Hi there everyone,
I am a bit confused these days due to some problem I am having. Its not a
technical problem. Asterisk is working fine. Most of the users are happy,
but some handful of users are getting calls in the middle of the night even
though they have enabled Anonymous Call Rejection
On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote:
Still last night there was a call to a customer. Plz help me figure out the
solution for this problem.
Can you be sure that the call _is_ coming through your Asterisk server,
rather than being the result of random scanning for your
use the timeout option in the Dial application like so
Dial(SIP/trunk,120)
If you dont specify the timeout the default timeout used bya sterisk is
probably more than 60 seconds.
On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote:
Hi
Does anyone know how i could extend
Thats what im unsure about. I think the calls maybe going to the user
directly through sip uri or some other method. How can i test that. I have
already tried to call those customers with direct sip uri dial but does not
work.
On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West
try this
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk aster...@ck-lee.comwrote:
I have a sip trunk connecting to a huawei softx3000. At the moment, I can
register and dial in.
However, peer status shows not reachable
sip show
Hello
The following, dead simple Bash script ran as AGI doesn't reply to
Asterisk:
= extensions.conf
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(CID=${CALLERID(num)})
exten = s,n,AGI(/var/tmp/basic.agi)
exten = s,n,Hangup()
= /var/tmp/basic.agi
#!/bin/bash
#Ripped
sorry i wasnt clear enough i meen inbound
On Thu, Feb 24, 2011 at 12:25 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
use the timeout option in the Dial application like so
Dial(SIP/trunk,120)
If you dont specify the timeout the default timeout used bya sterisk is
probably more than 60
The relevant part of my setup is something like:
SIP phones - local server - remote server - SIP-to-PSTN provider
I want _some_ of the SIP phones on the local server to be able to get
access to SIP-to-PSTN, but not all of them. The local-to-remote
connection is IAX2 over VPN.
Do I need to set
On Thu, 24 Feb 2011 11:56:25 +0100, Gilles codecompl...@free.fr
wrote:
The following, dead simple Bash script ran as AGI doesn't reply to
Asterisk:
Turns out Bash doesn't allow empty loops. This version does reply as
expected:
==
#!/bin/bash
read line
while [[ $line != ]] ; do
Hello
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
The whole thing works fine when the original call that triggers
Asterisk is from an internal extension (Xlite), but it fails when it's
from my cellphone ringing through the
In article jg9cm6pqkit0q3oi5aacabi7dfql7st...@4ax.com,
Gilles codecompl...@free.fr wrote:
Hello
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
The whole thing works fine when the original call that triggers
Asterisk is from an
Hi list,
Currently, one of my phones registers fine, and the other does not,
though for me they have the same config...
Can somebody help debug/understand why?
The logs in asterisk say:
[Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from
So you have an IP network, with SIP agents (cell phones ?), some of
those are manually
setup in you sip.conf file, but you want to allow unknown cell phones
users to self register
in your system ?
Yes, exactly.
Someone enter your network, dial 3001@your ipbx and get/set a
temporary
Hi Danny,
That's a nice log I'll try and do the same with a higher verbosity
level on my side too.
Just to make sure
- who called 3001? the roaming phone that had no extension yet?
-- Executing [3001@default:1] Verbose(SIP/sipuser-006f, Create
roaming extension) in new stack
- when you
Hi,all
I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are
opened, externip is configured in sip.conf. I think, that all relevant
configurations are checked. But, no voice hear between local and remote
extension. What I need to check, configure in router and PBX for resolving
On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote:
The relevant part of my setup is something like:
SIP phones - local server - remote server - SIP-to-PSTN provider
I want _some_ of the SIP phones on the local server to be able to get
access to SIP-to-PSTN, but not all of
Try something like this,
[general]
localnet=192.168.0.0/255.255.0.0 ; or your subnet
externip=x.x.x.x ; use your address
[YOURREMOTEPEER] ; your peer's name
nat=yes
qualify=yes; Force keepalives
On Thu, Feb 24, 2011 at 7:12 PM, Oleg Botvinkin
Do you have PRI card or FXO card?
--
Sent from my iPhone
On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com
wrote:
Thats what im unsure about. I think the calls maybe going to the
user directly through sip uri or some other method. How can i test
that. I have already
Its a pure VoIP setup. no cards.
On Thu, Feb 24, 2011 at 7:12 PM, Satish Patel satish...@hotmail.com wrote:
Do you have PRI card or FXO card?
--
Sent from my iPhone
On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:
Thats what im unsure about. I think the calls
Anybody else noticed that caller id for outbound calls via Google Voice
seems to be broken? It seems to be a Google Voice problem though, not an
asterisk issue.
