Re: [asterisk-users] ChanSpy : how to know channel name ?
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote:
Did a little research on this using my Asterisk 10.0. This should
work for you.
exten = 1246,1,answer()
exten = 1246,n,set(inuse=${CHANNELS(miq8)})
exten = 1246,n,extenspy(${inuse}@default)
exten = 1246,n,hangup()
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 9:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
there is very little information about the function CHANNELS().
If I know the peer name (that I always know for sure), do you see a
way of using the function CHANNELS() to get the right channel ??
If CHANNELS() gives a space-delimited list of active channels, and I
know miq8... how can I get /SIP/miq8-2419/ ?
Thanks !
On 01/24/2012 04:46 PM, Danny Nicholas wrote:
Extenspy(miq8@default) for miq8. I would either proceed under the
assumption that I'm going to be listening to my extensions in the
default context or set up an AGI or something to load my needed
ext@context information.
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 9:41 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
how to use ExtenSpy(extension@context) when conversations are named
like this ? :
/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419* sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
/
Can you tell me what is the extension ? How will I know the context ?
The context is not always the same...
On 01/24/2012 04:32 PM, Danny Nicholas wrote:
You are either going to be able to listen to SIP/miq8 or you are going
to have to know the sequence number like SIP/miq8-1. Maybe you
should just use ExtenSpy instead?
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 9:26 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?
Of course I can control the name of my SIP-peer. Why do you tell me
this ?!
Please answer my question : how do I know the channel name so I can
ChanSpy the correct channel ?
On 01/24/2012 04:13 PM, Danny Nicholas wrote:
It's not random. The Channel Name is Tech/peer-sequence (sequence
is in hex). You can control (to a degree) the peer portion in
sip.conf/users.conf.
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 9:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
thanks. miq8 is the name of the SIP peer account.
So when I know the SIP peer name, and I strip of the numbers of the
channel, then I can use ChanSpy. So this answers my original question.
The only problem I see : it is Asterisk that gives the channel its
name. How do I change this ??
As far as I know, Asterisk randomly gives a channel name which
consists of the technology (SIP), the peername (miq8) and some numbers...
How to change the channel name ?
On 01/24/2012 03:53 PM, Danny Nicholas wrote:
I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen
to a bridged call which (it seems to me) should pick up both sides.
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 8:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
OK thanks. But, I want to listen to the conversation (not just 1
channel out of 2 channels). How then do I use ChanSpy ?
On 01/24/2012 03:41 PM, Danny Nicholas wrote:
Strip off the --x. Just listen to SIP/miq8 and SIP/375382280 in
your example.
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On
Re: [asterisk-users] ChanSpy : how to know channel name ?
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
Have you tried giving it a go?
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally whisper into it.
[Description]
This application is used to listen to the audio from an Asterisk
channel.
This includes the audio coming in and out of the channel being spied
on.
If the 'chanprefix' parameter is specified, only channels beginning with
this
string will be spied upon.
While spying, the following actions may be performed:
- Dialing '#' cycles the volume level.
- Dialing '*' will stop spying and look for another channel to spy on.
- Dialing a series of digits followed by '#' builds a channel name to
append
to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
the
digits '1234#' while spying will begin spying on the channel
'Agent/1234'.
Note that this feature will be overridden if the 'd' option is used
NOTE: The X option supersedes the three features above in that if a
valid
single digit extension exists in the correct context ChanSpy will exit
to
it. This also disables choosing a channel based on 'chanprefix' and a
digit
sequence.
[Syntax]
ChanSpy([chanprefix][,options])
[Arguments]
options
b: Only spy on channels involved in a bridged call.
B: Instead of whispering on a single channel barge in on both
channels
involved in the call.
c(digit):
digit - Specify a DTMF digit that can be used to spy on the
next available channel.
d: Override the typical numeric DTMF functionality and instead use
DTMF to switch between spy modes.
4 - spy mode
5 - whisper mode
6 - barge mode
e(ext): Enable *enforced* mode, so the spying channel can only
monitor
extensions whose name is in the ext : delimited list.
E: Exit when the spied-on channel hangs up.
g(grp):
grp - Only spy on channels in which one or more of the groups
listed in grp matches one or more groups from the ${SPYGROUP}
variable set on the channel to be spied upon.
