i am working on call parking, i had made three extensions,
112
113
114
call parking range 8100-8199
now any call comes, exten 112 receives it after receiving it called party i.e
who received the call puts the caller on hold n than called party is hearing
moh. now plz tell me how exten 113
Ok but this is available today and works fine, so it can be used as a zero
day replacement. Any syntax change is welcome but will take time until it
gets in a public release and does not save you the hassle to change the
dialplans anyway - unless you implement it as a default behaviour at the SIP
While we continue discussing all possible solutions to this and build an
expanding knowledgebase, I would like to repeat myself and kindly ask everyone
that blogs, twitters, talks and teaches about Asterisk to please spread the
word and the links. Later today, there will be an official Asterisk
Hi,
when call is Hangup, Asterisk send X-Asterisk-HangupCauseCode in Bye
packet.
i want to remove Asterisk keyword from this string
X-Asterisk-HangupCauseCode.
please tell how i can remove Asterisk from above string at call hangup time.
--
Regards,
M. Asif Raza
--
17 feb 2010 kl. 11.13 skrev Mian Asif:
Hi,
when call is Hangup, Asterisk send X-Asterisk-HangupCauseCode in Bye packet.
i want to remove Asterisk keyword from this string
X-Asterisk-HangupCauseCode.
please tell how i can remove Asterisk from above string at call hangup time.
You need to
On 17 Feb 2010, at 10:13, Mian Asif wrote:
when call is Hangup, Asterisk send X-Asterisk-HangupCauseCode in
Bye packet.
i want to remove Asterisk keyword from this string X-Asterisk-
HangupCauseCode.
please tell how i can remove Asterisk from above string at call
hangup time.
Find it
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized
prilocaldialplan NPI modifier: k
[Feb 17 12:33:03]
David @ULC wrote:
I use IdeaSip with IPKall.
How may channels are open when we use IdeaSip ?
Incoming IdeaSIP SIP channels are unlimited; however, I believe IPKall
limits you to 94 channels via their DIDs.
You would, of course, need the bandwidth to be able to handle 94
simultaneous
17 feb 2010 kl. 12.37 skrev Håkon Nessjøen:
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized
On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call:
17 feb 2010 kl. 14.00 skrev Tzafrir Cohen:
On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03]
On Wed, Feb 17, 2010 at 2:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
Legal modifiers for the pridialplan are U, I, N, L, S, V, and R. See
chan_dahdi.conf.sample.
My guess: you have:
prilocaldialplan = unknown
Hi,
On Wed, Feb 17, 2010 at 02:23:25PM +0100, Håkon Nessjøen wrote:
On Wed, Feb 17, 2010 at 2:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
Legal modifiers for the pridialplan are U, I, N, L, S, V, and R. See
Lenz Emilitri escribió:
Ok but this is available today and works fine, so it can be used as a
zero day replacement. Any syntax change is welcome but will take time
until it gets in a public release and does not save you the hassle to
change the dialplans anyway - unless you implement it
In my asterisk setup, 112 would transfer the call to 8100 and get a message
back that the call was set to lot 8100 or another value up to 8199. 112
would then tell 113 to pickup 81xx and they would have 2 minutes to do so.
Regards,
--
Danny Nicholas
--
_
From:
Hello One and All,
I am a Linux admin, new to asterisk. I have been assigned the task of
setting up a dictation server for the company I work for. Our company is
into transcription. Currently we are using dictation server, which is
provided by another company. Now we have decided to have our own
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API action to originate a call:
Action: originate
Priority: 1
Context: trunk
Callerid: 100
Channel: Local/1...@callback/n
Exten: 123456789
There was a bug reported on this, I think ... yes #16581
Fixed in
r244070 | tilghman | 2010-02-01 11:46:32 -0600 (Mon, 01 Feb 2010)
Julian
On 17 February 2010 15:00, James Northcott / Chief Systems
ja...@chiefsystems.ca wrote:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and
17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API action to originate a call:
Action: originate
Priority: 1
Context:
17 feb 2010 kl. 16.32 skrev Olle E. Johansson:
17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API action to originate
On Wednesday 17 February 2010 07:00:26 Tzafrir Cohen wrote:
On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
On Wednesday 17 February 2010 09:32:50 Olle E. Johansson wrote:
17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API
James Northcott / Chief Systems escribió:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
Hi,
This is a known and solved bug:
https://issues.asterisk.org/view.php?id=16717
Give the latest 1.4.30-rc2 a try.