--
Chris
--
_
-- Bandwidth and Colocation Provided by
We are getting a lot of calls identified as Asterisk or out of area in
the middle of the night.
From other posts on the list, I have assumed these are null Caller ID calls
and Asterisk is plugging in pseudo ID. Is that correct?
It seems to me that Asterisk should simply say no caller ID
What kind of broken are you seeing.
It could be the ID is pseudo ID and may never reflect the actual caller.
CF
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Thursday, February 24, 2011 8:52 AM
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Thursday, February 24, 2011 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Google Voice outbound Caller ID broken
Anybody
On Thu, Feb 24, 2011 at 9:08 AM, William Stillwell
will...@stillwellsoft.com wrote:
Yes.. google it
I did. :)
This is what I have done to resolve it (I posted a few days ago on this)
exten = _9NXXNXX,1,Dial(gtalk/(value in gtalk.conf)/+1(googlevoice#)@
On Thu, 24 Feb 2011, Gilles wrote:
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
I don't think it has anything to do with the method used to create the
call file.
AGI script
#!/var/tmp/lua
--Must first empty stdin
while
you can also set some kind of authentication on the extensions for example
ask for a pin to dialout. etc
On Thu, Feb 24, 2011 at 6:51 PM, Daniel Tryba dan...@tryba.nl wrote:
On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote:
The relevant part of my setup is something like:
On Thu, 24 Feb 2011, Gilles wrote:
= /var/tmp/basic.agi
#!/bin/bash
#Ripped from
#http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html
while read -e ARG [ $ARG ] ; do
done
echo NOOP Here
while read line
do
On Thu, 24 Feb 2011, Gilles wrote:
Turns out Bash
Hi List,
We have 3 Asterisk servers (1.4 1.6) with about 200 users all
connecting over the internet. Our biggest problem is with dropped audio.
My question is what is the best way to debug this? Searching on the
internet does not turn up a lot of results for dropped audio. It seems
most
Chris,
Can you please provide more details.
What do you exactly mean by broken? Do your call recipients get a
random CID?
Have you tried to call from the GMail WEB interface? Are you getting
the same result?
-Vladimir
On 2/24/2011 8:51 AM, Chris Gentle wrote:
Anybody else noticed that
On Feb 23, 2011, at 4:39 AM, Ishfaq Malik wrote:
Hi
We're still testing out asterisk 1.8 (using 1.8.2.2 from rpm package)
before putting it into production and I'm observing an odd issue when
using the AMI
Every request I send to the AMI just results in a FullyBooted response
rather
On Thu, Feb 24, 2011 at 1:31 AM, Faisal Hanif fai...@vopium.com wrote:
PRI start from 3055 to 30550100 i have purchased a 100 number from
telco and our pilot number is 3055, now if some caller want to dial any
number but caller should shown is 30550008 like this.
Set the
On Thu, Feb 24, 2011 at 5:01 AM, Israel Gottlieb isr...@gmail.com wrote:
sorry i wasnt clear enough i meen inbound
You could always Answer() the call in your dialplan before you do anything
else, then Dial() whoever you're trying to reach and set your own timeouts
there.
--
Thanks,
--Warren
On Thu, Feb 24, 2011 at 7:24 AM, Axelle aaforti...@gmail.com wrote:
Hi list,
snip
in /etc/asterisk/extensions.conf:
exten = 2102,1,Macro(dialSIP,IMSI2081) ; this one registers ok
exten = 2111,1,Macro(dialSIP,IMSI20830061) ; fails
These lines have nothing to do with endpoint
On Thu, Feb 24, 2011 at 7:31 AM, Axelle aaforti...@gmail.com wrote:
Hi Danny,
Try this code, it uses the SIPCHANINFO function to get the peername of the
device that's attempting to create the roaming extension, instead of the
callerid. Basically, you need to store some kind of contact info
Hi,
I have this same behaviour on version 1.8.2.3 build from source. We are using
AMI to originate call from our CRM software, but we ignore that message.
The patch for the bug at https://issues.asterisk.org/view.php?id=18168 has been
committed (thanks FeyFre!). The FullyBooted event will
On Thu, Feb 24, 2011 at 10:32 AM, Jesse Cloutier je...@cronomagic.comwrote:
Whats the best way to start tracking this down?
Collect proper debug information[1].
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Thanks,
--Warren Selby, dCAP
Hi,
My phones stopped auto-answering when being paged, since I moved on to
Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk
1.6.2.16.
I looked at the wiki but nothing I try there works, even if I cut and paste
the same setup.