NOTE: both grp and ${SPYGROUP} can contain either a single group
or a colon-delimited list of groups, such as
'sales:support:accountin
g'.
n([mailbox][@context]): Say the name of the person being spied on
if that person has recorded his/her name. If a context is specified,
then
that voicemail context will be searched when retrieving the name,
otherwise
the 'default' context be used when searching for the name (i.e. if
SIP/1000
is the channel being spied on and no mailbox is specified, then
'1000'
will be used when searching for the name).
o: Only listen to audio coming from this channel.
q: Don't play a beep when beginning to spy on a channel, or speak
the selected channel name.
r([basename]): Record the session to the monitor spool directory.
An optional base for the filename may be specified. The default is
'
chanspy'.
s: Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.
S: Stop when no more channels are left to spy on.
v([value]): Adjust the initial volume in the range from '-4' to
'4'. A negative value refers to a quieter setting.
w: Enable 'whisper' mode, so the spying channel can talk to the
spied-on channel.
W: Enable 'private whisper' mode, so the spying channel can talk
to the spied-on channel but cannot listen to that channel.
x(digit):
digit - Specify a DTMF digit that can be used to exit the
application.
X: Allow the user to exit ChanSpy to a valid single digit numeric
extension in the current context or the context specified by the
${SP
Y_EXIT_CONTEXT} channel variable. The name of the last channel that
was
spied on will be stored in the ${SPY_CHANNEL} variable.
On Wed, 2012-01-25 at 10:15 +0100, Jonas Kellens wrote:
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote:
Did a little research on this using my Asterisk 10.0. This should
work for you.
exten = 1246,1,answer()
exten = 1246,n,set(inuse=${CHANNELS(miq8)})
exten = 1246,n,extenspy(${inuse}@default)
exten = 1246,n,hangup()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
there is very little information about the function CHANNELS().
If I know the peer name (that I always know for sure), do you see
[asterisk-users] Signalling and Media Configuration
Hi, I was provided with a bunch of ip addresses to be configured on my asterisk server. the first batch contains the signalling which has 2 ip addresses the second batch contains media which has 4 ip addresses How will I configure it to my asterisk box? Should I simply add the signalling to my sip.conf? Tried it as peer but they are unreachable. thanks! -rai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
Am 23.01.2012 um 23:25 schrieb Alec Davis: How can I test this solution on a 1.8.8.1 system ? If I'm not mistaken, diff https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1. I've just checked out 1.8.8.1 and download my patch from https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean, using the following on a debian lenny box: svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 asterisk-1.8.8.1 cd asterisk-1.8.8.1 wget --no-check-certificate https://reviewboard.asterisk.org/r/1619/diff/raw/ mv index.html r1619.diff.txt patch -p0 r1619.diff.txt Alec Hi Alec, that patch works fine for me in 1.8.9.0-rc3 too. Thanks a lot. Is there any chance that this would be included into the 1.8 branch? Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blocking in: ast_waitfor_nandfds
Hi I'm using asterisk 1.8.7.0 on a ContOS 5.6 machine Earlier today I spotted multiple occurrences of the following type of warning handle_response_invite: just did sched_add waitid(1955858) for sip_reinvite_retry for dialog etc... I recognised this as a hung channel, which I've not had since upgrading from 1.8.3.2 When I found the hung channel I noticed the following in the channel info Blocking in: ast_waitfor_nandfds Luckily this hung channel was able to be cleared by doing a channel request ahngup but I was wondering if anyone could shed some light on the issue Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Registration Hijacking
Can you please elaborate on rate limiting. Not how to implement it but rather how implementation is beneficiary. Reading up on it, it appears that it just checks the tcp connections and denys connection if limit is passed. In my thoughts, this is essentially a live fail2ban monitor in respects to attempted authentications. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim DeVito Sent: Saturday, January 21, 2012 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking Rate limiting (google) via iptables FTW! Good luck! - Original message - Alejandro Imass wrote 20.01.2012 18:09: I would like to know how to block this MF because he makes calls at 1-2 AM I use this construction on my servers [users] exten = _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1) [block] exten = _X.,1,HangUp(1) -- With Best Regards Mikhail Lischuk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Registration Hijacking
