On Jan 31, 2008 12:30 AM, John Von Essen [EMAIL PROTECTED] wrote:
Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:
Maybe the SIP config is wrong?
Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
goes silent.
Can you places
Hello,
We use Asterisk Realtime for our billing software. 200+ installations of
Asterisk with Realtime, but I see this for the first time.
Asterisk 1.4.17, Addons 1.4.5, No patches, no NAT - just plain simple
installation.
With debug I can see:
[Jan 30 22:38:21] DEBUG[27885]:
Hi,
I installed Asterisk, asterisk-addons, pwlib, h323plus,opal and gnugk.
I am searching for /etc/asterisk/ooh323.conf. It is not there.
Can anybody please tell me how to get ooh323.conf.
Thanking you,
Regards,
Preeta
Please do not print this email unless it is absolutely necessary. Spread
Hi,
Has anyone heard of SIP phones supporting LLDP-Med ?
Mitel or Avaya phones are supposed to support it but I don't if it applies
to SIP firmware enabled hardphones or not.
Regards
___
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Hi,
I have asterisk register two users (client-1, client-2) with a SIP proxy.
I have the same two SIP client registered with asterisk. Now my dial plan
setup is such that any call from client-1/client-2 is forwarded to the SIP
proxy and the SIP proxy then takes the routing decision. Calls
Hi, preeta.
ppwc Hi,
ppwc I installed Asterisk, asterisk-addons, pwlib, h323plus,opal and gnugk.
ppwc I am searching for /etc/asterisk/ooh323.conf. It is not there.
ppwc Can anybody please tell me how to get ooh323.conf.
In source of asterisk-addons there is a file
Hi,
I am moving my call center to 1.4. Previously I was recording calls in
agents.conf with the following config
recordagentcalls=yes
recordformat=wav
createlink=yes
So I had the filename in all calls which was *connected to agents*. I
am looking for a similar functionality for 1.4.
I am now
I have a very serious problem with calls between PAP2-NA and a TDM2400 (8
FXO). Almost every call dropped after between 20 and 30 seconds with
conversation.
I disable the sound card, serial and other things on my server, but the
problem still continues. I've changed the RPT Packet Size to .20 on
On Jan 31, 2008 6:45 AM, mccoy silva [EMAIL PROTECTED] wrote:
I have a very serious problem with calls between PAP2-NA and a TDM2400 (8
FXO). Almost every call dropped after between 20 and 30 seconds with
conversation.
I disable the sound card, serial and other things on my server, but the
pbx*CLI show application ParkAndAnnounce
-= Info about application 'ParkAndAnnounce' =-
[Synopsis]
Park and Announce
[Description]
ParkAndAnnounce(announce:template|timeout|dial|[return_context]):
Park a call into the parkinglot and announce the call over the console.
announce template:
First time or second time they hit transfer?
Dial plan config?
2008/1/30 Don Smith [EMAIL PROTECTED]:
Greetings,
I have an issue with the length of time that passes from when someone hits
the transfer soft key on a Cisco 7940, after doing an attended transfer, and
when the recipient's
On 1/31/08, Rajkumar S [EMAIL PROTECTED] wrote:
Hi,
I am moving my call center to 1.4. Previously I was recording calls in
agents.conf with the following config
recordagentcalls=yes
recordformat=wav
createlink=yes
So I had the filename in all calls which was *connected to agents*. I
am
John Von Essen wrote:
Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:
exten = 1000,3,VoicemailMain,s6000
It may be your syntax, try :-
exten = 1000,3,VoicemailMain(6000|s)
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
Try this:
exten = 1000,1,Answer()
exten = 1000,2,Wait(2)
exten = 1000,3,VoiceMailMain()
You do not specify the mailbox number in the call to the application.
You only specify the number to VoiceMail()
HTH,
Shane
On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote:
John Von Essen wrote:
Any ideas
It is my understanding that the cast majority of the compatibility
issues went away with the recent chipset change on the digium cards.
Soa compatibility list really isn't needed.
I've run the digium cards on all manner of Dell hardware (from
old-school desktops all the way to the high end
I have a friend with a small business running a small SIP based phone
system. He was looking into providing some SIP phones for a couple of
remote teleworkers, but as he started to look around and ask me questions he
ran across analog adapters which made him curious.
He proceeded to ask me if
Shane D wrote:
Try this:
exten = 1000,1,Answer()
exten = 1000,2,Wait(2)
exten = 1000,3,VoiceMailMain()
You do not specify the mailbox number in the call to the application.
You only specify the number to VoiceMail()
HTH,
Shane
On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote:
John
Okay, What I ment was you don't have to.
On 1/31/08, John Millican [EMAIL PROTECTED] wrote:
Shane D wrote:
Try this:
exten = 1000,1,Answer()
exten = 1000,2,Wait(2)
exten = 1000,3,VoiceMailMain()
You do not specify the mailbox number in the call to the application.
