no
any ideas? Monitoring using CLI, I noticed the device always select ulaw for
codec.
Thanks,
Motty
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Hello,
Best practices examples to upgrade Asterisk 13.13.1 to latest version?
Any suggestions?
Thanks,
Motty
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X,2,Dial(SIP/voip1/13781${EXTEN:1},80)
exten => _7XXX,n,Congestion()
exten => _7XXX,n,Hangup()
how would I change it? I have look in cdr.conf and logger.conf
Thanks,
From: Motty Cruz [mailto:motty.c...@gmail.com]
Sent: Monday, April 03, 2017 3:52 PM
To: 'Aster
Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number
omega*CLI> core show channels
Channel Location State Application(Data)
Message/ast_msg_queu 4002@sipphones:2 Up VoiceMail(4002@default,u)
"Message/ast_msg_queu" it's been up for the last day, how to hangup this
channel?
--
her MySQL or cdrlite and a quickie sql query? I could
> have added postgres, but I'm a DB bigot. That would work too.
>
>
> On 03/22/2017 01:46 PM, Motty Cruz wrote:
>
>>
>> Hello, I am looking for CDR reporting solution? Any suggestions? I am
>> using Asterisk 13.
Hello, I am looking for CDR reporting solution? Any suggestions? I am using
Asterisk 13.13.1
I would like a report on number of calls per extension.
Thanks,
Motty
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enticate user .*@.*
NOTICE.* .*: Sending fake auth rejection for device
.*\<sip:.*\@\>;tag=.*
NOTICE.* .*: Registration from '\".*\".*' failed for '' -
No matching peer found
NOTICE.* .*: Registration from '\".*\".*' failed for '' -
Wrong password
?
Thanks,
Motty
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https
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error:
-- SIP/voipeer-084b redirecting info has changed,
ytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1
>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
>>> Fresh installed CentOS 7.3 and Asterisk 13.13.1.
servers were similar in CPU, Memory
Any support on this matter is appreciated!
Thanks,
Motty
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kambiz sharifi
Sent: Saturday, January 28, 2017 5:13 AM
2b5ef77@192.168.0.191 for seqno 156 (Critical Request) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
any ideas?
Thanks!
Motty
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Asterisk àAdTran (total access 904 2nd gen)
Calls are being forward to AdTran from Asterisk!
Thanks,
Motty
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, December 06, 2016 3:49 PM
To: Asterisk Users
Thank you Carlos, you’re right I am using PJSIP. Should I not use it?
Thanks,
Motty
From: Carlos Chavez [mailto:cur...@telecomab.mx]
Sent: Saturday, December 31, 2016 5:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Motty Cruz
Subject: Re: [asterisk-users] how
ot;My ID" )
exten => _7XXX,n,Dial(SIP/mySIPprovider/1731${EXTEN:1},80)
This syntax does not work in Asterisk 13.13. has anybody dealt with this
issue?
Thanks
Motty
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yes
canreinvite=no
Asterisk server IP XX.XX.42.16 (Public)
Client IP: 192.168.1.56
I would look for typos in your configuration!
-Motty
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, December
="iuser 1005"
disallow=all
allow=ulaw
allow=alaw
username=1005
auth=md5
secret=819c8ebd2d1525235235325235
dtmfmode=rfc2833
host=dynamic
nat=yes
canreinvite=no
Thanks,
Motty
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Thanks for your support!
Motty
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New to Asterisk? Start here:
https
er.
Thanks,
Mottyh
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Sunday, November 06, 2016 12:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 to Asterisk 1
8008000>)
exten => _9XXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80)
Any ideas?
Thanks,
Motty
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arted
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Thanks for your support,
Motty
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n
Max-Forwards: 70
Expires: 90
Content-Length: 0
Sip.conf
[1006]
type=friend
username=1006
secret=mysecret
context=sip-phone
call-limit=5
callerid="iuser" <1006>
disallow=all
host=dynamic
allow=all
nat=yes
Is NAT value set to yes OK? Servers is
ll
any ideas?
Thanks,
Motty
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/communi
amic
mailbox=5007
nat=yes
canreinvite=no
transport=tls
Thanks for your help!
-Motty
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I finally secure SIP session between Asterisk server and a remote client. My
questions is the following; do I need to open port 5061 UDP on my firewall
or just port 5061 TCP for SIP sessions.?
I am not interested in securing RTP only SIP sessions.
Thanks for your help!
--
1 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8 secure SIP session only
Your CA cert is missing.
Add in sip.conf:
tlscafile=/etc/asterisk/keys/ca.crt
You don't need:
tlscapath=/etc/asterisk/keys
On 4 May 2016 at 19:43, Motty
Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I
keep getter an error,
== Problem setting up ssl connection: error:14094418:SSL
routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:
FILE * open
is
that possible to add dahdi/g1/6078880/g1/7068880 ?
