utes, and quite possibly is some big number someone
thought to set in something that "no one would ever hit".
A tcpdump would probably show you what's going on if the logs are otherwise
unclear, and you could also make sure you have sensible RTP timeout rules.
Andrew
--
___
nes into it using one fo the pipeline parsers; it' sjust a bit messy
config wise when using the stnadard docker containers. But if it's not a
"simple" answer it sounds like that's not a bad option.
Andrew
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___
o do this.
Ast 16 at the moment…
Andrew
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New to Asterisk
Did you check your security log?
There is usually a wealth of info there about who, what, where when and why.
Andrew
On Wed, 22 Jul 2020 at 11:22 pm, Jerry Geis wrote:
> >exten = i,1,Verbose(Incoming ANONYMOUS SIP call from ${CALLERID(name)}
> >${CALLERID(num)} SRC IP ${CH
, because it makes X or Y amazing”.
Andrew
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*business nbn™ advisor (advisor 01783150)*
Real World Technology Solutions - IT People you can trust
Voice | Data | IT Procurement | Managed IT
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*Real World is a DellEMC
16.11.1 was
pulled from the current release directory.
I realise it's probably better practice to pull source tarballs from
old-releases, but is there a reason that the current "released" version was
pulled?
Andrew
--
___
c?
Cheers,
Andrew
--
Andrew Ruthven, Wellington, New Zealand
MIITP
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Cloud : https://catalystcloud.nz
GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
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- Original Message -
> From: "John Novack SCII_U"
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
> , "Andrew Martin"
>
> Sent: Monday, October 8, 2018 4:29:41 PM
> Subject: Re: [asterisk-users] Dropped calls when all DAH
d?
Thanks,
Andrew
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Check out the ne
Ditto; a Gmail issue?
Andrew
On 12 June 2017 at 16:00, Marcelo Terres wrote:
> It is happening the same with me.
>
> Regards,
> Marcelo H. Terres
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
>
ded to thwart the fraud. A simple RFID pad
setup could be built to use low usage GSM plan to tag in the RFID on site.
But this is beyond the scope of telephony.
--
- Andrew "lathama" Latham -
--
_
--
System(pkill tcpdump);
> Hangup;
>
> Or whitout RTP:
>
> System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060
> &);
> Wait(1);
> Dial(SIP/${EXTEN});
> System(pkill tcpdump);
> Ha
in and out of band DTMF
> problems that we were having with various carriers.
>
> Tim
>
> On 2/17/17 3:07 PM, Derek Andrew wrote:
> > The SIP trace will be adequate but this is on a remote system with
> > limited disk space.
> >
> > I would love to turn on deb
> asterisk> dialplan show xxx@your_context
>
> Where xxx is the number you want to dial, from the context asigned to your
> extension.
>
> Cheers
>
>
> El 17/2/2017 19:44, "Derek Andrew" escribió:
>
>> I have some troublesome numbers that I would like to
, Feb 17, 2017 at 4:56 PM, Tim Pozar wrote:
> Why not capture the packets with something like tcpdump and run it
> through Wireshark?
>
> Tim
>
> On 2/17/17 2:43 PM, Derek Andrew wrote:
> > I have some troublesome numbers that I would like to capture the SIP
> > d
I have some troublesome numbers that I would like to capture the SIP
dialogue when I am calling them. When I am about to dial the number, is
there any way to turn on SIP debugging in the dial plan before I make the
call? (and turn it off after the call is completed?)
--
___
ut the "addcaller" stuff here:
https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration
Essentially you'd have a dialplan where you can call another number
which is then added to the confbridge.
Cheers,
Andrew
--
Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ
At
Thanks for confirming that Joshua. Thought that might be the case. I'll
look at a workaround.
Andrew
On 1 October 2016 at 20:20, Joshua Colp wrote:
> Andrew Ivins wrote:
>
>> Hi list.
>>
>> I use sorcery to configure an astdb backend to my pjsip endpoints.
>&g
them are effectively defaults. If I need to change one, I need to update
each endpoint in the astdb.
Any suggestions?
Andrew
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You won't see anything in the Asterisk logs because there's nothing to log.