You only specify
Here are my configs:
sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000
context=default
extensions.conf:
[default]
exten = 1000,1,Ringing
exten = 1000,2,Wait(2)
exten =
We are currently using Asterisk 1.4.9 with Unicall.
We are experiencing an issue with hint hanging taking the extensions out
of action until an asterisk restart. Details on this can be found at:
http://bugs.digium.com/view.php?id=10474
We would like to upgrade Asterisk to 1.4.17 but are
On 1/31/08, Prashant Sharma [EMAIL PROTECTED] wrote:
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got executed on
pressing a particular DTMF key sequence.
As I want to execute my own
A call comes in from the PSTN, Asterisk answers it, it goes to the directory,
and then to the extension the caller designates and the user at that extension
answers. The user at the extension then wants to transfer the call to another
extension; on the Cisco 7940 they push the “more” soft key,
On Jan 30, 2008 10:35 PM, Dan Austin [EMAIL PROTECTED] wrote:
Franklin wrote:
ztdummy can give you issues as a timing device.
Yes and no. See below
Any way you could try using a Digium card just
as a timing device to see if this helps?
Tomasz wrote:
I am using Debian OS kernel
Our guest is tomorrow Mobeen Khan is Chief Operating Officer of
Metaphor Solutions who offer Plug Play IVR On-Demand
http://www.metaphorivr.com
Instructions to join the conference: http://VoipUsersConference.org
IRC: freenode.net #voip-users-conference
The weekly Friday Noon VoIP Users
Thanks. I am getting a dual 3.0Ghz 2950 III.
On 1/31/08, Erik Anderson [EMAIL PROTECTED] wrote:
It is my understanding that the cast majority of the compatibility
issues went away with the recent chipset change on the digium cards.
Soa compatibility list really isn't needed.
I've run
I recently moved an installed and working Asterisk system from one PC to
another. I moved two Digium TDMXX cards and the OS as well (a live
distro). I tuned the hardware on the new PC, but for some reason analog
calls periodically have some electronic noise. It's like beeps, but more
musical.
On Jan 30, 2008 5:48 PM, Matthew J. Roth [EMAIL PROTECTED] wrote:
Tomasz Zieleniewski wrote:
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the
On Jan 31, 2008 12:28 PM, Matthew Yingling [EMAIL PROTECTED] wrote:
I recently moved an installed and working Asterisk system from one PC to
another. I moved two Digium TDMXX cards and the OS as well (a live
distro). I tuned the hardware on the new PC, but for some reason analog
calls
John Von Essen wrote:
Here are my configs:
[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000
I believe you need to include a context on your mailbox line, such as
[EMAIL PROTECTED]
Doug
--
Ben Franklin quote:
Those who would give up Essential
Very odd. Could you try taking the mailbox line out of sip.conf and
see what happens?
On 1/31/08, John Von Essen [EMAIL PROTECTED] wrote:
Here are my configs:
sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
[6000]
type=friend
Howdy,
Excuse the neophyte questions... I was wondering:
(1) what's involved in setting up a call with encrypted media (I'm on a
cable network and don't want my calls snooped);
(2) is there a cheat-sheet for configuring Sipura handsets/hardphones
like the SPA-942, and in particular for
Uhhh... just
export HOSTNAME
should be enough once it's been set.
Joost Kuif | Mobillion wrote:
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
List users,
A recent post on MeetMe timing mentioned the internal_timing option,
which can be configured to have Asterisk asynchronously generate
outgoing RTP when a timing device (ie. ztdummy) is available. This
allows Asterisk to produce outgoing audio in situations where no
incoming audio
Tomasz Zieleniewski wrote:
ztttest results show value below 99,98:
[EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5
snip
--- Results after 11 passes ---
Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference:
49.931827
This is the first thing I would address. Get that
I second that. IRQ issues are more than likely causing the problem. Check your
interrupts and see if your TDM cards are sharing IRQs with any other devices.
From past experience, I know we would get the same behavior when an analog card
was sharing an IRQ with a storage controller. Any amount
John Von Essen wrote:
Here are my configs:
sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000
context=default
extensions.conf:
[default]
exten =
On Thu, 2008-01-31 at 10:45 -0800, Philip Prindeville wrote:
(1) what's involved in setting up a call with encrypted media (I'm on a
cable network and don't want my calls snooped);
Using IAX, it's pretty simple. See
http://www.voip-info.org/wiki/view/IAX+encryption
(2) is there a
You are probably both correct. I noticed that both of our TDM cards, and
the Ethernet card are all sharing the same IRQ. Since we do VOIP
internally and analog externally, that IRQ is getting hit twice for any
outbound or inbound calls. The system is new, and the OS supports ACPI, so
I'm not
Purchased a Sangoma board for a company that went defunct on 11/5/2007.
