Hello
I want to know that how apps/enter.h data can be generated...
I want to do same for conf-muted / conf-unmuted but not getting idea how
data is generated for muted/unmuted same like apps/enter.h
Help me out...
--
Regards,
Chandrakant Solanki
On 1.6.1
Check out 'core show function QUEUE_MEMBER'
Don't have a 1.6.0 box anymore to check.
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Clark
Sent: Tuesday, 15 December 2009 12:59 p.m.
To:
I'm having a strange problem with a sip client and 2 asterisk servers
connected together with a sip trunk. Here's a rough layout
sip_client -- Asterisk A -[sip trunk] -- Asterisk B
when the sip client tries to dial an extension on Asterisk B, Asterisk
A sends the invite to B using
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote:
IAXDIAL is free on app store works great on WiFi even true NATs but seem
blocked for GPRS.
ta
HB
[snip]
Well I have a 3gs - will tell you how that goes.
installed (non cracked), but I am on wifi now, easy to configure and
I thought so- the fact the server has 20 different registry entries to 20
different account all at the same ITSP shouldn't matter?
Can't see any DDI info in the SIP headers unfortunately :(
John
2009/12/14 meetmecall i...@meetmecall.nl
The easiest solution to deal with this is to have one
On Tue, Dec 15, 2009 at 09:14:16PM +1100, Alex Samad wrote:
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote:
[snip]
My only concern with it - it's not just a voip client, its many other
things as well. not sure if I want to be a fring user as well as all the
other memberships I have :)
Hello,
I could successfully played with General Purpose Parameters (GPP_A, GPP_B)
and a TFTP server : whenever I change a GPP value in a configuration file,
my SPA3102 automatically updates the corresponding value its web server
shows.
My config file is :
flat-profile
GPP_Amyid
/GPP_A
On 15 Dec 2009, at 10:42, Olivier wrote:
Unfortunately, it seems macro expansion doesn't occur in Line1 tab :
when I type $A or $(A) or ${A} or $GPP_A or $UID1 in User ID field
(in Line1 tab), asterisk receives a REGISTER message like :
handle_request_register: Registration from '${A} sip:$
Hello,
I'm using Asterisk 1.6.X version and I'm creating IVR. My question : is it
possible create CDR record , before client is exiting from contexts ? My
test dialplan is:
context Sales {
_X. = {
Ringing();
Wait(4);
Answer();
Playback(tt-monkeys);
goto Techs|${EXTEN}|1;
}
}
Some recent issues I had with hardware seem to come back to not
understanding two very similarly named files:
/etc/asterisk/dahdi-channels.conf
/etc/asterisk/chan_dahdi.conf
I've modified the chan_dahdi.conf to work now, but it would appear all I
needed to do was include dahdi-channels.conf in
I am actually deploying on a 1.6.1.6 but it does not seem to work - maybe I
am using a wrong syntax?.
pbx-ch*CLI core show version
Asterisk 1.6.1.6 built by root @ pbx-ch on a i686 running Linux on
2009-09-11 16:54:55 UTC
I see this works:
exten = 100,hint,SIP/${EXTENSION}
pbx-ch*CLI core show
Thanks that's exactly what I was looking for! I had seen a patch for it but
did not notice this was in the main trunk.
l.
2009/12/14 Stephen Davies stephen.l.dav...@gmail.com
What you are missing is the new state-interface parameter to
AddQueueMember.
You can't use functions in a hint
2009/12/15 Steve Howes steve-li...@geekinter.net
On 15 Dec 2009, at 10:42, Olivier wrote:
Unfortunately, it seems macro expansion doesn't occur in Line1 tab :
when I type $A or $(A) or ${A} or $GPP_A or $UID1 in User ID field
(in Line1 tab), asterisk receives a REGISTER message like :
Hello,
We upgraded the Asterisk to 1.6.1.11. Now, there's no RTP reINVITE, but
the datagram handling of Asterisk is strange. Basically, it takes a
commission from both ends, and ends up overflowing:
Reminder, we're dealing in this example with a passthrough, where we
have an ATA device
In that case, you're going to have to talk to your provider.
