On 03/14/2011 05:06 PM, Steven Howes wrote:
On 14 Mar 2011, at 15:58, Jonas Kellens wrote:
dialplan :
exten = 67121212,1,NoOp()
exten = 67121212,n,Set(CALLERID(all)=3259 3259)
exten = 67121212,n,SIPAddHeader(P-Preferred-Identity:
sip:3259\;user=phone)
exten =
Hi all
how to send SIP HOLD Invite from asterisk to other sip client/server.?
Thanks
Nikhil
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Hi All
Just finished setting up a vm with centos 5.5 and asterisk 1.8.3
Using timerfd as a timing source.
Has anyone got a similar setup in production ?
How's performance?
Thanks,
Neeraj
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On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader... Where does
this come from ?! Not from my dialplan...
sendrpid in your sip.conf
Steve--
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On 03/15/2011 12:24 PM, Steven Howes wrote:
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader... Where
does this come from ?! Not from my dialplan...
sendrpid in your sip.conf
Steve
Not really :
[3259]
type=peer
host=ip_itsp
On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
On 03/15/2011 12:24 PM, Steven Howes wrote:
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader... Where does
this come from ?! Not from my dialplan...
sendrpid in your sip.conf
Not
Hi all,
What is the unit of asterisk AMI events timestamp value?
milli/micro etc ?
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Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Hello
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
== extensions.conf
;Play MoH for a few seconds, hang up, and
;check
I am using asterisk 1.8.3.
I am getting this error:
[Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite:
Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected
because extension not found in context 'smvoice-mediaport'.
dialplan show gives me that the context
You can try changing the priority of
'1104' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]
tp this
'1104' = 2. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]
On Tue, Mar 15, 2011 at 6:59 PM, Jerry Geis ge...@pagestation.com wrote:
I am using asterisk
Hello folks,
since I started with asterisk 1.8.2 I got this messages in my console when
finish a call.
-- Executing [1610@from-e1:1] Dial(SIP/xxx-0027, SIP/1610,60) in new
stack
== Using SIP RTP CoS mark 5
-- Called 1610
-- SIP/1610-0028 is ringing
--
Shouldn't that be
Exten = 1104, 1, Goto(smvoice-mediaport-public-address,s,1)
-Original Message-
From: Rizwan Hisham rizwanhas...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 15 Mar 2011 19:03:33
To: Asterisk Users Mailing List - Non-Commercial
Isrlgb,
Its the output from dialplan show command. The actual entry for the
extension has to be like what you said.
On Tue, Mar 15, 2011 at 7:16 PM, isr...@gmail.com wrote:
Shouldn't that be
Exten = 1104, 1, Goto(smvoice-mediaport-public-address,s,1)
-Original Message-
From:
Hey,
Could you give me some idea how to do this ? I meant record and play ? do you
want me to use .call file ?
-Satish
Date: Mon, 14 Mar 2011 16:29:19 +
From: a...@datavox.co.uk
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
If
On 11-03-15 10:11 AM, Fellipe Paes wrote:
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt: Retransmission
timeout reached on transmission
On 03/15/2011 12:39 PM, Steven Howes wrote:
On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
On 03/15/2011 12:24 PM, Steven Howes wrote:
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader...
Where does this come from ?! Not from my
On 15 Mar 2011, at 15:21, Jonas Kellens wrote:
On 03/15/2011 12:39 PM, Steven Howes wrote:
On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
On 03/15/2011 12:24 PM, Steven Howes wrote:
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader...
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr
wrote:
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
It looks like
On 03/15/2011 10:18 AM, Paul Belanger wrote:
On 11-03-15 10:11 AM, Fellipe Paes wrote:
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
[Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt:
On 03/15/2011 04:38 AM, Neeraj Chand wrote:
Hi All
Just finished setting up a vm with centos 5.5 and asterisk 1.8.3
Using timerfd as a timing source.
Has anyone got a similar setup in production ?