Will accept $500 or best offer. Note this board does have echo
cancelation in hardware. Will provide a copy of the receipt.
Details
A20101D Sangoma Board 2 Port FXO 2 Port FXS w/ Echo Can
20032D0-02679
FXS-03854
FXO-11991
(2) is there a cheat-sheet for configuring Sipura handsets/hardphones
like the SPA-942, and in particular for message-waiting indicator and
shared-line appearances?
MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf
for the phone
Make sure also to add voice mail number
Using IAX, it's pretty simple. See
http://www.voip-info.org/wiki/view/IAX+encryption
Jared, perhaps you could clarify something on that voip-info.org article. It
claims that encryption only works for auth=md5. Does that mean that using a
public/private key for authentication (auth=rsa) will
Hi,
I have configured my SNOM 360 to monitor another extension by setting
the following:
[default]
exten = user1,hint,SIP/user1
The next step was to define a function key on the phone as an extension
with the value sip:[EMAIL PROTECTED] and later with
sip:[EMAIL PROTECTED]|*8
When someone
Hi,
can anyone tell me how i do a sip trunk between an asterisk and a alcatel
omnipcx pbx with sip support
tx,
Pedro Santos
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asterisk-users mailing list
To UNSUBSCRIBE or update
Chris Bagnall wrote:
(2) is there a cheat-sheet for configuring Sipura handsets/hardphones
like the SPA-942, and in particular for message-waiting indicator and
shared-line appearances?
MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf
for the phone
Make sure
- Original Message
From: Mayur [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, 31 January, 2008 9:59:42 AM
Subject: [asterisk-users] Incoming call from SIP proxy to asterisk
Hi,
I have asterisk register two users (client-1, client-2) with a SIP
- Original Message
From: Don Smith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, 31 January, 2008 4:46:27 PM
Subject: Re: [asterisk-users] Default delay time for Attended call
A call comes in from the PSTN, Asterisk answers it, it goes to the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
d4rk f1br wrote:
He however is wanting something that connects using both SIP to the
server and PSTN. But his request does not stop there. He wants to be
able to choose on the fly which SIP or PSTN connection he utilizes for
any given outbound
Asterisk 1.4.18-rc4 is now available.
This release candidate includes an important fix for a regression related to the
use of codec_g729 that caused decoders to not get properly released. Additional
fixes added today that are included in this release candidate include:
- fixes for some locking
The snippet is asterisk telling you I'm just letting you know that the
correct caller id for Channel: SIP/103-098500d8 is CallerID: 103
This is absolutely correct, it's just not a piece of information you
expected to be receiving at that point.
You probably also received a packet like that
I've struggled with this recently. In short:
- Observed behaviour is expected as of asterisk 1.2 and later,
as previously described by Mojo
- If you want to get the caller id for the channel calling (dialling)
into that channel for that specific Newstate: Ringing event, you
Barry L. Kline wrote:
He however is wanting something that connects using both SIP to the
server and PSTN. But his request does not stop there. He wants to be
able to choose on the fly which SIP or PSTN connection he utilizes for
any given outbound call the user makes. Basically, analog
Hi,
The server log shows the following message.
[Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
Does it mean the server failed to file the mysql server? I have
installed mysql and both asterisk and mysql are
Johansson Olle E wrote:
In my series of articles about Asterisk 1.4, I've now arrived to the
new jitter buffer
that enhances voice quality for those of you using Asterisk as a PSTN
gateway.
Please read
http://www.voip-forum.com/category/asterisk/asterisk14/
I wrote a patch that lets
Rilawich Ango wrote:
Hi,
The server log shows the following message.
[Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
Does it mean the server failed to file the mysql server? I have
installed mysql and
I need to carry a variable over into the 'h' priority - so I can go back
and clean up DB entries in a mysql database (time of call and so on)
I tried using UNIQUEID but it seems that 'h' generates a new one.
Anyone have any ideas? What can I use to carry a variable over into
'h'??
Sorry for taking so long to reply,
This email got lost in translation, again.
Ian
Ian said the following on 30-Jan-08 03:57 PM
Thaks for the speedy reply
Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
Hi all
I have a small
Hi all,
I have installed Asterisk-addons-1.4.5. I was getting error
cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory
So, I did following steps:
cp asterisk-ooh323c/.libs/libchan_h323.1.0.1
asterisk-ooh323c/.libs/libchan_h323.so.1.0.1
make install
make samples
It
I have an SPA3000 and it works really great !!!
It can do more than you say but "Per
Call Authentication and Associated Routing", I dont understand
what you mean.
About your example with "press 8 ..." there are more eficient
scenarios. You can can create a dialplan that automatically selects
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