Displaying 20 results from an estimated 60000 matches similar to: "No subject"
2007 Jul 12
0
No subject
Enhanced OS.
General rules I use:
-Do not use SIP transformations (the VOIP tab), these cause random RTP =
issues, and once you start forwarding calls between users, all things go =
to heck. You are better off using NAT/qualify in your sip.conf.
-Do not use SonicOS Standard (all new Sonicwalls should come with =
Enhanced now anyway) as there is no method to increase the timeout for =
UDP rules,
2010 Oct 24
0
Default MOH not working on 1.6.1 [SOLVED]
2010/10/24 Olivier <oza_4h07 at yahoo.fr>
>
>
> 2010/10/14 Danny Nicholas <danny at debsinc.com>
>
>> ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier
>>
>> *Sent:* Thursday, October 14, 2010 3:34 PM
>> *To:*
2011 Apr 12
0
No subject
then the RTP must go through Asterisk since we need to know if any DTMF =
was pressed.
If both users and the server are not behind NAT and we set up for the =
RPT to go direct if I am using INFO for DTMF, when using the t or T =
option does the RTP go direct ?
I am asking since from what I understand out of band DTMF was created so =
that a leg in the middle can get the DTMF with out having to
2014 Jan 08
0
(no subject)
Hi, all
I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it "Asterisk11".
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then Asterisk11 will generate a CDR records to MySQL's cdr table(in
database "mydatabase") via cdr_adaptive_odbc.
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2005 Jul 01
1
no voice
Hi All
We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways
ERROR CONDITION
---------------
-- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and
2010 Aug 23
2
How to prevent soft hangup from being necessary ?
Hi,
2020 Sep 22
0
Asterisk Drop call
Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if both parties are getting ACK correctly after RTP
starts.
*--*
*Atenciosamente,*
*Luciano Moreira**(85)99974-2750*
*__Logic Telecom*
*0800-085-7799 | (85)4042-7799 | **(11)4210-7799*
Em ter., 22 de set. de 2020 às 13:35, Roberto <
roberto.medola at gasparimsantos.com.br>
2007 May 24
0
Re: asterisk-users Digest, Vol 34, Issue 114
I am running asterisk 1.2.12.1
JK,
Message: 26
Date: Thu, 24 May 2007 21:40:31 -0700
From: JK <jk@bingoconsulting.com>
Subject: [asterisk-users] Urgent: DTMF does not work with rtpmap:101
telephone-event/8000
To: asterisk-users@lists.digium.com
Message-ID: <465668BF.6080800@bingoconsulting.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello asterisk-users list.
I
2010 Feb 16
1
chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds
Hello My friends,
Today my asterisk stop working and i could see the following messags in
/var/log/asterisk/messages at the time that asterisk stop working:
[Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms
/ 2000ms)
[Feb 16 13:24:41] NOTICE[8230] chan_sip.c: Disconnecting call
'SIP/302-b720dd78' for lack of RTP activity in 301 seconds
[Feb 16 13:25:54]
2010 Sep 20
0
No subject
connection will remain a TCP connection unless it is broken and restarted.
Usually if I stop the client and wait for about 30 seconds to reconnect,
there is a much greater chance that the MTU probes work fine, and in about
30 seconds MTU is fixed to 1416.
Every time when the MTU probing fails, I see latency between 700 - 1000 ms
with 32 byte pings over a LAN.
Every time when the MTU probing does
2009 Jul 21
0
Audio lost on reinvite
Hello, all. We are having a problem where audio for sip channels is
dropping upon reinvite. Perhaps it reflects a misunderstanding of what
reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3.
SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set
to both yes and no. We have also tried extending the Asterisk rtp port
range to accommodate the differing default ranges of
2005 May 15
1
can't CLI> STOP NOW by zombie MOH
I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH
fine. After I stop MOH on Windows Messenger, if the hungup signal could
not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains.
Then the user trys again MOH, a new sip channel starts. And again
the hugup signal can not send to *,.........
When I 'stop now' from CLI> , * cleanups the remaining sip
2014 Oct 14
1
debugging T.38 issues
Hello list,
We're currently facing some issues concerning T.38 gateway faxing.
This is a device used almost exclusively for receiving faxes. Calls
are incoming to asterisk on a SIP trunk (sangoma netborder) using
G711A. Gateway mode is activated in the asterisk dialplan towards a
Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0
with the T.38 gateway patch applied (I know I
2004 Sep 01
2
Hung SIP channels
I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong.
The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops
2007 Jul 12
0
No subject
Olle ?) aiming to unify logging, eventing, monitoring (AMI, SNMP, ...)
APIs.
I think that thread occurred when it was decided to include a version number
in Manager interface.
I agree this is an interesting idea ...
The use case that made me ask this is here :
I've got a running system which is working ok up to a moment it stops to
dial out on ISDN-BRI spans (incoming calls are ok). When
2007 Jul 12
0
No subject
<digitmap
=20
dialplan.digitmap=3D"[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxx=
x
x|[2-9]xxxT"
dialplan.digitmap.timeOut=3D"3|3|3|3|3|3"/>
Don't think it's been modified from the original supplied.
...brig
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your =
recipient is using a codec that isn't ulaw or alaw).
=20
_____ =20
From: asterisk-users-bounces at lists.digium.com =
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel =
freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jul 20
0
No subject
stack
pbx.c: -- Executing [s at from-pstn-4:2] Answer("DAHDI/4-1", "") in new
stack
pbx.c: -- Executing [s at from-pstn-4:3] Dial("DAHDI/4-1", "SIP/1000")
in new stack
netsock.c: == Using SIP RTP TOS bits 184
netsock.c: == Using SIP RTP CoS mark 5
app_dial.c: -- Called 1000
app_dial.c: -- SIP/1000-00000012 is ringing
pbx.c: == Spawn