How's performance?
res_timing_timerfd currently has a nasty problem that can cause
deadlocks
Hello
For some reason, when dialing out through a call file and the remote
line is busy, Asterisk doesn't jump to the failed extension in the
context used by the call file:
== call file
Channel: Zap/1/5551234
Context: callbacktest
Extension: start
Priority: 1
MaxRetries: 1
==
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Tuesday, March 15, 2011 11:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [1.4] Failed callfile doesn't jump to
failedextension
Hi Paul,
thanks for your answer, I'll open this issue on the tracker, but now I have a
new question, is there some way to deactivate IPv6 on Asterisk 1.8?
Best regards,
Fellipe
Date: Tue, 15 Mar 2011 11:18:30 -0400
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject:
On 3/15/2011 11:18 AM, Paul Belanger wrote:
Theses are leftover issue with the IPv6 conversion for Asterisk 1.8.
Collect a complete debug log[1] and open a new issue on the tracker.
I believe one was entered a few months ago-
https://issues.asterisk.org/view.php?id=18514
--
Jeremy Kister
On 11-03-15 12:51 PM, Fellipe Paes wrote:
thanks for your answer, I'll open this issue on the tracker, but now I have a
new question, is there some way to deactivate IPv6 on Asterisk 1.8?
You can enable / disable IPv6 via the config files, but the core API /
ABI have changed.
But Kevin is
Hello guys,
I received the following answer from Digium, I understand and will change my
dialplan, but I have one more question now, why I can't use _. in my dialplan?
-
As suggested by Kevin Fleming on the asterisk-users list, this is not a
bug.
I believe the
On 03/15/2011 12:34 PM, Fellipe Paes wrote:
why I can't use _. in my dialplan?
Because it matches everything. In this case, it's matching the 'h' exten. So
when the call gets hung up, it goes to _. and does what you're seeing.
--
On 11-03-15 09:54 AM, Gilles wrote:
Even after callee at 5551234 hangs up, Asterisk keeps looping in
extension , and only runs 's Hangup after runs Hangup.
I also tried calling out through a callfile, same result.
Is there another instruction I should use in to have
Thought this might interest a few people on the asterisk list as well
Cheers,
Dean
From: Dean Collins
Sent: Tuesday, March 15, 2011 1:49 PM
To: 'newtec...@meetup.com'
Subject: RE: [newtech-1] Uncovering Spoken Phrases in Encrypted Voice
over IP
Hey Support,
I am planing to implement new page system with asterisk 1.8 we have 200 SIP
calls and page() will overkill my system if announce in one shot. so i am
planing to record and play page over 50...50...50 chunk..
I am planing to do with .call file for auto calling after record
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, March 15, 2011 1:06 PM
To: asterisk-users
Subject: [asterisk-users] call file for page auto-call
Hey Support,
I am planing to implement new page
Hi Jason,
well, I was thinking something like this, but don't hurt to ask :D
Thank you for all guys.
Best regards,
Fellipe
Date: Tue, 15 Mar 2011 12:37:05 -0500
From: jpar...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Some errors
On 03/15/2011 12:34 PM,
Thanks for you input but how to do SIPAddHeader(Alert-Info: Ring Answer) for
auto answer my polycom phones and how to create group in .call file I am
reading at http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out but
didn't found anything related group calling. may be i am missing
Hello guys,
one more question, if I have the following dialplan and can't use _. how can I
send everything that isn't 1620,1622,1610,9XXX to SIP/xxx?
Sorry I'm new with this * world.
root@*:/etc/asterisk# vim extensions.conf
[example]
exten = _.,1,Dial(SIP/xxx/${EXTEN},60)
exten =
Hello guys!
I've solved this question just adding X in dialplan and killing that h option,
and of course with your help:
root@*:/etc/asterisk# vim extensions.conf
[example]
exten = _X.,1,Dial(SIP/xxx/${EXTEN},60)
exten = _X.,n,Hangup()
exten = _1620,1,Dial(SIP/${EXTEN},60)
exten =
Hi All,
I am doing auto answering call from manager but it seems not working any idea ?
following commands i am passing to my manager. My phone only ringing not
answering we have asterisk 1.8
Action: Originate
Channel: SIP/7527
Context: all-page
Priority: 1
Variable: SIPAddHeader
Value:
Variable: SIPADDHEADER=Alert-Info: Ring Answer
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 20:59:23 +
Subject: [asterisk-users] Auto Answer in manager
Hi All,
I am doing auto answering call from manager but it seems not working any idea ?