They SHOULD be able to easily send the DID with the call...
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Taylor
Sent: Tuesday, December 15, 2009 5:17 AM
To:
On 15 Dec 2009, at 13:08, Olivier wrote:
I could successfully set this value using :
User_ID_1_myid/User_ID_1_
But, I'm still fighting to set parameters from PSTN Line tab. I
tried many combinations with tags like :
User_ID_PSTN_1_ or User_ID_Line_1_
Have you looked at the SPC tool to
Forkcdr may be the thing you need. As I understand it, it does a snapshot
cdr record and continues with the call.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giedrius Augys
Sent: Tuesday, December 15, 2009 6:08 AM
To:
2009/12/15 Olivier oza-4...@myamail.com
2009/12/15 Steve Howes steve-li...@geekinter.net
On 15 Dec 2009, at 10:42, Olivier wrote:
Unfortunately, it seems macro expansion doesn't occur in Line1 tab :
when I type $A or $(A) or ${A} or $GPP_A or $UID1 in User ID field
(in Line1 tab),
Hello list.
We just upgraded to 1.6.1.11.
We are using real time information stored on mysql databases. That is all
running fine.
Now, since we upgraded, some member don't get calls from queues.
In CLI: queue show shows something like:
611 (Local/6...@agents) with penalty 20 (realtime) (*In
2009/12/15 Steve Howes steve-li...@geekinter.net
On 15 Dec 2009, at 13:08, Olivier wrote:
I could successfully set this value using :
User_ID_1_myid/User_ID_1_
But, I'm still fighting to set parameters from PSTN Line tab. I
tried many combinations with tags like :
User_ID_PSTN_1_ or
On 15 Dec 2009, at 15:39, Olivier wrote:
Steve
With SPC tool, yow would still need to input correct parameter name
to get appropriate config file.
Hi,
Generate the sample with it. It has built in help. There are numerous
instruction guides online. You can use a tool such as Google to
Cyprus VoIP wrote:
This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353
v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
girgis Rasmy wrote:
Does anyone know a sip client that can be installed on Nokia /Symbian
that register to asterisk directly , i installed Fring ,seems that the
register goes to an intermediate server on Internet that forward it to
my asterisk server .
What Nokia phone do you have? I have
Because we already have a reduntant way to tell if the member is in a call,
we turned on ringinuse. It seems to work.
The member is still show as (In use).
Would anybody help?
Thanks.
2009/12/15 Tiago Geada tiago.ge...@gmail.com
Hello list.
We just upgraded to 1.6.1.11.
We are using
Because we already have a reduntant way to tell if the member is in a call,
we turned on ringinuse. It seems to work.
The member is still show as (In use).
Would anybody help?
Thanks.
2009/12/15 Tiago Geada tiago.ge...@gmail.com
Hello list.
We just upgraded to 1.6.1.11.
We are using
How about:
exten = 977,1,ExecIf($[${CALLERID(num)} =
733025975]?Set(CALLERID(num)=0317998975))
exten = 977,n,ExecIf($[${CALLERID(num)} = 1234]?Set(CALLERID(num)=317998977))
exten = 977,n,ExecIf($[${CALLERID(num)} = 5678]?Set(CALLERID(num)=317998978))
[..]
exten = 977,n,Dial(SIP/0317998977)
On
Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
I've got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I've set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls it still shows
(ULAW) (G.711) as the codec in
On Tue, 15 Dec 2009, Ben Schorr wrote:
Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
I've got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I've set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ben Schorr wrote:
I’ve got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I’ve set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls it still shows
“(ULAW)” (G.711)
Hi!
Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files
-in and -out.
It is not mixing them in the end.
queues.conf has monitor-type=MixMonitor...
Would somebody help me debug why it doesn't mix the sounds??
