Is reinvite set tp yes for the device?
-Original Message-
From: "Joseph L. Casale" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: 5/25/08 7:52 PM
Subject: Re: [asterisk-users] End call behaviour
>What type endpoint do you have ? Channel bank p
I have a lot of these messages popping up in my mesages, E1 connection
shows provisioned up active but I cant seem to be able to make a call.
It was previously working before but stopped working after I did a
reboot to the box this weekend. Anything I am missing out
May 26 08:21:38 asterisk[205]:
>What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a
>SIP phone ?
Hi,
These are SIP phones (Snom M3's and Astra 480i's), I didn't notice this when I
was testing with my softphone
but I cant recall :)
Thanks!
jlc
___
-- Bandw
RTP DEBUG IP from the asterisk CLI
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of ronald ramos [EMAIL
PROTECTED]
Sent: Sunday, May 25, 2008 4:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] trying directrtpsetup
Hi,
I recen
What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a
SIP phone ?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Joseph L. Casale [EMAIL
PROTECTED]
Sent: Sunday, May 25, 2008 11:58 AM
To: 'asterisk-users@lists.digium.com'
S
In your account settings (sip.conf) for the PAP2T device, do you have reinvite
enabled for one or both.. the SIP provider and/or the PAP2T device ?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Shaun Wingrin [EMAIL
PROTECTED]
Sent: Sunday, May 25, 2008 3
Bonjour, connaissez-vous Ziki.com ? Ziki est le lieu où chacun peut se promouvoir librement. Promouvoir votre identité, vos contenus, vos compétences, vos services... Etre référencé en 1ère position sur le moteur de recherche Google. Découvrir et contacter gratuitement des personnes qui vous ressem
Shaun schrieb:
> Hi All,
>
> This is puzzling me greatly.
>
> The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to
> Asterisk are SIP clients. Codec throughout G729 (only have 1 license on
> Asterisk server loaded though). When calling the SIP clients from PAP2T I
> c
I had my incoming call time set 120 seconds before going to voicemail,
apparently this
timeout is longer than some existing timeout of ~60 seconds and the call
terminates
before it reaches my voicemail command.
Is this an Asterisk default setting or could this be something on my SIP
providers
When I exit voicemail or an inbound caller hangs up I hear a busy signal for a
few seconds before Asterisk
terminates the call. I thought this behavior was handled in the dial plan with
a Hangup() command?
How can I correct this?
Thanks,
jlc
___
-- B
On Sunday 25 May 2008 09:38:28 Adrian Marsh wrote:
> I did wonder where the extra spaces were coming from, but I thought that
> was where the quotes were supposed to come into play... Well that got
> it working so thanks guys..
The quotes generally aren't supposed to take care of spaces; they are
Hi All,
What could be the cause why there is no audio coming form the participants.
ztdummy is loaded, ZTDUMMY/1 (source: HRtimer) 1.
I can hear "Please enter your PIN", "User blah blah has enttered"...etc etc
But when the particpants talk, we hear nothing. What are the possible mistakes
i did
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 25 May 2008 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Logical AND
On Sunday 25 May 2008 07:10:22 Adrian Marsh wrote:
> exten => s,n,ExecIf( $[ $[ "${PSTN_NUM:
On Sunday 25 May 2008 07:10:22 Adrian Marsh wrote:
> exten => s,n,ExecIf( $[ $[ "${PSTN_NUM:0:1}" != "0" ] & $[
> ${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
>
> -- Executing [EMAIL PROTECTED]:8] NoOp("SIP/427-b7d9a9a0",
> "0123456789") in new stack
> -- Executing [EMAIL PR
Hello Randy,I would like to read your daily updates in regards to Asterisk Tag in Berlin, Germany.Regards,Eddie
Original Message
Subject: [asterisk-users] Asterisk Tag Berlin live notes page
From: randulo <[EMAIL PROTECTED]>
Date: Sun, May 25, 2008 2:46 am
To: "VOIP Users Confer
Hi Steve,
I can see what yours does, but I still get the same end result (even
though theres only a single "0" result now)
:
exten => s,n,ExecIf( $[ $[ "${PSTN_NUM:0:1}" != "0" ] & $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp("SIP/
On Sun, May 25, 2008 at 10:45 AM, ronald ramos <[EMAIL PROTECTED]> wrote:
> Hi,
>
>
> I recently installed asterisk, i used sterisk-1.4.20.1, i i set
> directrtpsetup to yes, no whow would i know if the rtp/media is not passing
> to asterisk. any tool> or can u just sniff?
You could type rtp deb
Adrian Marsh schrieb:
> Hi All,
>
> I'm trying to figure out why in the below code, the PSTN_NUM variable is
> always amended
>
> exten => s,n,NoOp(${PSTN_NUM})
> exten => s,n,ExecIf( $[ "${PSTN_NUM:0:1}" != "0" ] & $[
> ${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
> exten => s,n,N
Thanks Sherwood,
But how do I send back a 302, once I'm already in the dialplan (hasn't
asterisk already sent back a 200 OK by this point??)
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: 23 May 2008 17:25
To: Asterisk User
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten => s,n,NoOp(${PSTN_NUM})
exten => s,n,ExecIf( $[ "${PSTN_NUM:0:1}" != "0" ] & $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten => s,n,NoOp(${PSTN_NUM})
-- Executing [EMA
Thanks Sherwood,
But how do I send back a 302, once I'm already in the dialplan (hasn't
asterisk already sent back a 200 OK by this point??)
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: 23 May 2008 17:25
To: Asterisk User
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten => s,n,NoOp(${PSTN_NUM})
exten => s,n,ExecIf( $[ "${PSTN_NUM:0:1}" != "0" ] & $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten => s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PR
ronald ramos wrote:
> Hi,
>
>
> I recently installed asterisk, i used sterisk-1.4.20.1, i i set
> directrtpsetup to yes, no whow would i know if the rtp/media is not passing
> to asterisk. any tool> or can u just sniff?
Sniff.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys
Hi,
I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup
to yes, no whow would i know if the rtp/media is not passing to asterisk. any
tool> or can u just sniff?
regards,
ron
___
-- Bandwidth and Colocation Provid
Hello,
1) The goal is to store the id of the operator who take the call in a mysql
database andI don't know the (best) way to know which device take the call when
we do a Dial cmd to a group of phones
Some $var as DIALEDPEERNUMBER ? some inheritance ? using the G extension ? (and
how to?)
T
Hey thanks for the help :)
though i already did that, and the sip debugging info shows me tht its ringing
on the respective sip extension (1002) but nothing is happening..
so i guess its true it IS a diala plan issue tht i am yet to figure it out ...
> Date: Sat, 24 May 2008 14:20:45 +0100
>
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