The error happens in the freeradius-client library and returns an integer.
On 29 Sep 2016 17:44, "Willem Offermans" wrote:
Hello Andrew and asterisk friends,
I suspect that asterisk has probl
You don't get anything in the Asterisk logs because freeradius-client
(formerly radiusclient-ng) returns a single failure code for any failure
when building a radius request.
Andrew
On 29 September 2016 at 15:55, Willy Offermans wrote:
> Hi Ahmed and asterisk friends,
>
> So a
instances where it returns ERROR_RC. You are almost
certainly running into one of these. I ended up putting in print debug into
that file and recompiling. I think in my case it was as simple as a
hostname not resolving. Once you're not working blind, you'll find what is
happening pretty
at
behaviour, I found the only solution was to tell the Dial() command to
ignore the 302 by adding the i flag. Problem solved.
Cheers,
Andrew
--
Andrew Ruthven, Wellington, New Zealand
MIITP, ITCP
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Cloud : NZs only real cloud -
I had a similar issue and i set a timeout which fixed the issue
SIP/trunk/ ${EXTEN},216,t
We only had this on one of our providers the rest we havent had the issue
- Original Message -
From: Steve Edwards
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sat, 30 J
il(box@context,option)
> same => n,Hangup
>
> ; Andrew Ruthven
> exten => 7231,1,Set(CALLERID(number)=yyy)
> same => n,Goto(pstn,xxx,1)
>
> On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven yst.net.nz> wrote:
> > Hey,
> >
> > I have fre
o far I've done this by duplicating the standard extension macro, and
adding this rule (where ARG1 is the extension):
exten => a,1,Goto(vmfwd,${ARG1},1)
Then in the vmfwd context I have rules like this (I need to set the
CALLERID(number) so our SIP provider accepts the call):
; Andrew Ru
Thanks Joshua. That did the trick.
On 4 July 2016 at 19:18, Joshua Colp wrote:
> Andrew Ivins wrote:
>
>> On 1 July 2016 at 17:41, Joshua Colp > <mailto:jc...@digium.com>> wrote:
>>
>>
>> exten => 1234,Set(CALLERID(all)="Jon Doe"
On 1 July 2016 at 17:41, Joshua Colp wrote:
>
>
>> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
>> same => n,Dial(PJSIP/phone123, 30)
>>
>
> Your exten line has no priority, is that how it is in your dialplan?
>
Actually no, I stole that line from an earlier email to this list. Mine ha
Asterisk 13.8
Is CALLERID(all) supposed to wok for pjsip? When I do this:
exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)
I expect the callerid to be as set, but is always seems to be "phone123",
t
Hi,
Asterisk 13.8.0. Can anybody explain why I get two objects whenever I use
ARI Push? (https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration
)
Below is what I see using the auth push example from the wiki, but I get
the same thing for endpoints and aors too.
root@vagrant-ubuntu-wil
rpose of s3fs-like addons (see [1]) to let S3 buckets be
> mounted on Linux and thus allow any application like Asterisk make
> use of it ?
>
> [1] https://github.com/s3fs-fuse/s3fs-fuse
>
> 2016-02-16 1:05 GMT+01:00 Andrew Ruthven .nz>:
> > Hey,
> >
> >
Hey,
I've found a bit of chatter about people using hacks to copy voicemail
messages into object storage (like S3) after they've been recorded. But
I was wondering if any work has been done on the VoiceMail app to
actually store and retrieve messages to/from an object store?
Chee
Hi Guys
I am seeing this error a lot in the CLI lately
What does it mean?
Prodding channel SIP/XXX failed
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Ok thanks Joshua
Do you know what this error means when I dial out in pjsip and the call
fails
Unable to create request with auth.No auth credent als for any realms in
challenge
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258
Do you know if this can be achieved with the standard sip stack in asterisk?