Cheers,
On 16 February 2010 19:51, Danny Dias ing.diasda...@gmail.com wrote:
Hello My friends,
Today my asterisk stop working and i could see the following messags in
/var/log/asterisk/messages at the time that asterisk stop working:
[Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now
Hi,
I run into a problem and I'm not shure what do I misconfigure. I've a
B410P ISDN card with bri_cpe signalling and two Openvox (A1200, A800)
cards with fxo_ks signalling, all with dahdi drivers. I can receive fax
from a public number, but I can't send fax. The CLI says it picks up the
line
Dialing DAHDI/21 will open DAHDI/21 as a line expecting further DTMF to dial
You need DAHDI/21/1234567
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter
Gelencser
Sent: Wednesday, February 17, 2010 10:11
That's what I've started doing.
Thanks,
--Warren Selby
On Feb 17, 2010, at 8:29 AM, Miguel Molina mmol...@millenium.com.co
wrote:
Lenz Emilitri escribió:
Ok but this is available today and works fine, so it can be used as a
zero day replacement. Any syntax change is welcome but will
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Hello,
We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of
voicemail you can press 3 for advanced options, 5 to leave a message and
enter an extension to leave a voicemail. This feature worked fine under 1.4.
Now under 1.6.1 all the prompts are the same but when you enter
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
According to my experimentation, Polycom VVX1500 phones work with
all versions of Asterisk as far
We use Asterisk 1.6.2 with the latest Polycom firmwares on the VVXs with no
problem.
They key is the new bootblock polycom released a little while back.
If you download the new BootBlock, BootROM and SIP Firmware from
http://www.polycom.eu/support/voice/business_media_phones/vvx1500.html it
On Wed, Feb 17, 2010 at 2:54 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
So which version of Asterisk is it and what do you have in
/etc/asterisk/chan_dahdi.conf ?
Running Asterisk 1.6.1 now.
But i'm pretty sure I saw the same in Asterisk 1.6.0 too.
-- snip -- chan_dahdi.conf -- snip --
On Wed, Feb 17, 2010 at 4:43 PM, Tilghman Lesher tles...@digium.com wrote:
Right diagnosis, wrong location. Those letters are used for modifying the
localdialplan at Dial time, as Dial(DAHDI/g1/N2125551212). Sounds like the
OP has letters in his dialstring, where the number ought to be.
Thanks I loaded the new firmware and Bootblock yesterday, but still the video
is not working. Maybe I have something misconfigured..
Doug
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jordan Kirby
- Steve Underwood ste...@coppice.org escreveu:
Hi Vinícius,
Don't post big things, like wireshark traces, to a mailing list. They
are likely to ban you.
The first two calls in your wireshark log decode to the attached
images.
There were no lost packets. The wireshark logs
On Wednesday 17 February 2010 11:32:04 Håkon Nessjøen wrote:
On Wed, Feb 17, 2010 at 4:43 PM, Tilghman Lesher tles...@digium.com wrote:
Right diagnosis, wrong location. Those letters are used for modifying
the localdialplan at Dial time, as Dial(DAHDI/g1/N2125551212). Sounds
like the OP
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf) files on the same asterisk server and with one set
On Wed, 2010-02-17 at 10:51 -0500, Miguel Molina wrote:
James Northcott / Chief Systems escribió:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
Hi,
This is a known and solved bug:
Hi,
Right now I have two machines and each one runs Asterisk and Openser. Both
machines have a MySql database where everything is stored and is replicated
using MySql Cluster. I would like to know how to setup an Active/Active
Asterisk system. Right now, Asterisk save changes to the spool
I have Audiocodes MP-114 gateway and it is registered per end-point
in sip.conf:
[pstn-]
type=friend
context=incoming
...
[pstn-9998]
type=friend
context=fax-incoming
all calls that comes IN are going, regardless of the port are going to context
fax-incoming even if call comes IN on
- Steve Underwood ste...@coppice.org escreveu:
Hi Vinícius,
Don't post big things, like wireshark traces, to a mailing list. They
are likely to ban you.
The first two calls in your wireshark log decode to the attached
images.
There were no lost packets. The wireshark logs
- Vinícius Fontes vinic...@canall.com.br escreveu:
- Steve Underwood ste...@coppice.org escreveu:
Hi Vinícius,
Don't post big things, like wireshark traces, to a mailing list.
They
are likely to ban you.
The first two calls in your wireshark log decode to the attached
hi,
i have solaris 9, the kind that runs on a pc. i tried downloading
gcc-3.3.2-sol9-intel-local.gz; gcc-3.4.6-sol9-x86-local.gz and
gcc_small-3.3.2-sol9-intel-local.gz. none of them decompressed properly. i
tried it a few times. i downloaded from sunfreeware.com.
any suggestions?
