Hi!
I am setting up a little call center, but don't know how the agents system
works, can you guys please give me a little help?
I need to know how asterisk will know when I log agent X, and asterisk know
that agent is in the IP Z with the extension Y.
Thanks a lot.
Hugs,
ARPE
--
Hi mates!
Please, I need to understand how to search for an specific log by date/time
on asterisk logs, but can't understand how this works, can you guys please
give me an example about how those logs works?
Best regards,
Asterisk Guy
1and1 dedicated server's service has been down for a few hours , unable
to reach them by phone or email. do anyone know what is going on there ?
Mario
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1 there is a correct file in /var/spool/asterisk/outgoing
2 i run asterisk -r to monitor it , it gives out the following error
-- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold()
(Retry 1)
Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such
I have setup wake up call in * ( 1.2crc1) following those instructions
http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP
i can enter the time after dialing 77 , and i see there is wakeup files in
/tmp
but * nevers make the wakeup call when it is due , what can be the
are there any good softphone on PDA window mobile 2003 / 5.0 ?
tried sjphone, sound quality is unacceptable.
Mario
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hi
Can anyone recommend a good DID provider offering numbers in Toronto ?
( 1 very stable 2 support porting numbers from Bell, primus, telus.. )
Mario
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looking for good sip softphone for wifi and 3G network.
1 are there any sip softphone ( with gsm/g723/G729 codec ) for
smartphone such as Nokia N90 / 93 / N95 ?
2 are there any sip softphone ( with gsm/g723/G729 codec ) for Window
mobile5 Or wm2003 ?
3 How is the sound
sjphone
firefly (3rd party version)
--source is available ? where?
On 6/15/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jun 15, 2006 at 06:06:40AM -0700, Derek Whitten wrote:
Asterisk guy wrote:
are there any open source sip softphone (Window OS version
are there any open source sip softphone (Window OS version )?
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password online.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
Sent: Tuesday, February 07, 2006 7:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Free IAX login
how to set up iax.conf , so IAX clients
online.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
Sent: Tuesday, February 07, 2006 7:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Free IAX login
how to set up iax.conf , so IAX clients with any
.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
kevin ling wrote:
Not sure answer your question? Try to write some html code and let user
register the username password online.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
how to set up iax.conf , so IAX clients with any user name and any
secret can login to * ? ( no authorize for login )
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how to adjust voice volume for sipura 2000 and cisco ata186?
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does it include the patch for VAD?
( dropping extra frame of G.729 since we already have a VAD frame at the end )
Mario
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How to set asterisk in pass-through mode ?
could you give a sample configure for passthrough?
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does the following patch work for 1.2? how to apply it to 1.2? ( I
am not a programmer, don't know how to use .diff file).
http://bugs.digium.com/view.php?id=5374
silence-suppression-2.diff
On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Asterisk guy wrote:
does it include
does 1.20rc2 includes the patch? will this patch be included in 1.20rc3?
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To
Is the asterisk server actually pushing the bits for a call or just
doing call setup and connecting the two endpoints directly?
how to force asterisk just doing call setup and connecting the two
endpoints directly?reinvite=yes ?
if UA is behind NAT with reinvite=yes , will asterisk
dropping extra frame of G.729 since we already have a VAD frame at the end-
just tested 1.20rc1, it is still there. where to get patch for 1.20rc1?
does 1.20rc2 includes the patch?
On 10/29/05, Kanishka Somaratne [EMAIL PROTECTED] wrote:
Hi
I get the following error when i make a call
When get sip respond 6xx ( such as 603 decline), I want asterisk to
play a voice file to the caller, how to do this in extensions ?
for example, when get 603 respond, play decline.gsm to caller
when get 604 respond, play doesnot-exit.gsm to caller
when
When get sip respond 6xx ( such as 603 decline), I want asterisk to
play a voice file to the caller, how to do this in extensions ?
for example, when get 603 respond, play decline.gsm to caller
when get 604 respond, play doesnot-exit.gsm to caller
when
Are there any command to show a calls is in transcoding or
passthrough codec mode ?
Mario
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I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323
in
--may i know which version of asterisk and oh323?
On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Juanjo,
can you provide some more detail about which version you are using both for
asterisk and
After more test, get the following:
Oh323's call-id works on asterisk1.07+ oh323 (0.66), doesn't work
on ( asterisk1.2.0beta1+oh323 0.73), any pathch to get oh323 0.73
works?
