Should a Linksys Sipura 2102 be configured with nat=yes even if it is on
the local network?
I have been having some troubles with a Linksys Sipura 2100 series, which
suffers from NO AUDIO after a few calls.. Because it is on the same subnet
as Asterisk it is configured with nat=no. When you think
My customer needs to set a forwarding based on number of rings,i.e.,
if the phone rings 5 times (user-selectable), then try another number.
Is there a way to do such a thing with Asterisk? I could not find way
to do it based on the documentation of the Dial function. The protocol
is SIP only,
On Fri, 2012-02-17 at 04:00 -0500, CDR wrote:
My customer needs to set a forwarding based on number of rings,i.e.,
if the phone rings 5 times (user-selectable), then try another number.
Is there a way to do such a thing with Asterisk? I could not find way
to do it based on the documentation of
Simply, without checking for BUSY, DND or TIMEOUT
I'm assuming each ring period is 3 seconds.
exten = 8512,1,Dial(SIP/8512,15)
exten = 8512,n,Dial(DAHDI/GO/101233456,15)
Or another way.
Maybe the FollowMe application, allow multiple numbers to be tried, each
after a configured timeout.
from
On Friday 17 February 2012, CDR wrote:
My customer needs to set a forwarding based on number of rings,i.e.,
if the phone rings 5 times (user-selectable), then try another number.
Is there a way to do such a thing with Asterisk? I could not find way
to do it based on the documentation of the
Try this
exten= yournumberhere,1,Dial(SIP/peern1,60)
exten= yournumberhere,n,GotoIf($[${DIALSTATUS} != ANSWER]?4)
exten= yournumberhere,n,Hangup
exten= yournumberhere,n,Dial(SIP/peer2,60)
exten= yournumberhere,n,GotoIf($[${DIALSTATUS} != ANSWER]?9)
exten= yournumberhere,n,Hangup
you can
On 02/17/2012 12:09 AM, Sammy Govind wrote:
Hello,
Thanks for taking out tome for my query. Yes I do have an actual
problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers
port mirrored to it). My end points(soft-phones) are sending RTCP
connection strings to asterisk, and
Hello list,
Kevin I agree with you on independent monitored entity for A leg while the
outbound leg has separate QoS measures. But after this thread I went to my
monitoring tool and saw that for some calls on the same asterisk setup I had
no RTP or RTCP while there were calls with both RTP and
Hi members,
I have a question regarding presence in asterisk.
I have two PBX systems and would like to connect them. After configuring
each other as sip providers calls between users of the pbx systems are
possible.
Now I'm trying to implement presence between the systems. PBX1 sends
I'm reading some information that recommends using SER / OpenSER for
large installation to offload SIP traffic from the Asterisk server.
http://www.voip-info.org/wiki/view/Asterisk+at+large
However, it looks like the information might be dated.
I'm looking at a potential 750 SIP phone and 150
Hi Team,
Does any one know how to set IDLE FEATURES on the Avaya 4610sw IP Phone when it
registers directly on the Asterisk Server ?
I have done many things like setting it through the 46xxsettings.txt but it
doesn't work:
SET IDLEFEATURES 326
SET FBONCASCREEN 0/1 ;;; ( did both
What does the CLI show? The ringing you year is likely the telco ringing, not
the asterisk ringing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: Friday, February 17, 2012 1:26 AM
Richard
I tried this, but it did not work. What can be the problem?
[PABX]
exten = _x.,1,Proceeding()
same = n,GotoIf($[${CHANNEL(reversecharge)} =-1]?allow:block)
same = n(allow),Dial(SIP/1584,30,tT))
same = n(block),Hangup()
Att,
Rafael Saraiva
2012/2/15 Richard Mudgett
Did you set CHANNEL(reversecharge) somewhere?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Friday, February 17, 2012 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I prefer multiple servers sharing the load. All asterisk based. This
let me scale up the power of the system just adding more servers. I
use asterisk 1.8 realtime with all the data (peers, voicemails, ivr
messages and so on) stored in a pair of mysql database with
multimaster replication. Phones
This is a variable received from the isdn channel.
