply
On 10-12-17 06:17 AM, Olivier wrote:
Hi,
What is currently missing in Asterisk ecosystem to get 2 servers active-active
redundancy such as when server 1 is failing (in some circumstances), its ongoing
calls (or most of them) are kept alive and handed over to server 2 ?
I remember that a couple
I'm looking for a wireless desktop VoIP phone. Does any exist?
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The setup is this:
2 sip handsets (a Cisco 7960 and a 7961) exten 401/402
1 fxs/dahdi cordless phone, exten 201
rhino fxo/fxs analog card
asterisk 1.4.31
This is running on a Soekris 5501 with Astlinux 0.7.2
While I do have FXO capabilities, no POTS lines are connected.
When a call comes in
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic
Hi Dave,
On Thu, Dec 16, 2010 at 1:52 PM, dave george dgeo...@teletoneinc.com wrote:
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten = _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
On Fri, 17 Dec 2010 09:00:39 -0500, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
You have to tell it the host to request the extension from. All you're doing
is
dialing SIP/*031600, which with that format, is going to try and call
[*031600]
as defined in sip.conf.
You're missing the host
- Original Message -
HI,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
regards
*
No employee or agent is authorized to conclude any
On Friday 17 Dec 2010, Khaled W. Chehab wrote:
HI,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
# /etc/init.d/ircd stop
# chmod -x /etc/init.d/ircd
Should do the business :)
--
AJS
--
Ishfaq Malik wrote:
Has anyone seen the following in 1.4 (1.4.17)
1.4.17 is quite old, I'd suggest running the most current 1.4.38 and see
if it fixes your problem.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve
I'd like to find out how to block everyone outside of
the our LAN. The following phone call got through:
1. accountcode: Blank
2. src: Caller*ID number Blank
3. dst: Destination extension 901185294464086
4. dcontext: Destination context DLPN_DialPlan1
5. clid: Caller*ID with
On Fri, 17 Dec 2010, Khaled W. Chehab wrote:
How can I stop the ircd attack
This isn't an Asterisk issue.
0) Turn off your IRC service.
1) Add some rules to iptables.
2) Investigate fail2ban and see if it is an appropriate response.
--
Thanks in advance,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, December 17, 2010 9:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Freeze In 1.4
On Fri, Dec 17, 2010 at 12:40 PM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
--
Many phones like the snom 870 include a USB connector for a wireless adapter.
~~~ Andrew lathama Latham lath...@gmail.com ~~~
--
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote:
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
How would you _expect_ to be able to specify a destination server from a
telephone keypad?
--
On Fri, 17 Dec 2010, Matt wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
DECT?
(as in Siemens Gigaset)
Or are you looking for a box with handset that you can lift and a
dialpad/display on the base type of thing?
Gordon
--
At 07:40 AM 12/17/2010, you wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
Possibly one of the Aastra phones, 480i-CT or maybe a 57i-CT.
Ira
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On Fri, Dec 17, 2010 at 10:40 AM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
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Le 17/12/2010 16:52, Gilles a écrit :
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
Is that user trying to forward to xxx in the same context?
On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Experiencing a problem when users attempt to forward a voicemail from within
VoiceMailMain(Option 8) I see the console message:
Couldn't not find mailbox XXX in
This list is being attacked by some Khaled I guess. How can we stop him?
On Fri, Dec 17, 2010 at 9:34 AM, Khaled W. Chehab kche...@xplorium.com wrote:
Hi,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
Snom
Sent from Droid
On Dec 17, 2010 12:36 PM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
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New to
On Fri, 17 Dec 2010 10:40:00 -0500, Matt wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
I beleive that snom supports the use of a wifi usb dongle in the 8x0
series phones. Also, Linksys/Cisco offered an 802.11g adapter that
could be paired with their phones, making them
On Fri, 2010-12-17 at 10:40 -0500, Matt wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
Linksys Cisco SPA525 has integrated WiFi and Bluetooth
Snom 820 or 870 with optional USB adapter
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Cisco also make a wireless adapter for the 500 series phones.
On Fri, Dec 17, 2010 at 7:40 AM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
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On Fri, 17 Dec 2010, Gary Kuznitz wrote:
I'd like to find out how to block everyone outside of the our LAN. The
following phone call got through:
0) What makes you think it came from outside?
1) iptables/fail2ban
2) bind Asterisk to the IP address of the 'inside' interface. Search for
I use grandstream with the linksys/cisco adapter.
Bryant
On Dec 17, 2010, at 3:04 PM, Michael Graves mgra...@mstvp.com wrote:
On Fri, 17 Dec 2010 10:40:00 -0500, Matt wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
I beleive that snom supports the use of a wifi usb
Trying again... I think this got lost in the mailing list interruptions during
the last day or two...
- Forwarded Message -
From: Tim Nelson tnel...@fudnet.net
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 15, 2010 5:07:20 PM
Subject: [asterisk-users] Echo Cancellation
Grandstream GXV3140 has a WiFi USB adapter.
Alex Saavedra
On Fri, Dec 17, 2010 at 11:40 AM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
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I belive the WBP54g cisco/LINKSYS adapter is what we are using with the
Grandstream phones. You have to buy a Cisco/Linksys power supply but it
works great. I have over 200 of them out there.
Bryant
From: Jeremy Betts jer...@freevoicepbx.com
Sent:
On 17/12/10 5:56 PM, Olivier wrote:
Hi,
Did you use libpri 1.4.11.5 or 1.4.12-beta ?
Recently l tried 1.4.11.5 on a live system and it failed (Asterisk
1.6.1.18 and dahdi trunk, Junghanns QuadBRI, PtmP lines).
Going back to 1.4.11.2 solved it.
Unfortunately, I couldn't note what error message
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