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net
Web: http
Colp Date:
2015/10/19 13:03 (GMT+02:00) To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Modify Contact in PJsip
On 15-10-19 07:41 AM, Andrew Colin wrote:
> Hi Guys
>
> We are using the wizard to configure our pjsip trunk(see below)
>
> How do we get this
Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the
username of the trunk
[trunk_defaults](!)
type = wizard
transport = transport-udp
en
Hi Guys
I keep getting this "Warning" when I dial out via pjsip and the calls fail
But if I do a pjsip reload it works for 1 minute
WARNING[6707]: res_pjsip_outbound_authenticator_digest.c:135
digest_create_request_with_auth_from_old: Unable to create request with
auth.No auth credentials
You can use this
exten => h,1,Set(CDR(userfield)=Hangupcause:${HANGUPCAUSE})
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ross Beer
Sent: Friday, October 9, 2015 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi Guys
Does anyone know of a way I can change the contact field in the sip invite
to display sip:username:ip instead of sip:did:ip
We need to do this without changing the from field.
I tried using fromuser=username but that just modifies both the contact and
the from parameter
I know in
pouts, or what I should
look at next for additional debug information?
Thanks,
Andrew Martin
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> An unknown state would be a device that has a valid configuration but isn't
> registered.
>
John,
Thanks for the clarification and your help resolving this issue!
Andrew
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mpty=unavailable,invalid,unknown
timeout=18
member => SIP/100
member => SIP/101
Is there any reason that using any of these options would be a
problem, in particular "unknown"? It is not very well defined
what an "unknown" state is exactly.
Thanks,
Andrew
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alplan never execute. How can I correct this behavior
so that even if the queue has no registered members, the dialplan is still
followed?
Thanks,
Andrew
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Hi Aj
Can you perhaps show me an example as to how you would do it as I have tried
setting it very early but still doesn’t work
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls
Hi Helvio
I will be interested to test your product and give you some feedback. .
Sent from my Samsung Galaxy s6 smartphone.
Original message
From: Helvio Junior
Date: 29/06/2015 20:58 (GMT+02:00)
To: Abdul Basit , Asterisk Users Mailing List -
Non-Commercial Discussion
king process is modifying another part of the packet too?
Both devices are on the same subnet, so although these switches do route
traffic as well, that shouldn't be coming into play here.
Andrew
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- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, May 13, 2015 10:50:02 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, May 13, 2015 10:10:25 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, May 12, 2015 5:42:57 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Andrew Martin"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 4:18:58 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Andrew Martin"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 1:35:07 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
se are directly route-able so no
NAT is involved). However, I have now been able to reproduce the problem between
two devices directly on the 10.10.32.0/21 network as well.
Thanks,
Andrew
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- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 12:32:06 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Andrew Martin"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, May 8, 2015 5:12:28 PM
> Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls
>
;Retransmission timeout" problem?
Here is my sip.conf:
general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0
[123]
secret=11
host=dynamic
type=friend
Th
different list of phones.
Thanks,
Andrew
- Original Message -
> From: "James Thomas"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 7, 2015 10:20:10 AM
> Subject: Re: [asterisk-users] Phones don't stop
P/265-2931'
This only happens occassionally; most of the time the phones will all
immediately stop ringing once one of them picks up. Do you have any ideas about
what could be wrong here or what else I can do to debug?
Thanks,
Andrew Martin
--
__
Hi Guys
We have a strange issue whereby one phone has delayed rtp
So what happens is when the lady answers the phone for the 1st 1 second
they can not hear her and then everything is fine
I am running asterisk 1.8.28.0
Has anyone seen this before?
--
_
percentage of loss for received packets for 192.168.32.26
> > seems suspicious. Do these statistics indicate a problem?
> >
> > Thanks,
> >
> > Andrew
>
> Hi Andrew,
>
> is this a linux machine? If so, check your NIC with ifconfig for
> hardware errors.
>
- Original Message -
> From: "Administrator TOOTAI"
> To: asterisk-users@lists.digium.com
> Sent: Friday, May 1, 2015 6:42:38 AM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
>
> Le 01/05/2015 00:05, Andrew Martin a écrit
onservative.
Daniel,
Thanks for the information. Do you have an example or documentation on the
siproxd configuration that you used?
Thanks,
Andrew
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New to Asteri
ebin.com/uZSMKczk
What else can I try to debug these problems? Since it is intermittent, I am not
always able to reproduce (sometimes the calls work just fine).