Thanks
___
--
On Tue, 15 Dec 2009, Ben Schorr wrote:
Ahhh...yes, I think that may have been it. I moved G.729 to the top of
that list (just below disallow) and now I have a restart when
convenient pending. Is that sufficient or do I have to actually reboot
the server for the change to take effect?
Ahhh...yes, I think that may have been it. I moved G.729 to the top of
that list (just below disallow) and now I have a restart when
convenient pending. Is that sufficient or do I have to actually reboot
the server for the change to take effect?
Best wishes and aloha,
Ben M. Schorr
Chief
You should only need a reboot for DAHDI changes (not always then...)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 1:08 PM
To: Asterisk Users Mailing List -
O.K., thanks, I'm catching on (slowly). Waiting for the next call to
see if the SIP.CONF change did the trick.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From:
O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it seems
to be using G.729 now.
Is there something in Dahdi that I need to configure so that inbound
calls (from the PRI on a Digium TE205) use G.729 to get to the
hbk fo...@online.no writes:
Where to look for forgotten DTMF detection settings?
Try relaxdtmf=no. sip show settings to check that it worked.
/Benny
___
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asterisk-users mailing
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it seems
to be using G.729 now.
Is there something in Dahdi that I need to configure so that inbound
calls (from the PRI
Gavin Spurgeon gspurg...@dageek.co.uk writes:
iSip (£2.39)
http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality from iSip, at least from
the other end so to speak. It shares the basic flaw of not being able
to run
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones (or
at least a particular group of them) and my Asterisk server will use
G.729. Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.
All of our
IMO you can only use the G.729 on a SIP call. If the call falls onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
Oh, dear. So my users with less-than-ideal bandwidth are stuck with
drop-outs and poor sound quality because they can't use the reduced
bandwidth codec for those calls? :-(
They've been complaining that they often end up on a call where one or
both parties are cutting in and out. Unfortunately
Why not restrict these 8 users to a SIP provider like (but not)
bandwidth.com? By eliminating the PRI element, you should completely
resolve the problem.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
- Ben Schorr b...@rolandschorr.com wrote:
Oh, dear. So my users with less-than-ideal bandwidth are stuck
with
drop-outs and poor sound quality because they can't use the reduced
bandwidth codec for those calls? :-(
They've been complaining that they often end up on a call where one
Well, I know I still have a LOT to learn about Asterisk but...how will
they get their incoming phone calls from their DIDs (which the TelCo
sends to their PRI) if I move the remote office onto a SIP provider?
The PRI doesn't seem to cause any problem for the majority of the users
(at the home
I thought I already did that - which is how they now get some (but not
yet all) of their calls on G.729. scratching head
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original
Do your routers allow giving these users maximum priority? What is the
effective bandwidth on the VPN connection?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009
Watch the calls on the console. Try both ways. Document what you see and
your codec settings on both the phone, and sip.conf.
You may have to tell the phone that the only codec it can use is G.729,
don't just make that first choice. Make it the only choice.
Cary Fitch
-Original
And tell Asterisk that G.729 is the only codec for that number as well!
Cary Fitch
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Yes, the routers are another issue we're dealing with. We've configured
them to prioritize traffic to/from our Asterisk server but I'm not
convinced that setting is really working as expected. So we're working
with the vendor on that.
The effective bandwidth is 6Mb/sec down x 1.4Mb/sec up (so
- Ben Schorr b...@rolandschorr.com wrote:
I thought I already did that - which is how they now get some (but
not
yet all) of their calls on G.729. scratching head
VERY SIMPLIFIED VIEW
Allowing G.729 in your configurations (disallow=all, allow=g729) enables those
endpoints to use that
That's a bit misleading. Yes calls that travel over a PRI will be using
ulaw, but only over the PRI leg of the call. The SIP leg can still be
using G.729 with asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers you are working
O.K., I think I'm catching on. I only have a single SIP.CONF file that
ALL of the extensions are using so I'm gathering that I need to set up a
separate SIP.CONF file (or perhaps just an included file) for the 8
users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how
O.K., so for now (as a test) I just commented out the allow=ULAW line
in the SIP.conf (actually it's sip_general_additional.conf on this
FreePBX box) and that does seem to be forcing all traffic to G.729.