On Wed, 17 Feb 2010, Givon Zirkind wrote:
i have solaris 9, the kind that runs on a pc. i tried downloading
gcc-3.3.2-sol9-intel-local.gz; gcc-3.4.6-sol9-x86-local.gz and
gcc_small-3.3.2-sol9-intel-local.gz. none of them decompressed
properly. i tried it a few times. i downloaded from
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet,
and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet
parameters are all set correctly in sip.conf. An inbound call from Sipphone
works great until the local channel places the call on hold. During
Brent Torrenga wrote:
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the
localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost
and localnet parameters are all set correctly in sip.conf. An inbound
call from Sipphone works great until the local channel places
All,
I am trying to set a monitor file from the queue.conf as specified on
http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf In order to
avoid the default MONITOR_FILENAME format wich is:
agent-x-uniqueid.wav for example agent-10017-1266438575-26.wav
As you may now, when using
I need to extract the event header info from an incoming SIP call. Is
this accessible from within the dialplan?
I've reviewed RFC 3265 but I'd like to start with just dumping everything to
do with event (if accessible, in other words Asterisk doesn't strip this
away)
Thanks!
MD
--
A little bit of a strange request.
Basically I want all calls that go to one user go to voicemail
immediately if the user is on the phone. The user is using the Linsys
SPA941, and even though he can be on the phone, calls will still ring
his phone. I tried disabling the rest of the lines on the
I dont have a Static IP.
How can I ask IPKall to send call to my Asterisk ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Set his call-limit to 1 in users.conf. Other than that, you could check the
channel before dialing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike A.
Leonetti
Sent: Wednesday, February 17, 2010 3:07 PM
I am running Trixbox PRO.
I don’t know if this is a config issue, since it would seem to be odd that
an inbound SIP call into asterisk would answer the call even during ringing.
Check out the SIP trace below.
It’s a call from the PSTN into an asterisk DID assigned to an ext.
On the
Ok
I can use
Dyndns.org
I registered myself.
easy.selfip.com
https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com
successfully activated.
Hostname
https://www.dyndns.com/account/services/hosts/?field=fqdnsort=dService
Looks like IdeaSip need STATIC ip else it doesnt work.
.
On Thu, Feb 18, 2010 at 3:02 AM, David @ULC ucoms2...@gmail.com wrote:
Ok
I can use
Dyndns.org
I registered myself.
easy.selfip.com
https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com
successfully activated.
Perfect. Thanks.
Mike A. Leonetti
As warm as green tea
Evolution CE
3468C Lawson Boulevard
Oceanside, NY 11572
www.evolutionce.com
516-536-5006 ext 105
516-208-4679 (Direct)
Danny Nicholas wrote:
Set his call-limit to 1 in users.conf. Other than that, you could check the
channel before
On Thu, 18 Feb 2010, David @ULC wrote:
I dont have a Static IP.
On Thu, 18 Feb 2010, David @ULC wrote:
Ok
I can use
Dyndns.org
I registered myself.
Congratulation. Doesn't it feel great to help yourself rather than
bothering the mailing list with questions that have nothing to do with
Michelle Dupuis wrote:
*I need to extract the event header info from an incoming SIP call.
Is this accessible from within the dialplan?*
**
*I've reviewed RFC 3265 but I'd like to start with just dumping
everything to do with event (if accessible, in other words Asterisk
doesn't strip this
Wednesday, February 17, 2010, 10:40:18 PM, Steve wrote:
Congratulation. Doesn't it feel great to help yourself rather than
bothering the mailing list with questions that have nothing to do with
Asterisk? And it only took you 17 minutes!
Much better than cool dude :)
--
Best regards,
Gergo
Is it possible to just send an event from one Asterisk server to another?
(Perhaps some custom event that I could define?) Or would that break the SIP
protocol/handling in asterisk?
Aside from SUBSCRIBE, anything else use events packages?
Thanks,
MD
-Original Message-
From:
Wednesday, February 17, 2010, 10:40:18 PM, Steve wrote:
Congratulation. Doesn't it feel great to help yourself rather than
bothering the mailing list with questions that have nothing to do with
Asterisk? And it only took you 17 minutes!
On Wed, 17 Feb 2010, Gergo Csibra wrote:
Much better
hmmm Ok..
Is this a Asterisk Question ?
I have a setting as :
Global Settings :
---
SipIpkall = SIP/fwd
Dialplan Entry :
exten = 11012012600,1,Ringing call ringing
exten = 11012012600,2,Wait(1) Wait 1 second for CID delivery
On Wed, 17 Feb 2010 17:12:01 +, Steve Davies wrote:
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
According to my experimentation, Polycom
Feb 17 19:19:04 NOTICE[2554]: chan_sip.c:10077 handle_response_peerpoke:
Peer '11012012600' is now TOO LAGGED! (2567ms / 2000ms)
On Thu, Feb 18, 2010 at 5:34 AM, David @ULC ucoms2...@gmail.com wrote:
hmmm Ok..