Hi there,
i called one asterisk server from another asterisk server. The calling
server played back a audio data und the answering server recorded the audio
sample using record() function.
I tried both ISDN, VoIP connections. Camparing with the original audio data,
the recorded samples from both
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Tuesday, March 15, 2011 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] signal amplified by asterisk
Hi there,
how can I correct it to use the right Microsoft PCM wav? Where can I find
the File foo?
Thank you!
best regards
Felix
2011/3/15 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Hi,
With Asterisk 1.8.3, I can't figure out how to pass an argument to a macro
which is used within an originate command.
Here is my sample dialplan to illustrate:
exten = 123,1,Answer()
exten = 123,n,Originate(SIP/20,app,Macro,foo,bar)
exten = 123,n,NoOp(This is the NoOp after the originate
#1 look for wav49 and change it to wav
#2 foo is a generic term - file foo = file whatever the name is.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Tuesday, March 15, 2011 4:29 PM
To: Asterisk Users
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Hopkins
Sent: Tuesday, March 15, 2011 4:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Passing an argument to a macro within an
Originatecommand
Could you please explain further more? Where should I look for wav49?
Sorry for that foolish question. But I really don't understand.
2011/3/15 Danny Nicholas da...@debsinc.com
#1 look for wav49 and change it to wav
#2 “foo” is a generic term – file “foo” = file “whatever the name is”.
Hello Bobola, thanks for your response.
So, I'm using Euroisdn with a Siemens HiPath 3750 and Qsig with Siemens
HiPath 4000.
Because we don't need to facility enable in this case (HiPath 3750) just
ANI interchange between user's, ok?
In another response I was send to you a configurations sample
On 11-03-15 04:18 PM, Fellipe Paes wrote:
I've solved this question just adding X in dialplan and killing that h option,
and of course with your help:
Since you are just learning Asterisk, I would *highly* recommend not
using 'exten = _X.'; this is a bad practice. Taking the time to
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New to Asterisk? Join us for a live
Helo folks.
I am new with asterisk, so please be patient with me!
First of all, Where i am working we gonna try an pilot project to see if we
are able to change our all PABX and VOIP system to an open source. We
atually are using an proprietary one.
We have a big envoriment. So we need multiple
I checked the recorded files. They were recorded in MS PCM wav format with
rate128kbs. Has anyone another idea?
2011/3/15 Danny Nicholas da...@debsinc.com
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*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On
On 03/15/2011 04:18 AM, Nikhil wrote:
how to send SIP HOLD Invite from asterisk to other sip client/server.?
Asterisk's chan_sip does not yet have the ability to *send* 'hold'
re-INVITEs.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com |
On 03/10/2011 03:42 PM, Benny Amorsen wrote:
--[ UxBoD ]--ux...@splatnix.net writes:
Hi,
With Asterisk 1.8 is it now possible to register the same SIP account at
multiple endpoints and for both to ring when the associated extension is
dialed ?
No. Our solution is to give each phone an
On 11-03-15 05:52 PM, deeps backup wrote:
Does your migration plan include testing asterisk 1.8 before moving it
into production?
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
--
On 11-03-15 06:19 PM, Henrique Fernandes wrote:
Have many diferenet locations that have convencional phones that need to
call others locations with convencional phones. And we can not change this,
I was reading and asterisk cannot handle it self this kind of setup, it
needs an separated serrver
[]'sf.rique
On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-03-15 06:19 PM, Henrique Fernandes wrote:
Have many diferenet locations that have convencional phones that need to
call others locations with convencional phones. And we can not change
this,
I was
ok..that means I have to modify chan_sip . I wondering why this is not
available in asterisk.
Thanks
Nikhil.
On 03/16/2011 04:39 AM, Kevin P. Fleming wrote:
On 03/15/2011 04:18 AM, Nikhil wrote:
how to send SIP HOLD Invite from asterisk to other sip client/server.?
Asterisk's chan_sip
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