--
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Motty
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sip.c: Failed to authenticate
device 100<sip:100@X.X.X.X>;tag=a121ab55
X.X.X.X is the IP of my Server, I don't know who is the attacker IP
unless I monitor for the server using the following command:
tcpdump -lni eth0 -f "udp port 5060"
Please advise.
Thanks,
Motty
On 12/02/2015 01:53
where this attack come from I use the following command
"tcpdump -lni eth0 -f "udp port 5060" is there an easy way to get the
attacker's IP?
Thanks,
Motty
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Thank you very much Dave,
_Motty
On 10/31/2015 10:47 AM, Dave Platt wrote:
Thanks Jeff, just to confirm, password are not sent in plain text? I
want to safeguard against man in the middle attacks, sniffing traffic of
clients.
That's correct.
The way it works is:
- Both the client, and
Hello,
I am trying to forward number, in the past I was able to use this:
;;; 201-704-4482
exten => 4695,1,Dial(dahdi/8/w73#w7044482)
exten => 4695,2,Congestion
exten => 4695,102,Congestion
is that correct way to forward? the phone is with AT company. On AT
site is says to use *73#7044482 ?
Thanks Jeff, just to confirm, password are not sent in plain text? I
want to safeguard against man in the middle attacks, sniffing traffic of
clients.
Thanks,
_motty
On 10/30/2015 07:37 AM, Jeff LaCoursiere wrote:
On 10/29/2015 04:01 PM, Motty wrote:
On 10/29/2015 01:11 PM, Jeff
Thanks Jeff,
I don't want SIP over TLS. I would like to encrypt password only, I
suppose over TLS.
Thanks,
_motty
On 10/29/2015 01:11 PM, Jeff LaCoursiere wrote:
On 10/28/2015 06:37 PM, Pete Mundy wrote:
Hi Motty,
Isn't the whole point of the nonce in a SIP registration to ensure
Hello,
I am searching for a solution to encrypt authentication from Asterisk
server to clients. Searching srtp seem to encrypt traffic, I just want
client authentication with encryption. Can someone point to the right
direction? has anybody used ZRTP? experience with ZRTP?
Thanks,
_motty
--
Hello,
I would like to encrypt password between Asterisk servers and clients.
is there an easy way to do so? I am running Asterisk 1.8.22.0 built on
CentOS 6.3
Thanks,
.Motty
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Thanks Oliver,
I am using this: http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret
however, I wanted to know if there is a way to use some sort of
Certificates.
http://www.voip-info.org/wiki/view/Asterisk+encryption
has anybody used this feature?
Thanks,
.Motty
On 07/29/2015 03:40 PM
Hello,
this worked for me:
500 = mysecret,Motty
mailbox,mo...@domain.com,,tz=pacific,attach=yes|delete=1
Thanks,
Motty
On 07/10/2015 07:45 AM, Luca Bertoncello wrote:
Hi again,
I'm trying to send two E-Mails when a message comes in the voicemail,
the first WITH the attachment
would like to add background music if authentication failed, then
after 6 minutes hangup
any ideas, suggestions?
On 07/07/2015 09:09 AM, Motty Cruz wrote:
Hello,
I used this guide, it worked for me:
http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html
Thanks,
On 07/06/2015 04
+Application_Authenticate
You can either give it a single PIN to use for all calls, Authenticate
using a value in the Asterisk Database, Or use a plain text file for
the PIN's
On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz motty.c...@gmail.com
mailto:motty.c...@gmail.com wrote:
Hello All,
I will like
Hello All,
I will like to configure Asterisk to use PIN Code for all outgoing
international calls.
Also, any suggestions as to when should I prompt users for code prior to
dialing the number or after dialing the number?
can someone provide with a example on how to accomplish this goal? I
Hello,
I would like to setup a mechanism to trigger an alarm if user is deal
too many numbers within a very short period of time. Safeguard against
users hacked accounts.
can someone help?
Thanks,
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this code worked for me,
here is what I did and worked for me:
exten = 1381+NXX,1,Set(CALLERID(number)=3817383444)
exten = 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)
Thanks for you help!
On 04/27/2015 02:56 PM, Matt Riddell wrote:
On 27Apr, 2015, at 16:39, Motty Cruz motty.c
I apologize, I coppied the wrong code,
here is the code I am using:
; Adding Area code and striping 9 for local numbers
exten = _9XXX,n,Set(CALLERID(all)= 3817383444)
exten = _9XXX,n,Dial(SIP/intelepeer/1381${EXTEN:1},80)
Thanks,
motty
On 04/28/2015 11:54 AM, Chad Wallace wrote
Thanks for your reply,
[globals]
AREACODE=381
[outbound]
exten = _NXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80)
did not work for me, any ideas?