Is this a Asterisk Question ?
I have a setting as :
Global Settings :
On 15/02/10 11:55 PM, Emre Kurnaz wrote:
Hi all,
Now a days we are planning to run two asterisk boxes on XEN with DNS
Failover. But even using the default configuration asterisk shuts itself down
at least 5 times in a day with an exit status of 139 (i think it should be
139-128=11 there
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, all!
I'm being based on this document [1] to send and to receive calls using
ekiga.net. But I'm seeing, in an Asterisk console, several messages of
this type:
[Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout:--
Registration
On Thu, 18 Feb 2010, David @ULC wrote:
Is this a Asterisk Question ?
[snip]
register
=11012012600:passw...@66.54.140.4611012012600%3apassw...@66.54.140.46
Nope.
ipkall.com does not accept registration. If you had googled for asterisk
ipkall registration you would have had your answer an
Is there any asterisk guru who can explain me how how asterisk knows which
context forward the call to?
--
Joseph
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
So, this will change :
register = 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
insecure=very
fromdomain=proxy.ideasip.com
exten = _1101XXX,1,SetCallerID(Your Name 11012012600)
exten =
http://i50.tinypic.com/120rwya.jpg
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC ucoms2...@gmail.com wrote:
So, this will change :
register = 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
Feb 17 20:42:17 NOTICE[2746]: chan_sip.c:5529 sip_reg_timeout:--
Registration for '11012012...@proxy.ideasip.com' timed out, trying again
(Attempt #119)
-- parse_srv: SRV mapped to host proxy.ideasip.com, port 5060
-- Got SIP response 479 Please don't use private IP addresses back
from
Hi,
Right now I have two machines and each one runs Asterisk and Openser. Both
machines have a MySql database where everything is stored and is replicated
using MySql Cluster. I would like to know how to setup an Active/Active
Asterisk system. Right now, Asterisk save changes to the spool
Your question is a little vague. I assume that you would be looking for the
GoTo application. The syntax is explained here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+goto
http://www.voip-info.org/wiki/view/Asterisk+cmd+gotoAlso, you can look on
page 426 of the Asterisk book, which is really
On Wed, Feb 17, 2010 at 2:47 PM, Mariano Lecuona mlecu...@gmail.com wrote:
Could anyone get the MONITOR_FILENAME set from the queue.conf with
variables like:
MEMBERINTERFACE is the interface name (eg. Agent/1234)
MEMBERNAME is the member name (eg. Joe Soap)
MEMBERCALLS is the number of
Apology for not posting too much details.
I'm trying to figure it out how the ATA adapter knows which context (from
sip.conf) send the call to?
I'm puzzled as I have never encounter this problem before.
I have for example two ATA adapters (Linksys and Audiocodes) both register with
asterisk
On Wed, Feb 17, 2010 at 6:53 PM, Daniel Bareiro daniel-lis...@gmx.netwrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost instead of
externip?
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
On Wed, Feb 17, 2010 at 8:59 PM, Joseph syscon...@gmail.com wrote:
Apology for not posting too much details.
I'm trying to figure it out how the ATA adapter knows which context (from
sip.conf) send the call to?
I'm puzzled as I have never encounter this problem before.
I have for example
On 02/17/10 21:09, Warren Selby wrote:
On Wed, Feb 17, 2010 at 8:59 PM, Joseph syscon...@gmail.com wrote:
Apology for not posting too much details.
I'm trying to figure it out how the ATA adapter knows which context (from
sip.conf) send the call to?
I'm puzzled as I have never encounter this
On Wed, Feb 17, 2010 at 8:50 AM, Dave Poirier dpoir...@mesd.k12.or.us wrote:
Hello,
We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of
voicemail you can press 3 for advanced options, 5 to leave a message and
enter an extension to leave a voicemail. This feature worked
Hi,
I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The
setup consists of the following components:
- A Digium TE121 for connectiong to E1 ISDN
- Debian box with Asterisk 1.4
- Grandstream GXW-4008 SIP ATA to which the Fax machines connect
I am aware of the problems with
Hi,
When Jason Goecke talks, VoIP ideas become reality, and this makes my
day. On this call we’ll talk about the newest features in Tropo and
how to get started with telephony apps in the cloud without adding new
infrastructure. Here's a chance to speak directly to Jason (or JSON as
we now call
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