Att,
Rafael Saraiva
2012/2/17 Danny Nicholas da...@debsinc.com
Did you set CHANNEL(reversecharge) somewhere?
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of
I would put a Verbose statement after Proceeding to verify the value returned
from ISDN channel, like this:
- Same = n,Verbose(RC value ${CHANNEL(reversecharge)})
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos
The value is always -1. I must enable something in chan_dahdi to pass the
correct value?
++
[PABX]
exten=_X.,1,Gotoif([${CHANNEL(reversecharge)} = -1]
?entrada,${EXTEN},1:hangup,${EXTEN},1)
+++
rssr305*CLI -- Accepting call from '5132083300' to '1584' on
channel 0/18,
From what I read, your libpri may be out of date.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Friday, February 17, 2012 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Which version do you recommend? Mine is 1.4.12.
Att,
Rafael Saraiva
2012/2/17 Danny Nicholas da...@debsinc.com
From what I read, your libpri may be out of date.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of
On Thu, Feb 16, 2012 at 12:30 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/11/2012 06:59 PM, Bruce B wrote:
If your server is open to the internet and in SIP general section you
have nat=no and in peers you have nat=yes or vice versa then it's
possible to enumerate your extension.
Well, that’s what is on asterisk.org to be downloaded – this one has “left my
pay grade” - good luck.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Friday, February 17, 2012 11:33 AM
To: Asterisk
I'm attempting to pull SIP users from LDAP, following the instructions from
here:
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
However, when I attempt to register a user from LDAP, I see on the console:
chan_sip.c:24431 handle_request_register:
The value is always -1. I must enable something in chan_dahdi to pass
the correct value?
++
[PABX]
exten=_X.,1,Gotoif([${CHANNEL(reversecharge)} = -1]
?entrada,${EXTEN},1:hangup,${EXTEN},1)
+++
rssr305*CLI -- Accepting call from '5132083300' to '1584' on
Wouldn't a shell script be a band-aid solution?
CLI verbose should have absolutely no effect on other loggings. I have been
saying this forever that Asterisk logging should be very strong and
separate of anything else including what we see on the CLI. This is
important for security reasons. You
Reversecharge not appear in debug.
I'm in Brazil, the signaling is different here?
Att,
Rafael Saraiva
2012/2/17 Richard Mudgett rmudg...@digium.com
The value is always -1. I must enable something in chan_dahdi to pass
the correct value?
++
[PABX]
Reversecharge not appear in debug .
I'm in Brazil , the signaling is different here ?
Please capture the incoming SETUP from libpri for the collect call.
pri set debug on span x
Richard
--
_
-- Bandwidth and Colocation
Hello,
with the command lsof -i I notice the following network connections of
the asterisk proces :
asterisk 23006 root 12u IPv4 1088961 UDP
*:mgcp-callagent
asterisk 23006 root 13u IPv4 1088964 TCP *:sieve
(LISTEN)
asterisk 23006
4520 is for DUNDI. Obviously your install uses H323 in some flavor.
Mgcp-callagent is for jitterbuffering? And sieve and complex-main I have no
clue (perhaps H323 tag-alongs)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
On Thu, 16 Feb 2012 19:41:16 +0100, Olivier oza_4...@yahoo.fr wrote:
You mean you can receive SMS on a landline in France (or the opposite) ?
Supposedly, but I never used it either.
www.google.fr/search?q=sms+ligne+fixe+asterisk
If a gateway has its own SIM card and GSM stuff, should it receive
I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga
on Fedora 16 x86_64 for my tests.
On Behalf Of Jonas Kellens
with the command lsof -i I notice the following network connections of the
asterisk proces :
asterisk 23006 root 12u IPv4 1088961 UDP *:mgcp-callagent
asterisk 23006 root 13u IPv4 1088964 TCP *:sieve (LISTEN)
asterisk 23006 root 16u IPv4 1088966 UDP *:iax
Yes, it is telco ringing and asterisk answered that line .
thanks
Dhaval
On Fri, Feb 17, 2012 at 8:35 PM, Eric Wieling ewiel...@nyigc.com wrote:
What does the CLI show? The ringing you year is likely the telco ringing,
not the asterisk ringing.
-Original Message-
From:
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