Thanks,
On 04/27/2015 01:59 PM, Phil Reynolds wrote:
On 27 April 2015 21:32:42 BST, Motty Cruz motty.c...@gmail.com wrote:
Hello,
I
wrote:
Motty
Yes
From your dial plan accept 9 + 7 digits then concat your dialed number
together with your areacode.
This s a brief example.
exten = _9XXX,1,Set(l_HomeAreaCode=555)
exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;;
This line should combine your area code
Hello,
I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my
SIP provider is requiring to dial 10 digits. is it possible to add area
code?
Thanks,
Motty
forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS.
Thanks,
On 04/27/2015 02:38 PM, Motty Cruz wrote:
here is what I have:
exten = _9XXX,1,Set(l_HomeAreaCode=381)
exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1})
exten = _9XXX,n,Dial(SIP/SIP-Provider
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
--
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Motty
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Thank you AJ, I will certainly not copy and past; I want to believe I
understand the risk. I needed some kind of direction, thank you for your
support.
-Motty
On Fri, Sep 19, 2014 at 2:51 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:
On Thursday 18 Sep 2014, motty cruz wrote:
Hello, I
Hello, I would to allow users to place calls overseas such as India and
Malaysia but only with a security code. if they don't have a security code
I want to be able to drop the calls.
can someone point me to a right direction to achieve this goal?
Thanks,
Motty
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk wrote
Thanks Eric, for respectfully pointing that link, it is the reason why I am
posting my question for lack of knowledge. I had been working on Asterisk
for the last 4 years, I am always learning something knew.
- Motty
On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling ewiel...@nyigc.com wrote:
Your
absolutely not what I meant, I really meant to say thank you for
respectfully pointing that out.
-Motty
On Thu, Sep 18, 2014 at 2:32 PM, Eric Wieling ewiel...@nyigc.com wrote:
It is unfortunate
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP
Hello,
a user outside the office regularly gets a call from ext. 101 but that
extension does not exist in my extensions.conf. when the user pickup the
phone no one answers. Any Idea how to fix this issue? that user uses
Polycom SP 450,
Thanks in advance,
Motty
Thanks Eric, for point to polycom instructions I will give it a try.
Kevin, I am sure called is not originating from our system,
Thanks for your support.
On Tue, Sep 16, 2014 at 9:08 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
Hello,
a user outside the office regularly gets a
sure you have tcpdump installed.
tcpdump -lni eth0 -f udp port 5060
monitor your server to make sure you catch the attacker, also you could do
tcpdump -w capture.cap
and analyze it with wireshark.
Thanks,
Motty
On Thu, Sep 11, 2014 at 2:28 PM, Rusty Newton rnew...@digium.com wrote:
On Thu
to cover any holes that attacker is
trying to exploit.
Thanks,
Motty
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,
-Motty
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http://www.asterisk.org/hello
asterisk-users mailing list
...@ovm-group.com wrote:
Am 04.09.2014 16:44, schrieb motty cruz:
Hi All,
I see this kind of attack on our Asterisk Server, do you know how to block
that IP?
[Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite:
Call from '' (213.136.81.166:9306) to extension '34422
Hi A J,
believe me, I wish i do as you suggested, however I have a few extensions
outside the office with dynamic IPs, so that is not a possibility. Thanks
for your suggestions, I will try fail2ban. I don't know how complicated is
to implement that on production server.
Thanks,
-Motty
On Thu
Thank you all for your support, your suggestions are welcome.
Thanks,
On Thu, Sep 4, 2014 at 9:26 AM, Chris Bagnall aster...@lists.minotaur.cc
wrote:
On 4/9/14 4:58 pm, Eric Wieling wrote:
If we don't need to allow access from outside the USA we block access
from all non-ARIN IP addresses
Hello, I want to share mailbox between two extensions
Ext. 101
Ext. 102
I want the messages to go to mailbox 101, when when checked mailbox from
extension 102 to be able to clear the bliking red light.
here is extensions.conf
exten = 102,hint,SIP/${EXTEN}
exten = 102,1,Dial(SIP/101SIP/102,20,t)
checks the voicemail, do they hear the correct voicemails? Ours clears just
fine in this situation.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 04:37:26 PM:
From: motty cruz motty.c...@gmail.com
To: Asterisk
Thanks Kevin,
can you provide me with example of your code? if you don't mind?
Thanks,
On Tue, Jun 24, 2014 at 3:46 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM:
From: motty cruz motty.c...@gmail.com
:)
--
*From:* asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
*Sent:* Monday, May 12, 2014 5:43 PM
*To:* Asterisk Users List
*Subject:* [asterisk-users] Asterisk 1.8.22
Hello,
recently I have seen spike in attacks on my
Hello,
recently I have seen spike in attacks on my asterisk server, this is what I
get on the LCD of my phone: 201@76.220.5.205
or calls from 1000 sip1000@76.2230.5.205,
have any idea on how to stop this calls?