Thanks,
Andrew Martin
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Hi Guys
Is it possible to register Kamallio directly to our SIP provider then load
balance the RTP to 2 asterisk servers?
We cant do the registration from the asterisk boxes as we want to do it
directly from Kamallio.
Is this possible?
--
___
On Thu, Apr 9, 2015 at 12:23 PM, Derek Andrew wrote:
> SNOM phones can be configured using files on a TFTP server.
>
> On Thu, Apr 9, 2015 at 11:14 AM, jg wrote:
>
>>
>> Does anyone know how to program Snom phones using a Mac addresses like
>> what is done wi
one of the provisioning methods.
>
> jg
>
--
Copyright 2015 Derek Andrew (excluding quotations)
+1 306 966 4808
University of Saskatchewan
Peterson 120; 54 Innovation Boulevard
Saskatoon,Saskatchewan,Canada. S7N 2V3
Timezone GMT-6
Typed but not read.
--
_
; to
department2
exten => s,19,GotoIf($["${callersname}" = "Company7"]?internal,36,1:20); to
reception
exten => s,20,GotoIf($["${callersname}" = "Company8"]?internal,13,1:21); to
department3
And later in same file:
; Phone 36 reception
> *exten =
en => s,1,Set(thedid="${SIP_HEADER(TO)}")
> exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
> exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
> exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
> ; Direct the DID accordingly.
> exten => s,5,GotoIf($["${p
orrect handset as the
business needs, but it seems that all incoming calls are being labeled as
though coming in on a different account. The effective problem is that the
calledID is now wrong.
I'm after some general advice on how to handle the problem.
Ta,
-Andrew
--
4 Port PRI sangoma a104
From: jg [mailto:webaccounts...@jgoettgens.de]
Sent: Wednesday, March 18, 2015 2:09 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 4 Port PRI
I have a 4 port PRI card that I need to setup each port in their
Hi Guys
I have a 4 port PRI card that I need to setup each port in their own
group.
In chan_dahdi.conf I have the following which works for one port
How do I add the rest of the ports in their own groups so that I can have
different signaling on each?
[channels]
language=en
switchtyp
Originally we used just POE but now each of the 3 panels has its own PSU
From: jg [mailto:webaccounts...@jgoettgens.de]
Sent: Friday, March 13, 2015 11:18 AM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Yealink t26 and T28 Panels
Hi Guys
We have a strange a strange issue at a client they have 3 panels on their
phone and every so often the panels reboot themselves yet the phone stays
on.
We decided to replace the T26 for a T28 to see if it fixes the issue and
still have the exact same issue.
Has anyone seen this bef
Hi Guys,
We are getting a strange issue on certain polycom phones, sometimes when a
call comes in it just "flashes" to show there is a call but does not play
any sound.
This problem is very intermittent and happens to maybe 2 out of 10 calls.
Has any else experienced this before?
--
_
RFC2833
The strange thing is how asterisk is not registering she has pushed ## on
those "Rare" occiasions"
> On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin wrote:
>
>> The strange thing is its only sometimes my dial string is as follows
>>
>> exten =&g
age From: Kevin Larsen
Date:16/02/2015 17:11
(GMT+02:00) To: Andrew Colin ,Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
BlindXfer Sensitivity
> Hi Guys
>
> We have a client running on a polycom vvx400 IP phone on our
> asteris
IONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, PRACK, MESSAGE
Supported: replaces, timer, 100rel
Content-Length: 0
--- CUT --
TIA
--
Andrew McRory
Sayso Communications, Inc.
Tallahassee, Florida
--
__
Hi Guys
We have a client running on a polycom vvx400 IP phone on our asterisk
1.8.18 system
The issue we have is the switchboard lady uses ## to transfer calls but
sometimes it just does not work and just plays the DTMF tone to the
calling party.