I think ultimately I'd like to let the local users use ULAW because it
seems to sound better
I don't know how FreePBX works, but with vanilla Asterisk you would do
something like this with your sip.conf:
[general]
disallow=all
allow=ulaw
allow=g729
[localA]
callerid=Local phone A 100
username=localA
secret=blahblah1
[localB]
callerid=Local phone B 101
username=localB
secret=blah1blah
On Tue, Dec 15, 2009 at 12:37:59PM +, listu...@spamomania.co.uk wrote:
Some recent issues I had with hardware seem to come back to not
understanding two very similarly named files:
/etc/asterisk/dahdi-channels.conf
/etc/asterisk/chan_dahdi.conf
I've modified the chan_dahdi.conf to
At 12:45 PM 12/15/2009, you wrote:
The PRI doesn't seem to cause any problem for the majority of the users
(at the home site) it's just the 8 users at the remote site who are
complaining of quality issues.
So out of curiosity, if you were to limit the phone usage for an hour
at the remote site
If I installed a Digium echo cancellation module on my TE121 card, do I need to
remove the echocanceller line under the system.conf? How should I have it?
This is my system.conf:
bchan=1-23
dchan=24
echocanceller=mg2,1-23
Thank you!
Hin
From: hin lee
O.K., I restored the Allow=ulaw in the sip_general_additional.conf file,
then I found the individual extension settings in the
sip_additional.conf file and I added
disallow=all
allow=g729
to each of the extensions at the remote site. Then I did a SIP RELOAD.
So we'll see how that goes.
Thanks
Hi,
I think you need to remove the line echocanceller in system.conf
You could also try to use fxotune, it'a really improving things.
You also need to put echocancel=yes in chan_dahdi.conf
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
On Mon, 2009-12-14 at 11:49 -0600, Bob Smither wrote:
This has to be easy, but I have spent a fair amount of time looking for
a solution to no avail. I am trying to get multiple phones to ring when
a call comes into an Asterisk box from a particular phone number. What
happens is that only
I'm trying to get two server communicate with each other and call from one to
the other but, I'm having a lot of problems.
I tried to create a iax trunk between the two:
At the server:
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
Shouldn't you have 'nat=yes' in your peer context [sipconnect.sipgate.de]
and not in [general]?
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I'll check it out, but Grandstream HT503 doesn't have a good introduction on
voip-wiki web-page:
http://www.voip-info.org/wiki/view/HT-503
--
Joseph
On 12/11/09 19:37, jonas kellens wrote:
Grandstream HT503
Noy a really big problem to configure, but in my case the FXO port always
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
Gavin Spurgeon gspurg...@dageek.co.uk writes:
iSip (£2.39)
http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality from iSip, at least from
the
Did you check the jitter settings on asterisk the phones as well?
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Date: Wednesday, December 16, 2009, 1:26 AM
trust both the side giving IP address
in the sip.conf
I did this in the iax.conf file
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
context=from_client
host=172.16.0.11
trunk=yes
qualify=yes
Alex Samad wrote:
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
Gavin Spurgeon gspurg...@dageek.co.uk writes:
iSip (£2.39)
http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality
Hi everyone,
I'm having a trouble while developing monitoring tool for queues. I'm
using C# Sharp I follow the instruction on
http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+C+Sharp.
My question is how can I get information about how many people are there
waiting on queue with
Cyprus VoIP wrote:
This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353
v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
See if you find this tutorial on IAX peering useful:
http://astrecipes.net/index.php?n=204
Thanks
l.
2009/12/15 Landy Landy landysacco...@yahoo.com
Hello List.
I have a question regarding connecting two asterisk servers. I'm trying to
learn how asterisk comunicates from server to server. I
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