Thanks,
--
_
--
Hello All,
one of the extensions fall into a loop, I don't know how to hangup that
channel
-- Executing [i@autoatten:2] Goto(Local/100@sipphones-01b2;2,
s,2) in new stack
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
:
On Mon, 5 May 2014, motty cruz wrote:
one of the extensions fall into a loop, I don't know how to hangup that
channel
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
Local/200@sipphones-01b2;2
any ideas?
If you're asking how to prevent
.
Since all out clients are under our control we use openvpn a lot and
yealink and other phones have it built in so they can connect directly once
initially setup
Cheers Duncan
On 5/04/2014, at 4:36 am, motty cruz motty.c...@gmail.com wrote:
that sounds feasible, Thanks Michelle,
On Fri
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673
setup.
How many sip phones do you have outside your network? If few and in
well-known IPs, consider limiting access to only those (and the sip
provider you are using).
On 04/04/2014 09:00 AM, motty cruz wrote:
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00
absolutely right A J, thanks for the heads up.
I do not intent to implement that solution in production server, I hope to
learn it first, build a test server and monitor for a few days or weeks.
Thanks again,
On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On
Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.
Thanks for your support.
On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 4 April 2014 15:22, motty cruz
:* asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
*Sent:* Friday, April 4, 2014 11:15 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around
look like the issue continues, I am unable to overwrite callerid from
sip.conf in extensions.conf,
In sip.conf under
[general]
trustrpid = no should i change it to yes?
Thanks
On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote:
Thank you for your reply, I updated
Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,
Thanks,
On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite
Hello, I'm having issues with my phone Polycom sp450 not subscribing to
Asterisk server. Asterisk server is fine, firewall is not the issue because
a secondary phone is working fine, my connection to the server is fine too,
any ideas or suggestions are welcome.
-Motty
number is dark same as the background
so that means is not subscribing to the Asterisk server.
Thank you very much.
On Thu, Jan 2, 2014 at 8:19 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM:
From: motty cruz
Thank you all,
After setting the phone to factory defaults, entered configuration
parameters, phone is working again. I really don't know why all sudden stop
working. at least know i have a working phone I will go thoroughly through
the logs, I hope to find the answer, if I do I will post it
Hello, I have a fully functional Asterisk Server, I want to configure this
server to be able to process call from Skype, can someone point me to a
howto? or if there are suggestions on best way to approach this problem.
Thanks,
--
.
Mitul
On Friday, November 8, 2013, motty cruz wrote:
Hello, I have a fully functional Asterisk Server, I want to configure
this server to be able to process call from Skype, can someone point me to
a howto? or if there are suggestions on best way to approach this problem.
Thanks
Hello All,
I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get
meetme feature to work when dial meetme extension, can you please help?
It always worked before, also I do not have dahdi installed on this
machine, never did.
-- Executing [104@sipphones:1]
Thanks Johan,
I did noticed /etc/dahdi so you're right it was installed on one point, I
re-install dahdi and problem went away.
Thank you very much!
On Thu, Jun 6, 2013 at 1:45 PM, Johan Wilfer li...@jttech.se wrote:
2013-06-06 22:21, motty cruz skrev:
Hello All,
I upgraded Asterisk
Hello,
i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
but no success, I do prefer not to install any web server on the server
running Asterisk.
Thanks in advance.
-Motty
--
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-- Bandwidth
Thanks for the suggestion Carlos,
do you have a HowTo? can you point me to one.
I unsuccessfully follow one found using google. I'm using CentOs 6.0
Thanks,
Motty
On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote:
Monitor what parts exactly?
Right this moment I'm
Thanks for your help; I just want to monitor the queue, calls on hold
average time, incoming out going call, I only want to monitor Asterisk, not
the server Asterisk in running on.
thanks,
-Motty
On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote:
http://opennms.org/wiki
Rose jr...@digium.com wrote:
motty cruz wrote:
Hello,
I'm running Asterisk 1.8.10 on Linux box, when I'm in a
conference(meetme) with another person, and a third person join our
conference when the third person leave the conference I get
disconnected from the original conference
Hello,
I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme)
with another person, and a third person join our conference when the third
person leave the conference I get disconnected from the original conference
with a second party. I hope this clear.
This does not happen
I have Polycom IP550. The Forward No Answer is working fine when
enabled. I was looking at the sip.cfg but don't know exactly what to look
for, can you give me a hint to where would i find that option?
Thanks,
On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill
justin.sherr...@americanrocksalt.com
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