Any one has any idea of what I
If you compare a working config with a non-working you will see something with
the answer type. I had that issue until I down rev'd. Look for something like
Ring Answer, I forget the exact details now.
From: asterisk-users-boun...@lists.digium.com on behalf
Hi Terry,
I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.
Mike
From: asterisk-users-boun...@lists.digium.com
From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x
Hi Terry,
I did that, and did find a
On Thu, Feb 24, 2011 at 1:41 PM, Mike l...@net-wall.com wrote:
Hi,
My phones stopped auto-answering when being paged, since I moved on to
Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk
1.6.2.16.
I looked at the wiki but nothing I try there works, even if I
From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x
Hi Terry,
I did that, and did find a
Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.
Mike
Hi Terry,
I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am
Is 3.3.x downloadable for non-paying people yet?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
Polycom are at 3.3.1 now, so 3.3.0 should be fair game.
It has nothing to do with paying or not, the company that sold you the phone
should be able to give you the latest version no? Unless you bought from a
guy who found a box that fell off a truck.or some third-rate reseller.
Mike
Hi to all!
I'm trying to create a context for integration with extensions.lua and
libsql.mysql, but I'm not getting to run. When I reload the module
pbx_lua.so the following error appears:
[Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua
extension: error loading module
Sorry, I realize my tone might not go down well. I didn't mean to sound
like a jerk, but I was just stating that resellers are also authorized to
distribute the firmware to their customers if I recall correctly, so
everybody can get the firmware for free, just not directly from Polycom.
And
Hello!
I bought a virtual IP line to my ISP to use with my asterisk but when I try to
connect it to my ISP tells me I can not use and I can only use with a softphone
that gives me, xlite ready configured.
I use ngrep to see what information sent on xlite for communication, the
User-Agent was
Actually, I don't think that has been the case for quite a while. Anyone
can get the latest firmware directly from polycom. Including, 3.3.1F
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
On 02/24/2011 03:32 PM, Mike wrote:
Sorry, I realize my tone might not go down
I had an issue today where receive_fax caused an asterisk switch to crash.
The switch is 1.8.2.3 version. The call was coming from a fax machine. The
call started receive_fax answered and then asterisk stopped responding. I
was able to log into asterisk but it would not do a core restart now nor
On Thu, Feb 24, 2011 at 6:05 PM, Bryant Zimmerman brya...@zktech.com wrote:
I had an issue today where receive_fax caused an asterisk switch to crash.
The switch is 1.8.2.3 version. The call was coming from a fax machine. The
call started receive_fax answered and then asterisk stopped
On Thu, 24 Feb 2011 08:27:13 -0800 (PST), Steve Edwards
asterisk@sedwards.com wrote:
Bash has a thing about syntax too. Note you're not 'done' with your second
loop.
Sorry about this :-/
--
_
-- Bandwidth and Colocation
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten = _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten = s,1,AGI(getchannel.php|${ARG1})
exten = s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
exten = s,3,Hangup()
but for some
On 11-02-24 04:08 PM, Andrew Latham wrote:
There are many updates in 1.8.2.4 that may fix your issue. If you are
running any version of 1.8 it should be a quick update.
I wouldn't say many. There is one fix in 1.8.2.4 over 1.8.2.3.
From the ChangeLog:
* Asterisk 1.8.2.4 Released.
On Thu, Feb 24, 2011 at 7:43 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 11-02-24 04:08 PM, Andrew Latham wrote:
There are many updates in 1.8.2.4 that may fix your issue. If you are
running any version of 1.8 it should be a quick update.
I wouldn't say many. There is one fix in
On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Has this issue been fixed in this release of 1.8 (or even in the
previous 1.8.2.3)?
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
Thanks
Ish
Ishfaq, I spoke to soon and was looking at the
On Feb 24, 2011, at 5:27 PM, Ron wrote:
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten = _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten = s,1,AGI(getchannel.php|${ARG1})
exten =
On 2/24/2011 9:10 PM, Ben Klang wrote:
On Feb 24, 2011, at 5:27 PM, Ron wrote:
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial
plan:
[callback-outbound]
exten = _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten = s,1,AGI(getchannel.php|${ARG1})
exten =
Chris,
Let me summarize:
1. GV Outbound CID shows Unknown, Unavailable, Out of area
(depending on a recipient's carrier) starting some time around
02/15/2011 if a call is placed via Google Chat/Google Talk/Google
Mail/Asterisk GTalk channel. See
Further analysis showed that a call placed using a GTalk channel which
came as Restricted was not recorded under History / Placed in Google
Voice.
A call placed using the same GTalk trunk an hour later was terminated to
the same recipient's phone with the proper CID.
It looks like a call routing
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