On 10/2/05, Asterisk guy [EMAIL PROTECTED] wrote:
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73
may i know which version oh323?
On 10/4/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in
Buenos Aires, Argentina. Currently I'm using direct connections to the
telephone company's (iplan) H.323 gateway, but I'm working
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote:
I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I
set the caller id correctly in my perl AGI script
$AGI-set_callerid($ani); , the
I am trying oh323(version 0.67) , make call from sip UA to h323 gateway,
can't get Call-id pass from sip UA to h323 gateway, h323 always gets
call-ID sent from Asterisk as *. are there any configure to pass
the correct call-id from sip UA to h323 gateway? or this is a bug in
oh323 0.67?
HI
I got the following error during compiling ooh323.
In function `h323_set_rtp_peer':chan_h323.c:2745: error: structure has no member named `tech_pvt'chan_h323.c: At top level:chan_h323.c:68: error: storage size of `h323_tech' isn't knownmake[2]: *** [chan_h323.lo] Error 1
make[2]: Leaving
for sip calls, asterisk is able to convert a incoming g729 cal to a outgoing G.711 call.
Foroh323, I am unable toget asterisk to convert a incoming g729 call to a outgoing G711 call .
my question is :For h323, how to configure asterisk to convert a incoming h323/g729 calls to a outgoing h323
got a DID from libretel. Libretel sends CIDnumber as +number ( such as +71898765421) or just number( such as 71898765421)
I want to get rid of + before the number, how to setup this in extensions.conf ?
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i have an account with BV on my asterisk, how to forward a unanswered incoming call to my mobile phone ( when there is no one to answer the incoming call after 3 rings) ?
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for every call, * gives out:
Spawn extension (default, 00x, 3) exited non-zero on 'SIP/201.50.117.161-081628e0'
Spawn extension -what does this mean ? how to avoid this ?
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oh323 show conf :
..
max call rate (ingress direction) 1.00/30 --what does1.00/30 mean ? how to increase it ?
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To
sometime, * gives out tons of :
chan_oh323.c:2143 oh323_write: warning : OH323/L1648 invalid
frame size for G.729( 2 bytes)
chan_oh323.c:2143 oh323_write: warning : OH323/L1648 invalid frame
size for G.729( 12 bytes)
is there anything wrong ?
how to fix it?
Mario
you can tell * donesn't load these modules in modules.conf
On 5/3/05, Kumara Jayaweera [EMAIL PROTECTED] wrote:
Greetings to all!
Sorry for the numerous postings. but How could I slim my Asterisk PBX.
Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't
have any
how stable is oh323 ?
is there any production implement ? ( which version 0.65 or 0.7?)
could you share the experience ?
Mario
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1 after giving command oh323 show channels,
i want to disconnect a call, is there any command to disconnect a call?
2 how asterisk kill a hung/dead call ? for most commercial
softswitch, there are a setting for maximum duration for a call. they
will hang up it l if its duration reachs
which TCP port is used when asterisk -r ?
is there a command to connect to a remote machine ?
( asterisk -r remote-machine-ip ?)
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there is jitter setting for h323 in oh323.conf
where to set min/max jitter buffer for SIP ?
i am getting bad voice via *, maybe this jitter buffer setting will help
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there is jitter setting for h323 in oh323.conf
where to set min/max jitter buffer for SIP ?
i am getting bad voice via *, maybe this jitter buffer setting will help
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i am trying to get G723 passthrough
get the same error.
how to configure passthrough for g723/g729 ?
On 4/24/05, Brian Capouch [EMAIL PROTECTED] wrote:
jltaylor wrote:
;;;
Brian,
Add to the [general] section in sip.conf the following:
disallow=all
i get the same problem.
My option is to change my asterisk box to work
completely on a different port, but after set bindport=5061 in sip.conf,
asterisk still listens on 5060 after resarting.
it seems bindport setting doesnt work.
any idear on how to change the sip listening port ?
G729 will not work without a licensecan't G729 work in
passthrough mode without license?
if yes, how to configure it work in passthrough mode?
On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED]
www.mutualphone.com
On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
calls. Calls that are being routed to wrong numbers. DID's that ring
busy. For the pass 2 days I am unable to pass CID. Is anyone else
Hi,
i am trying to connect to mutualphone.com service. but they only
support G.729 /G.723?
Must I buy/install G.729 codec at asterisk in order to connect to
mutualphone?
or I can use the pass-through mode to connect to it without G.729
installed ? if yes, how to configure G.729
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