You can use the voipmonitor sniffer.
www.voipmonitor.org.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Thu, 6 Dec 2018 at 00:13, Steve Edwards wrote:
>
> On Wed, 5
On Wed, 5 Dec 2018, Saint Michael wrote:
Is there a way to configure the old SIP channel to stay in sip set debug
all the time, without human intervention and also at boot time, by
default?
If your goal is capture all SIP traffic, there may be other tools better
suited.
For example, tcpdum
sipdebug = yes
sip.conf
---
I'm SoCIaL, MayBe
El 05/12/2018 a las 17:11, Saint Michael escribió:
Is there a way to configure the old SIP channel to stay in sip set
debug all the time, without human intervention and also at boot time,
by default?
--
_
Is there a way to configure the old SIP channel to stay in sip set debug
all the time, without human intervention and also at boot time, by default?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check
I certainly agree that the first and best solution is to deal with the
hardware issues, and we've started working on that already.
I'll investigate the suggested Asterisk ideas and post here if anything
works for my purposes.
Mitch
On 11/08/2013 12:13 AM, Mikhail Lischuk wrote:
Mitch Clabo
to levitate several feet in the
> air. I imagine it would have a similar effect on a call center agent.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
> Sent: Thursday, Nove
Mitch Claborn писал 08.11.2013 02:51:
> Is it possible to
catch the fact that an IP phone has died in the middle
> of a call and
do something with it in the dialplan?
Maybe you can connect agents and
callers via MeetMe, and when AMI gets the "MeetMe Leave" event, put the
caller on hold and r
On 13-11-07 07:51 PM, Mitch Claborn wrote:
Asterisk 11.1
Is it possible to catch the fact that an IP phone has died in the middle
of a call and do something with it in the dialplan?
Background: we run a small call center. Our agents sit in two groups,
with their IP phones running from 2 differ
enter agent.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Thursday, November 07, 2013 7:51 PM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Capture dead phone?
Asterisk 11.1
Is it po
Asterisk 11.1
Is it possible to catch the fact that an IP phone has died in the middle
of a call and do something with it in the dialplan?
Background: we run a small call center. Our agents sit in two groups,
with their IP phones running from 2 different switches. Every once in a
while the
2013-10-14 20:05, CDR skrev:
Right now,there is no way know to capture the Media IP.
I've seen serval suggestions for you on the list. I suggest you go back
and read them again.
Gareth Blades even handed you a solution to get sip-traces with all the
signaling. That's a good solution. (I thi
Right now,there is no way know to capture the Media IP. The channel
variable does not know about it. It requires adding anew variable to
CHANNEL(), and also it entails to force every channel to update that
variable. New channels like PJSIP do not even update the known
variables like CHANNEL(recvip)
+1000 to Matt point.
Many many talented developers who can assist in customizing your needs.
On Mon, 2013-10-14 at 19:55 +0530, Mitul Limbani wrote:
> Nailed it to the point Matt +1 on.this entire philosophy of open
> source.
>
> Mitul
>
>
> On Oct 14, 2013 7:19 PM, "Matthew Jordan" wrote:
>
Nailed it to the point Matt +1 on.this entire philosophy of open source.
Mitul
On Oct 14, 2013 7:19 PM, "Matthew Jordan" wrote:
>
>
>
> On Sun, Oct 13, 2013 at 2:06 PM, CDR wrote:
>
>
>
>
>> I need Digium to store this IP in the CDR. I will be honest with the
>> government and let them know th
On Sun, Oct 13, 2013 at 2:06 PM, CDR wrote:
> I need Digium to store this IP in the CDR. I will be honest with the
> government and let them know that my tool is incapable of saving lives
> or safeguarding our national security because nobody thought about
> this.
> PD: I am not paying for a p
On 13/10/13 20:06, CDR wrote:
I am quite surprised about the degree of surprise in the group. A few
days ago, somebody called a school and issued a threat, through my
network. The call came from China, but of course it was US caller. The
DA wants to know where call came from. The caller ID is "Re
On 13-10-13 03:06 PM, CDR wrote:
I am quite surprised about the degree of surprise in the group. A few
days ago, somebody called a school and issued a threat, through my
network. The call came from China, but of course it was US caller. The
DA wants to know where call came from. The caller ID is
Hi,
I also doubt that the IP would do any good, anyway you store whatever you
want in your cdr, just Set(CDR(something)=${SIP_HEADER(Contact)}); and then
have the field something in your cdr storage
On 13 October 2013 21:25, jg wrote:
> I doubt that a media IP would really help, because there
I doubt that a media IP would really help, because there are proxies out there. If you need this
kind of monitoring, then there are probably better ways to take care of this and they are
independent of Asterisk.
What you could do is to tap any traffic in the background, e.g. with tcpdump using
I am quite surprised about the degree of surprise in the group. A few
days ago, somebody called a school and issued a threat, through my
network. The call came from China, but of course it was US caller. The
DA wants to know where call came from. The caller ID is "Restricted"
and the chinese carrie
On 10/11/2013 10:05 PM, CDR wrote:
I am not proxying the media, but never the less I am forced to store
the source media IP in my CDR, for regulatory reasons. Asterisk gets
that information when the reinvite comes, but how do I store it?
