On 09-09-13 23:11, Niccolò Belli wrote:
Hi,
I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a
web page to confirm the subscriptions. How can I achieve it? Is
Asterisk of any use to send SMS with the Portech? I really have no
idea because I know nothing about the whole SMS
Hello all,
I would like to set a different caller-id for the second leg of a call
when doing an originate.
For example:
Action: Originate
Channel: sip/1234
Context: mycontext
Exten: 1
Priority: 1
Callerid: 123 123
Async: true
This sets the caller-id correctly when dialing sip/1234, but I would
Hi all,
I face the subject strange behavior: calls arre dropped after 15 minutes
on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
through OpenVPN seems to have the problem.
From CDR, I see for 3 calls from this morning I'm aware of, that
asterisk hangup after
hi,
it seems your vpn connection drop.
is you vpn on WiFi of any other high latency network?
On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI ad...@tootai.netwrote:
Hi all,
I face the subject strange behavior: calls arre dropped after 15 minutes
on an asterisk 1.8.15.0. Only phones
Hello fellow asterisk users,
I've been facing a problem when using MeetMe's admin functionality to
unmute users in a conference using *Asterisk 1.6.2.11*.
I've tried:
1) MeetMeUnmute (AMI)
2) MeetMeAdmin(AMI)
3) MeetMeChannelAdmin(AMI)
and also tried via console : asterisk -rx 'meetme unmute
Hello Everyone,
I have a new problem where when placing the call, asterisk will
automatically go into music on hold until the call is connected (ie,
no ringing). It was kind of confusing because sometimes `SESSION
PROGRESS` takes longer than others, during this time we are in MOH.
The call does
Nick Cameo wrote:
There is two way audio, it's just during ringing that this happens.
If you can put the SIP signaling and Asterisk console output up
somewhere then we can have a better idea of what Asterisk is being told,
and what it is doing.
--
Joshua Colp
Digium, Inc. | Senior Software
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?
Asterisk does not assign ports. The IAX2 channel driver, by default,
binds to a single UDP
Does your Dial() command include the m option?
jg
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On 09/09/2013 07:48 PM, Eric Wieling wrote:
Try this as an example of why it doesn't matter.
1) On windows open a cmd prompt or on linux open up a local terminal.
2) open a web browser and connect to a web site like cnn.com
3) on windows type netstat -n in the command prompt, in linux type
There is two way audio, it's just during ringing that this happens.
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Hello Jg,
Thank your for your response. No m option on dial. I think it's a RTP
relay issue however, do not know how to diagnose the SDP payload. Any
help would be appreciated.
N.
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On 9/10/2013 7:05 AM, Administrator TOOTAI wrote:
I face the subject strange behavior: calls arre dropped after 15 minutes
on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
Just for kicks, I would disable session-timers to see if the problem
goes away. in the general
Some providers send a reinvite after 15 min and if asterisk doesn't respond
will disconnect the call
Maybe playaround with canreinvite
--Original Message--
From: Jeremy Kister
Sender: asterisk-users-boun...@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Nick Cameo wrote:
I ran a test call with trace can be found here:
http://pastebin.com/f8MuxaFV
I also wanted to mention that yes we have * setup with disallow=all
and allow=g729 for testing,
maybe permanently if we can successfully setup G729 pass through. That
being said, the same problem is
Maybe the ringtone from downstream is not
reaching asterisk, and thus a2billing is appending the `m` to the dial
command?
With digital systems (starting with ISDN, or so), ringing is signaled, or indicated. The
ringtone is produced locally, either by the PBX or by the SIP phone itself. Since
Yeah of course. Still digging into it :). Will post the solution if I
find it. a2billing forum takes for ever to answer...
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I have no idea where the `m` is coming from. I even looked into the
A2Billing script. Still digging
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Sean Darcy wrote:
On 09/10/2013 12:15 PM, Joshua Colp wrote:
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?
Asterisk does not assign ports.
Yes of course, I just did not want to overwhelm you guys with SIP
trace. Before that though, I realized something:
[Sep 10 12:03:30] WARNING[8178]: res_musiconhold.c:802 set_moh_exec:
SetMusicOnHold application is deprecated and will be removed. Use
Set(CHANNEL(musicclass)=...) instead
-- AGI
On 09/10/2013 05:27 PM, Joshua Colp wrote:
Sean Darcy wrote:
On 09/10/2013 12:15 PM, Joshua Colp wrote:
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them
On 09/10/2013 12:15 PM, Joshua Colp wrote:
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?
Asterisk does not assign ports. The IAX2 channel
Hope this helps someone save a day of running around.
So my issue was with a2billing. The warning `No remote address on RTP
instance '0xb6d16a28' so dropping frame`
was not related to the music on hold coming on during ringing.
The Problem:
We have a script that loads rates into
Did you the a2billing settings for a music on hold setting
I remember seeing some setting
-Original Message-
From: Nick Cameo sym...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 10 Sep 2013 12:46:54
To: Asterisk Users Mailing List - Non-Commercial
Nick Cameo wrote:
Yeah!!! The Dial command setting:
http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704
I know this is not an a2billing mailing list, and I am sorry however,
I do think that the No remote address on RTP instance may have
something to do with it. Maybe the ringtone from
Oh scandalous Instead of playing the MOH, I would like to play the
ringtone that is on the machine. Ummm, where is it? :)
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I ran a test call with trace can be found here:
http://pastebin.com/f8MuxaFV
I also wanted to mention that yes we have * setup with disallow=all
and allow=g729 for testing,
maybe permanently if we can successfully setup G729 pass through. That
being said, the same problem is still there using
i have used a2billing some time ago maybe there is somthing new .
you can try shoot up loglevel to 4 and see the verbose of agi that may give
you some hint.
On Tue, Sep 10, 2013 at 7:34 PM, jg webaccou...@jgoettgens.de wrote:
Maybe the ringtone from downstream is not
reaching asterisk,
Yeah!!! The Dial command setting:
http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704
I know this is not an a2billing mailing list, and I am sorry however,
I do think that the No remote address on RTP instance may have
something to do with it. Maybe the ringtone from downstream is not
hello:
it really depends on number of calls transcded. it the number is less 20
calls, the CPU should be ok.
but it is very sure that software based take take much more than
hardware based transcoder.
? 2013-9-9 15:53, Gopalakrishnan N ??:
Hi,
How much CPU utilization will it take when I use
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