Hi,
I've a problem configuring my Asterisk. What I try to reach is to
interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP)
with 1 constraint I can't change : every RTP flow needs to pass THROUGH
Asterisk, and are NOT nated
What I observe :
- a call made from a SIP Phone
hi, group
there are two module can used for meeting. MeetMe and
Conference(which is a plugin)
My question is :
which is better for large conference(maybe above 100 people in a meeting)?
--
Thanks Regards
Sucan
--
_
--
Hello,
I wish to make a simple system to limit peers at x minutes depending
of buyer voip packet. Can someone help me with some directions?
I intend to make a separate dial plan and every calls to be in cdr
table in mysql. Is any chance to make some scripts to drop calls after
peer
used all
BTW, using the most common Asterisk distros out there that happen to sport a
very complex dialplan, we see a lot of lost events, so that tracking calls
on the basis of AMI observation alone becomes practically impossible.
:-(
l.
2010/8/8 Nasir Iqbal na...@ictinnovations.com
Hi,
Hi,
It is simple to use max_limit perameter in dial command.
Regards,
Faisal Hanif
On 8/9/2010 2:01 PM, Catalin S. wrote:
Hello,
I wish to make a simple system to limit peers at x minutes depending
of buyer voip packet. Can someone help me with some directions?
I intend to make a separate
Hi
Use Set(TIMEOUT(absolute)=XYZ) in your dialplan or timeout parameter in
Dial and Originate commands. Get maximum available seconds from your db for
calling peer and use it as timeout. But after every call you have to
deduct used time from you db for calling peer.
Regards
On Mon, Aug 9, 2010
Hello Miguel molina,
I did what you asked, but still the voice is too bad
Regards
On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina mmol...@millenium.com.cowrote:
El 05/08/10 14:50, Tim Nelson escribió:
- michel freiha mich...@gmail.com mich...@gmail.com wrote:
Dear Sir,
I tried
Hi max,
Have look on my blog regarding this.
http://ashikalim.blogspot.com/2010/08/config-asterisk-india-pstn-lines_09.html
Thanks,
Ashik
On Sat, Aug 7, 2010 at 11:15 AM, Max Alex max.aster...@gmail.com wrote:
Hi All,
I have Sangoma A200 Card installed on my system,
I have centos 5.5 with
Ladies, Gentlemen
We are experiencing an unusual problem in our asterisk 1.4.34.. We are
attempting to determine if channels are in use before paging to them.
This works correctly, as in it pages the phone.. however, we see the error
message below on the console... after googling, we
maybe, can use the sneplivre for this...
www.sneplivre.com.br
detail is that in portuguese, this can be translated easily (i think)
Renato dos Santos
ren...@opens.com.br
OpenS Tecnologia Ltda
Rua Padre Marcelino Champagnat, 236
Jardim Atlântico - Florianópolis - SC - Brasil
+55 (48) 3954-8000
Hallo Keane,
I truly have a nagios server, up and running 24/7
--
Richard Zulu
Managing Director
Time Information Company
P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug
Mobile :+256752624006
Skype: zulu.richard
--
How does one redirect calls based on incoming number or caller ID or the
lack thereof?
current I have for number 123-4567 that it redirects all 800 , 877 and
866 numbers to Voicemail directly.
If the primary area code is 352 then accept this and pass it to
extension
exten =
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
kisho...@techroutes.com
Subject: [asterisk-users] Scilence problem on running call
Importance: High
Dear All,
I am getting scilence for 2-3 second in running calls on E1 CAS in
Asterisk ..
El 09/08/10 05:30, michel freiha escribió:
Hello Miguel molina,
I did what you asked, but still the voice is too bad
Regards
On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina
mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote:
El 05/08/10 14:50, Tim Nelson escribió:
-
On Mon, Aug 9, 2010 at 12:08 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
BTW, using the most common Asterisk distros out there that happen to sport a
very complex dialplan, we see a lot of lost events, so that tracking calls
on the basis of AMI observation alone becomes practically
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun bit...@gmail.com wrote:
hi, group
there are two module can used for meeting. MeetMe and
Conference(which is a plugin)
My question is :
which is better for large conference(maybe above 100 people in a meeting)?