This is actually an interesting concept however I do think I want to restrict dialing during a specific time period. If someone is in the office, I would have to reprogram the route so allow dialing which adds overhead. Again, I do like the concept though. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk Sent: Friday, January 20, 2012 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking Alejandro Imass wrote 20.01.2012 18:09: I would like to know how to block this MF because he makes calls at 1-2 AM I use this construction on my servers [users] exten = _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1) [block] exten = _X.,1,HangUp(1) -- With Best Regards Mikhail Lischuk mailto:mlisc...@itx.com.ua -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Registration Hijacking
I appreciate your 2-cents worth. However, I do not believe they have access to machine If so, they are clever to create three failures in the logs for my benefit before entering the correct one for hijacking. Additionally, I have a lot of sip extensions to hijack and he keeps going for the same one. I was hoping this was something with the MP-118 and someone experienced the same thing with that device. Either way, I posed two questions which are still unanswered and probably I will never get answered: 1 - is this a vulnerability in the MP-118 2 - what method could they possibly be using to hijack a number-alpha extension which is creative to begin with ie) 203-Joes_Insurance_Service with an openssl generated password of 12 characters. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Saturday, January 21, 2012 1:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking On 20/01/2012 9:36 AM, eherr wrote: I have a honey pot box with extensions that are not just numbers ie ) 100-MySipUserName And the passwords are from an openssl generated password ie) Gq5VNIjDFWIQoUT6 Is the password stored in sip.conf in plain text or as an MD5? If it is stored in plain text then it may suggest the hijacker has greater access to your system than you realise. My 2-cents worth. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy : how to know channel name ?
Read the documentation. If I do Set(a=${CHANNELS()}) a will return a list
of channels with a space between each entry like you originally said. I did
my test using Set(a=${CHANNELS(1107)}) and it returned a=SIP/1107, therefore
I logically assumed that ${CHANNELS(miq8)}) would return SIP/miq8. If you
PROVE me wrong so be it, but I'm gonna SHUT UP now since you don't give a
rat's behind about my wasting time trying to help you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, January 25, 2012 3:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote:
Did a little research on this using my Asterisk 10.0. This should work for
you.
exten = 1246,1,answer()
exten = 1246,n,set(inuse=${CHANNELS(miq8)})
exten = 1246,n,extenspy(${inuse}@default)
exten = 1246,n,hangup()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
there is very little information about the function CHANNELS().
If I know the peer name (that I always know for sure), do you see a way of
using the function CHANNELS() to get the right channel ??
If CHANNELS() gives a space-delimited list of active channels, and I know
miq8... how can I get SIP/miq8-2419 ?
Thanks !
On 01/24/2012 04:46 PM, Danny Nicholas wrote:
Extenspy(miq8@default) for miq8. I would either proceed under the
assumption that I'm going to be listening to my extensions in the default
context or set up an AGI or something to load my needed ext@context
information.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
how to use ExtenSpy(extension@context) when conversations are named like
this ? :
SIP/378680644-2 default
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789
SIP/372083610-2 default
SIP/cedhou0-24 sub-uit789
SIP/travel3-2 pbx-routing
SIP/INTELin-2 pbx-routing
SIP/375382280-2 default
SIP/miq8-2419 sub-uitGSM
SIP/3749378004- default
SIP/instlpr0-2 sub-uitinternation
Can you tell me what is the extension ? How will I know the context ? The
context is not always the same...
On 01/24/2012 04:32 PM, Danny Nicholas wrote:
You are either going to be able to listen to SIP/miq8 or you are going to
have to know the sequence number like SIP/miq8-1. Maybe you should just
use ExtenSpy instead?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Of course I can control the name of my SIP-peer. Why do you tell me this ?!
Please answer my question : how do I know the channel name so I can ChanSpy
the correct channel ?
On 01/24/2012 04:13 PM, Danny Nicholas wrote:
It's not random. The Channel Name is Tech/peer-sequence (sequence is in
hex). You can control (to a degree) the peer portion in
sip.conf/users.conf.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
thanks. miq8 is the name of the SIP peer account.
So when I know the SIP peer name, and I strip of the numbers of the channel,
then I can use ChanSpy. So this answers my original question.
The only problem I see : it is Asterisk that gives the channel its name. How
do I change this ??
As far as I know, Asterisk randomly gives a channel name which consists of
the technology (SIP), the peername (miq8) and some numbers...
How to change the channel name ?