If I don't figure this out my next email will be from Feder
On Sat, Oct 12, 2013 at 2:19 PM, CDR wrote:
> The CHANNEL() function has no idea about the media IP, and also
> SIP_HEADER(), since the media IP is not known until the call has been
> established and a reinvite has been received and dispatched. I am
> using of course, directmedia=yes and directrt
The CHANNEL() function has no idea about the media IP, and also
SIP_HEADER(), since the media IP is not known until the call has been
established and a reinvite has been received and dispatched. I am
using of course, directmedia=yes and directrtpsetup=yes. Hence my
question to the group.
--
_
hi,
you have not mentioned which cdr backend you are using.
peer ip is saved in variable CHANNEL(peerip).
if you are using mysql for cdr backend you can create a field in cdr table
(field name can b any of your choice)
in dialplan assign the value of CHANNEL(peerip) to you ip field and
asterisk wil
On Fri, Oct 11, 2013 at 9:05 PM, CDR wrote:
> I am not proxying the media, but never the less I am forced to store
> the source media IP in my CDR, for regulatory reasons. Asterisk gets
> that information when the reinvite comes, but how do I store it?
> If I don't figure this out my next email w
I am not proxying the media, but never the less I am forced to store
the source media IP in my CDR, for regulatory reasons. Asterisk gets
that information when the reinvite comes, but how do I store it?
If I don't figure this out my next email will be from Federal Prison.
Kindly help me stay away f
2013/1/21 Mitch Claborn
> Asterisk 11
>
> Occasionally we will have a partial power outage, or a piece of network
> equipment will fail, and our queue agents who are on active calls with
> callers will be disconnected from the caller. What I'd like to do is
> capture those calls and put them bac
Un-topposted
Eric Wieling writes:
> Using qualify=10 ?
qualifyfreq=10 is fine, but Asterisk will not AFAIK do anything to a
call just because the peer goes unreachable qualify-wise. You are still
stuck with running a script that listens to qualify-unreachables and
does the appropriate thing to
Non-Commercial Discussion'
Cc: 'Benny Amorsen'
Subject: RE: [asterisk-users] Capture queue agent drop and put caller back in
queue
A qualify value that low would be a resource hog (some phones can't even
re-register in 10 seconds). The Nagios solution would require a custom
Sent: Tuesday, January 22, 2013 4:12 PM
> To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial
> Discussion; Danny Nicholas
> Cc: Benny Amorsen
> Subject: RE: [asterisk-users] Capture queue agent drop and put caller back
> in queue
>
> Using qualify=10 ?
>
ewiel...@nyigc.com]
Sent: Tuesday, January 22, 2013 4:12 PM
To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion;
Danny Nicholas
Cc: Benny Amorsen
Subject: RE: [asterisk-users] Capture queue agent drop and put caller back in
queue
Using qualify=10 ?
-Original Message-
ailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
> Harrington
> Sent: Tuesday, January 22, 2013 5:11 PM
> To: Danny Nicholas
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Benny Amorsen
> Subject: Re: [asterisk-users] Capture queue agent
; Benny Amorsen
Subject: Re: [asterisk-users] Capture queue agent drop and put caller back in
queue
How do you propose that Asterisk determines that the endpoint has vanished off
the network without waiting for a 10-90 second timeout period?
On Tue, Jan 22, 2013 at 4:07 PM, Danny Nicholas
How do you propose that Asterisk determines that the endpoint has vanished
off the network without waiting for a 10-90 second timeout period?
On Tue, Jan 22, 2013 at 4:07 PM, Danny Nicholas wrote:
> Not doubting how quickly Nagios can respond, but if the Nagios solution is
> going to place a ca
Christopher
Harrington
Sent: Tuesday, January 22, 2013 3:30 PM
To: Benny Amorsen
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capture queue agent drop and put caller back in
queue
On Tue, Jan 22, 2013 at 3:01 PM, Benny Amorsen wrote:
Can a
On Tue, Jan 22, 2013 at 3:01 PM, Benny Amorsen wrote:
>
> Can a Nagios-based solution provide quicker failover than the 90 seconds
> provided by sip timers or the 10-30 seconds provided by rtptimeout?
>
Certainly; Nagios can detect missed ping responses with a granularity of
single seconds.
--
Christopher Harrington writes:
> Since nobody seems to have come up with an Asterisk-specific solution, it
> sounds like the real approach here is something more generic.
> You can set up Nagios to fire off an event if it detects endpoints or
> infrastructure are suddenly dead. In particular, Nag
On Mon, Jan 21, 2013 at 12:03 PM, Mitch Claborn wrote:
> How can I accomplish my goal?
Since nobody seems to have come up with an Asterisk-specific solution, it
sounds like the real approach here is something more generic.
You can set up Nagios to fire off an event if it detects endpoints or
in
Mitch Claborn writes:
> Shouldn't asterisk somehow know when the agent disappears?