There's at least one more
Hi,
I have problem in initiating an dial out call with SIP response 500 Server
Internal Error
The sip debug as
== Using SIP RTP CoS mark 5
Audio is at 113.253.226.92 port 18284
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Subject: Re: [asterisk-users] SIP response 500 Server Internal Error
Hi,
I have problem in initiating an dial out call with SIP response 500
Server Internal Error
The
Hello list!!
I want to connect an open call with an extension. I call in with a DID, them
redirect to the extension using AGI. Can I use agi's originate to make the
second call without dropping the first DID call? How would I go about this?
I had something like this in mind:
first answer the
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones
Subject: [asterisk-users] Connecting two calls with Originate
Hello list!!
I want to connect an open call with an extension. I call in with a DID,
them redirect to the
Wow, that was fast. Thanks for your reply!!!
So if I were to do:
Action: login
Username:
Secret:
Events: off
Action: Originate
Channel: SIP/trunk
Context: context-for-second-call
Exten: secondCall
Priority: 1
Callerid: CallerID
Timeout: 30
I could connect the 2 calls?
It's my first
Greetings and salutations Asterisk community,
I've been contacted by a man who has generously posted some prompts he
commissioned from Allison Smith. If you haven't heard Allison in humor
mode, you owe it to yourself to hear this. Joey Lindstrom has decided
to place these in the public domain and
s,n,GoToIf($[${AVAILORIGCHAN} = ]?fail:autoanswer) ; SUSPECTED
ISSUE
You need quotes around your variable as well as your evaluation ($
{AVILORIGCHAN} = ).
Thanks,
--Warren Selby
On Aug 9, 2010, at 7:27 AM, Positively Optimistic
positivelyoptimis...@gmail.com
wrote:
Ladies,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones
Sent: Monday, August 09, 2010 11:22 AM
Subject: Re: [asterisk-users] Connecting two calls with Originate
On Mon, Aug 9, 2010 at 10:00 AM, Danny Nicholas da...@debsinc.com
Hello,
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For example, i would like to get the output of following System application
and use its value in next line
for decision making
exten = 5000,n,System(command)
--
Barry Fawthrop wrote:
How does one redirect calls based on incoming number or caller ID or the
lack thereof?
current I have for number 123-4567 that it redirects all 800 , 877 and
866 numbers to Voicemail directly.
If the primary area code is 352 then accept this and pass it to
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino
Subject: [asterisk-users] 'System' application in asterisk
Hello,
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For
On Monday 09 August 2010 13:08:19 Tino wrote:
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For example, i would like to get the output of following System application
and use its value in next line
for decision making
exten =
Hi guys,
is there a way to see how many channels of an specific tecnology are being
used?
Like, i have a zap card, e1 (30 channels), and there are 10 channels being
used at this moment. When the E1 reaches 15 busy channels I need to receive
a call or something like this, telling me that 15 of 30
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the correct way to send Caller-ID by set standards?
--
_
-- Bandwidth and Colocation Provided by
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Subject: [asterisk-users] check channels
Hi guys,
is there a way to see how many channels of an specific tecnology are being
used?
Like, i have a zap card, e1 (30
Thanks Danny,
but the system won't know exactly how many channels are being used right? if
I use the asterisk -rx cmd, this is the result:
Zap/63-1 (None) Up Bridged Call(SIP/xxx)
It won't show how many zap channels are busy . I need to count the busy
channels,
On Mon, Aug 9, 2010 at 4:14 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Subject: [asterisk-users] check channels
Hi guys,
is there a way to see how many channels of an
On Mon, 9 Aug 2010, Felipe Figueiredo wrote:
is there a way to see how many channels of an specific tecnology are
being used?