On 01/24/2012 03:53 PM, Danny Nicholas wrote:
I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List -
[asterisk-users] play sound file
Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Executing Script after MixMonitor is called
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is
called but it doesnt execute the command, I tried the command in terminal it
worked, any help please ... below is my dial plan
exten=6500,n,Set(MIXMONITOR_EXEC= nice -n 19 /usr/local/bin/lame -b 8 -t -F
-m m --bitwidth 8 --quiet /var/spool/asterisk/monitor/${CALLFILENAME}.wav
/var/spool/asterisk/monitor/${CALLFILENAME}.mp3 rm -f
/var/spool/asterisk/monitor/${CALLFILENAME}.wav)
exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Pickup calls coming from queues
Great to hear it's working for others. Regarding inclusion into 1.8 branch, as it's a new feature, it would only ever go into trunk, unless there is an outcry from the community. To assist others implementing this, and from a different viewpoint, would you mind documenting how you implemented it. I'm sure I've over complicated my examples. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Keuter Sent: Thursday, 26 January 2012 2:23 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pickup calls coming from queues Am 23.01.2012 um 23:25 schrieb Alec Davis: How can I test this solution on a 1.8.8.1 system ? If I'm not mistaken, diff https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1. I've just checked out 1.8.8.1 and download my patch from https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean, using the following on a debian lenny box: svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 asterisk-1.8.8.1 cd asterisk-1.8.8.1 wget --no-check-certificate https://reviewboard.asterisk.org/r/1619/diff/raw/ mv index.html r1619.diff.txt patch -p0 r1619.diff.txt Alec Hi Alec, that patch works fine for me in 1.8.9.0-rc3 too. Thanks a lot. Is there any chance that this would be included into the 1.8 branch? Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
Am 25.01.2012 um 20:24 schrieb Alec Davis: Great to hear it's working for others. Regarding inclusion into 1.8 branch, as it's a new feature, it would only ever go into trunk, unless there is an outcry from the community. Outcry! :-) To assist others implementing this, and from a different viewpoint, would you mind documenting how you implemented it. Sure, I did it slightly different than you, because in your way the only Hint worked fine for me, but Pickup not (I guess because of my 'notifycid=ignore-context' in sip.conf. I put all into my [test] context, but the hint in my standard hint context [blf] and used PICKUPMARK: ;; Queue Pickup with Hint :;exten = 8501,hint,Queue:itg_queue ;Provide a hint for the queue = in [blf] exten = _*98501,1,Pickup(itg@blf) ;Pickup the queue exten = 8501,1,Set(__PICKUPMARK=8501) exten = 8501,n,Queue(itg_queue,crhH,,,127) ;Ring the queue [blf] exten = 8501,hint,Queue:itg_queue I'm sure I've over complicated my examples. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Keuter Sent: Thursday, 26 January 2012 2:23 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pickup calls coming from queues Am 23.01.2012 um 23:25 schrieb Alec Davis: How can I test this solution on a 1.8.8.1 system ? If I'm not mistaken, diff https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1. I've just checked out 1.8.8.1 and download my patch from https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean, using the following on a debian lenny box: svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 asterisk-1.8.8.1 cd asterisk-1.8.8.1 wget --no-check-certificate https://reviewboard.asterisk.org/r/1619/diff/raw/ mv index.html r1619.diff.txt patch -p0 r1619.diff.txt Alec Hi Alec, that patch works fine for me in 1.8.9.0-rc3 too. Thanks a lot. Is there any chance that this would be included into the 1.8 branch? Michael Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping incompatible voice frame error
Greetings, We have an old analog phone system running Asterisk 1.2.13 (not my choice lol). Everything has been working wonderfully until today. The site is experiencing dropped and missed calls. When I tried calling the site, I did get through however the CLI was flooded with hundreds of copies of the following: Jan 25 15:46:54 NOTICE[2343]: channel.c:1956 ast_read: Dropping incompatible voice frame on Local/4227@from-sip-a3d8,2 of format ulaw since our native format has changed to slin I tried Googling the error but unfortunately, there were lots of reports of the problem and not a single solution or fix. My thinking is that it has to do with the service provider but I want to do my homework before pointing the finger. Any assistance would be greatly appreciated. Thanks! Kevin Oravits Phone Sys Admin/Tech Admin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