Is there any way to adjust the sensitivi
Hi
queue reload(queue name) or queue reload all
for example
queue reload reception
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, January 8, 2015 2:10 PM
To: Asterisk Users Mailing List - Non-Comm
I currently am running on a
Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz
Codec im using is
codec_g729-ast18-icc-glibc-x86_64-core2.so
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: +27 (0)10 591 4607
I currently am running on a
Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz
Codec im using is
codec_g729-ast18-icc-glibc-x86_64-core2.so
--
_
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New to Ast
I am using the free g729
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: +27 (0)10 591 4607
Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net
Web: http
Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: +27 (0)10 591 4607
Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: <mailto:and...@convergedgroup.net> and...@convergedgroup.net
Web
Hi Rainer,
I am using roundrobin
From: Rainer Piper [mailto:rainer.pi...@soho-piper.de]
Sent: Thursday, September 25, 2014 6:21 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime ERROR
Am 25.09.2014 um 16:24 schrieb
Hi Guys,
I have recently moved my database servers to a different database cluster
that runs on haproxy.
Every minute or so I get the following error in asterisk
MySQL RealTime: Ping failed (2006). Trying an explicit reconnect
The strange thing is if I do realtime mysql status
It sho
;http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Copyright 2014 Derek Andrew (excluding quotations)
+1 306 966 4808
Information and Communications Technology
University of Saskatchewan
Peterson 120; 54 Innovation B
7; in context 'demo3'
> -- Executing [h@demo3:1] NoOp("SIP/101-000d", "terminating call")
> in new stack
> [Sep 9 02:11:14] NOTICE[9984]: pbx_spool.c:402 attempt_thread: Call
> completed to SIP/101/009871888729
>
> Anurag Rana
> h
> 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
>
--
Copyright 2014 Derek Andrew
Can you explain?
Sent from Samsung Mobile
Original message From: Tiago Geada
Date:03/07/2014 9:04 PM (GMT+02:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
recording in mp3
no need.
mixmonitor has a argument that is a scr
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
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[asterisk-users] Call rating software
On Wednesday 02 Jul 2014, Andrew Colin wrote:
> Can you try maybe assist with this, as I have tried for ages and still cant
> get it right.
Firstly, have you got CDR working and writing to some sort of database? We
use cdr_mysql; although the more
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call rating software
On Tuesday 01 Jul 2014, andrew Colin wrote:
> Hi Guys
>
> Does anyone know of any good cdr rating software.
>
> I am looking for something that I can pull reports by extension.
> Not a full bil
Hi Guys
Does anyone know of any good cdr rating software.
I am looking for something that I can pull reports by extension.
Not a full billing solution like a2billing.
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interface?
On 1/7/2014 19:13, andrew Colin wrote:
Problem with this is client needs to listen to the call recordings and my
interface will only display .wav or .mp3 so they will moan if they
have to wait until the next day for today's recordings
Sent from Samsung M
at 3:11 PM, andrew Colin
wrote:
Hey guys
Is it possible to record with mixmonitor straight into mp3.
I am trying to reduce disk space and want my calls to be recorded in mp3
Instead of wav.
Sent from Samsung Mobile
Original message
From: Sameer Rathod
Date:30/06/20
Hey guys
Is it possible to record with mixmonitor straight into mp3.
I am trying to reduce disk space and want my calls to be recorded in mp3
Instead of wav.
Sent from Samsung Mobile
Original message From: Sameer Rathod
Date:30/06/2014 9:23 PM (GMT+02:00)
To: asterisk-
Block the ip?
You should only enable sip for your specific clients in iptables.
Sent from Samsung Mobile
Original message From: arun kumar
Date:27/06/2014 4:42 PM (GMT+02:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Attac
Does a reload (not a sip reload) reload everything or does it also require
the sip.conf file to be modified?
On Wed, Apr 30, 2014 at 5:00 AM, Administrator TOOTAI wrote:
> Le 30/04/2014 12:39, Administrator TOOTAI a écrit :
>
>> Le 30/04/2014 12:15, Administrator TOOTAI a écrit :
>>
>>> Hi,
>>>
Geoip works well to block all countries except your own
Regards
Andrew Colin-mobile
Vsave(PTY)Ltd
Original message
From: Eric Wieling
Date:19/01/2014 8:40 PM (GMT+02:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
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