See? From where? Within the dialplan or from an external process?
Like, i have a zap card, e1 (30 channels), and there are 10 channels
being used at this moment.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt
Subject: [asterisk-users] Correct Caller-ID
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the correct
Continental US-48.
On Mon, Aug 9, 2010 at 3:36 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Matt
*Subject:* [asterisk-users] Correct Caller-ID
I've seen caller-id come through
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt
Subject: Re: [asterisk-users] Correct Caller-ID
Continental US-48.
On Mon, Aug 9, 2010 at 3:36 PM, Danny Nicholas da...@debsinc.com wrote:
From:
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the correct way to send Caller-ID by set
standards?
The correct answer to this depends on where you are.
IMO the answer would be #2, but #3 would probably be
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt
Subject: Re: [asterisk-users] Correct Caller-ID
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the
On Monday 09 August 2010 14:03:36 Matt wrote:
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the correct way to send Caller-ID by set standards?
Given that you can only send CallerID on a digital circuit, it's going
I have been working on this for a while today, and still no luck. This is my
script:
#!/usr/bin/php
?php
$errno=0;
$errstr=0;
$fp = fsockopen (localhost,5038,$errno,$errstr,20);
if (!$fp) {
echo $errstr ($errno)br\n;
} else {
fputs($fp, Action: Login\r\n);
fputs($fp,
I try to disable firewall but no working. I use a softphone to connect on
the same lan segment, it works. Dial in is no problem but dial out always
have this error
On Mon, Aug 9, 2010 at 11:21 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com
Matt wrote:
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the correct way to send Caller-ID by set
standards?
The correct answer to this depends on where you are.
IMO the answer would
On Mon, 9 Aug 2010, Kathryn Jones wrote:
I have been working on this for a while today, and still no luck. This is my
script:
#!/usr/bin/php
?php
$errno=0;
$errstr=0;
$fp = fsockopen (localhost,5038,$errno,$errstr,20);
if (!$fp) {
echo $errstr ($errno)br\n;
} else {
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, August 09, 2010 7:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connecting two
Steve,
you are right, i'm gonna use the group function, I tested here and it works
pretty fine. Thanks.
Danny, thanks for the help once again!
On Mon, Aug 9, 2010 at 4:36 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 9 Aug 2010, Felipe Figueiredo wrote:
is there a way to see how
Hi all,
How can I playback a file within an active call?
I've tried with ChanSpy whisper mode like this (using AMI):
Action: Originate
Channel: Local/9...@default
Priority: 0
Variable: MSG=test
Application: ChanSpy
Data: SIP/1234-123
Async: 1
and in the dialplan:
[default]
exten =
On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup,
such as:
== Registered custom function 'SIP_HEADER'
[Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot
find variable 'SIPPEER' in tree 'description'
== Registered custom function 'SIPPEER'
[Aug 9
On Monday 09 August 2010 21:13:49 sean darcy wrote:
On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup,
such as:
== Registered custom function 'SIP_HEADER'
[Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot
find variable 'SIPPEER' in tree 'description'
Your ami packet is not setting the w option for chanspy, nor I am sure you can
do this.
You might want to create an additional exten that takes a variable from your
ami packet and does the chanspy that way.
I use an ami packet like this with extension that do the work.
Action: Originate
You need to write an external
application either on AMI to keep track of channels
or an external application can get channel list by using shell
command "/usr/sbin/asterisk -rx 'show channels'|grep zap" and then
can count the output and generate a callback file to
Dear all,
What is the difference between SIPp and SER(Sip Express Router)? Which one
is better load performance testing?
Is there any one who knows about this? Could you please give me details
informtaion?
Thans in advance
Nahar
--
Hi,
SER is a most powerful SIP router but a SIPp is a VoIP load generation
software. So both are totally different and can not be used interchangably.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S//
On 8/10/2010 10:44 AM, kamrun nahar bina wrote:
Dear all,
What is the difference
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