You are a bit out of luck since SIP session timers, the obvious
solution, cannot be set lower than 90 seconds.
rtptimeout set to e.g. 10 seconds may work, but you need to then set
rtpholdtimeout higher and hope t
On Mon, Jan 21, 2013 at 12:03 PM, Mitch Claborn wrote:
> Shouldn't asterisk somehow know when the agent disappears?
>
Asterisk will only notice that the agent is gone when a timeout has
occurred. When you're pulling out the Ethernet cable, there's no
opportunity for any equipment to signal to Aste
On Mon, Jan 21, 2013 at 11:03 AM, Mitch Claborn wrote:
> How can I accomplish my goal?
>
http://lmgtfy.com/?q=battery+backup
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- Bandwidth and Colocation Provided by http://www.api-
Asterisk 11
Occasionally we will have a partial power outage, or a piece of network
equipment will fail, and our queue agents who are on active calls with
callers will be disconnected from the caller. What I'd like to do is
capture those calls and put them back in the queue (at a high priorit
HI,
*Asterisk 1.8 *allows to read SIP response codes in the dialplan with *
${HASH(SIP_CAUSE,)}* - This is working fine for me too.
This is the dialplan line just after the DIAL()
same => n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)}):4:3})
Ref: http://www.voip-info.org/wiki/view
Hello,
I am using a mix of Call files and AMI telnet from a perl app to place
calls. I sometimes get this in the CLI:
-- Attempting call on sip/551234@for 1@:1
(Retry 1)
[Feb 27 13:47:07] == Using SIP RTP CoS mark 5
[Feb 27 13:47:07] -- Got SIP response 503 "No Circuit Available"
ohh Didnt notice !!
On Mon, Feb 15, 2010 at 6:55 AM, Global Meds wrote:
>
> How to capture keys entered through soft phone and what action asterisk is
> taking on that key ?
>
> Any Log of that ?
>
>
--
_
-- Bandwidth and Coloc
On Mon, 2010-02-15 at 06:55 +0530, Global Meds wrote:
[cut]
FYI: You may not be aware but with your 'Global Meds' name you are going
to be ending up in lots of spam filters. I've just had to fish your
posts out of a Quarantine.
>
--
__
How to capture keys entered through soft phone and what action asterisk is
taking on that key ?
Any Log of that ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCR
It is fairly trivial to modify chan_sip to expose headers from final
replies to the SIP_HEADER container or some other channel variables.
Just make sure to disambiguate names of headers if you put them in a
conflicting namespace.
There is one big switch statement that dispatches handling b
There is, actually, a Server header. It is the equivalent of User-
Agent for UASs.
--
Sent from mobile device
On May 17, 2009, at 9:19 AM, David Backeberg
wrote:
> On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski
> wrote:
>> Unfortunately SIP_HEADER(FROM) is not an option for me.
>>
>>
"User-Agent" header is present in SIP *request* i.e. "INVITE" received
by Asterisk from UAC.
RFC 3261 - 20.41 User-Agent
The User-Agent header field contains information about the UAC
originating the request. The semantics of this header field are
defined in [H14.43].
Revealing the
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski wrote:
> Unfortunately SIP_HEADER(FROM) is not an option for me.
>
> What I want to do is record in CDRs "User-Agent" header of calling
> party (this can be easily done with ${CHANNEL(useragent)}), and SIP
> "Server" header of called party (from 2
Hi David,
Thanks for your post.
Unfortunately SIP_HEADER(FROM) is not an option for me.
What I want to do is record in CDRs "User-Agent" header of calling
party (this can be easily done with ${CHANNEL(useragent)}), and SIP
"Server" header of called party (from 200 OK response to INVITE
generated
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski wrote:
> I am trying to capture "Server" header in a 200 OK reply message.
> My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
> and inside of GetOtherPartyInfo macro use SIP_HEADER function.
> unfortunately the above doesn't seem
Hi,
I am trying to capture "Server" header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
For example:
[default]
exten => _X.,1,Dial(SIP/u...@domain,30,M(GetOtherPartyInfo))
exten => _X.,
On Wednesday 06 August 2008 22:35:01 Positively Optimistic wrote:
> We've searched but thus far have not successfully found a solution for
> this…
>
> We're looking for a way to set a variable using get digits for a DISA
> application. Sometimes we're away from the office and get a voicemail
> th
We've searched but thus far have not successfully found a solution for this…
We're looking for a way to set a variable using get digits for a DISA
application. Sometimes we're away from the office and get a voicemail that
I need to respond to quickly and would prefer for the caller to be present
I would like extensions within a specific group to capture calls (with the
typical *8).
All's fine if the extensions are registered to the same asterisk server.
However, I don't know how to make it work if they register to different
Asterisk servers within the same LAN.
I use DUNDi to call fro
EMAIL PROTECTED] On Behalf Of Lee
NorvallSent: Saturday, June 12, 2004 9:30 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Capture
user input
Hi
I just wanted to
know if anyone has done the following, or knows how to.
When a customer
dials into *, we would then ask them t
Title: Message
Hi
I just wanted to
know if anyone has done the following, or knows how to.
When a customer
dials into *, we would then ask them to for an account number (which they would
type in with the key pad), we would then ask them to select and option (1 to
9).
We would then like
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