[Dec 14 09:23:18] ERROR[15198]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR
transmission error to XX.XX.XX.65:5037, rtcp halted Operation not
permitted
This is a log entry on a public Asterisk-server. My SIP-client
(Grandstream GXP2010) can perfectly register to this public
Asterisk-server.
My
11 dec 2009 kl. 17.18 skrev Jerry Geis:
Where in the code does something like:
register = user[:secret[:authuse...@host[:port][/extension]
from sip.conf 1) get parsed 2) actually register.
I tried looking in channels/chan_sip.c and don't see where that happens.
Can
11 dec 2009 kl. 23.21 skrev John Taylor:
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
This is an effect of the
Hi everybody,
I have a sip phone (Siemens) which has no sip functions at all.
Is is possible to press #4 by example to put the call on hold then
dial #2 to get the call back ?
I'have look at features.conf but i did not find the solution.
I know the call parking functionnality, but i would
look at Random()
2009/12/12 Landy Landy landysacco...@yahoo.com
Hello List.
I would like to know how I can use two or more service providers with
asterisk to be used randomly for ei, if an user tries to make a call I would
like to randomly use a provider. It doesn't matter where the call is
That's a pretty crappy phone huh? :)
Anyway you should be able to do it on features.conf, in the applicationmap
section. I'm not entirely sure there's a dialplan app that allows you to put a
channel on hold and take it back later.
Vinícius Fontes
www.asteriskforum.com.br - Informações e
Hi there,
i'm using dahdi to manage a B400P openvox BRI card.
All works as expected, i would like to know if there ia a way to put the
call in REMOTE HOLD, like pressing R button on ISDN phone.
This can be done by CAPI using the proper application ,
It is implemented on DAHDI ?
Regards Andrea
The diagnostic tool is dahdi_test or -- if memory serves correctly
zttest. If you don't see a minimum of 99% after a minute or two, you
are in trouble.
hbk fo...@online.no wrote:
Thank you, very interesting!
As I understand the Digium card is used as a interrupt source for Asterisk?
Hi,
I've noticed that a small but meaningful quota of calls from my Alcatel PBX to
Asterisk are failing.
This does not always happen and it is not easily reproducible but on high
traffic I do get a large number of cases.
Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145
Thx!
Did try callcounter=yes and it worked the way u told me!
It might have solved another problem 2, need to do some more tests...
On Sun, 13 Dec 2009 15:14:22 -0500, Leif Madsen wrote:
Philipp Kempgen wrote:
Magnus Benngård schrieb:
Set
call-limit=10
(or any other value 0)
Actually,
Did move 0317998975 phone from my home to my office, didnt need any:
nat=yes in sip.conf, everything worked.
I did also add callcounter=yes in sip.conf so I am not sure how it
will work when I move the phone to my home and need nat=yes again.
Will do some tests later tonight when I am at home.
On
Daniel Stefanus wrote:
Hi,
I want to reconfigure my asterisk dialplan.I have a problem.I have 4
agents in a queue.How is the configuration for the asterisk dialplan if
I want to have only 4 agents maximum who can receive the phone,so if the
fifth caller try to entering the queue they will be
Hi there,
I'm using asterisk meetme function like:
exten = 9070,n,MeetMe(|dcM)
and everything works pretty well. But I would like to add a review of
the entered conference number before the user jumps into the conference.
Somthing like:
*:Please enter the conference number followed by the
snip
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
/snip
Your provider is probably sending the DID in the SIP header TO:
On Mon, Dec 14, 2009 at 05:32:21AM -0800, Vieri wrote:
Hi,
I've noticed that a small but meaningful quota of calls from my Alcatel PBX
to Asterisk are failing.
This does not always happen and it is not easily reproducible but on high
traffic I do get a large number of cases.
Example:
I'm using asterisk meetme function like:
exten = 9070,n,MeetMe(|dcM)
and everything works pretty well. But I would like to add a review of
the entered conference number before the user jumps into the conference.
Somthing like:
*:Please enter the conference number followed by the
Bump! And some more information (see below for initial problem):
This problem is intermittent, but you don't have to wait long for it to happen.
Also, sometimes when the reregister happens (and the client has been wrongly
unregistered) asterisk sends the correct response to the client indicating
Are you sure this isn't a Windows zeroconfig problem? If Win drops the
connection while * is talking to your client, the registration could drop
too..
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: Monday,
I have tried this with windows firewall both on and off - same problem.
Thanks,
Nic.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: 14 December 2009 14:53
To: 'Asterisk Users List'
Subject: Re: [asterisk-users]
Hello,
I've been trying to setup asterisk with zaptel for the last 3-4 days. I had
a lot of errors and fixed all of them but asterisk still didn't work. Then I
saw that zaptel couldn't be loaded because I was on a vserver and I didn't
have the devices under /dev/zap/. I asked the system guys to
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external SIP
provider calls coming in and have put together a little call platform
and I'm stunned at the flexibility.
There is one issue for me.
The SDP response is the IP of the Trixbox server.
On Fri, Dec 11, 2009 at 1:38 PM, Christian Victor
christ...@victormedia.dewrote:
Hi!
Are you sure you are getting Astrisk out of the media path? I guess
reinvite must be allowed. Then it should work without transcoding
licenses.
Maybe you
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
well.
After a while of juggling it
On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external SIP
provider calls coming in and have put together a little
I have never used that card myself, but I have never seen an analog board
reporting a RED alarm. Probably there is something incorrect in your
configuration. Please post your /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf.
Vinícius Fontes
www.asteriskforum.com.br - Informações e
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 one of the phones rings.
I have several GrandStream BT200
2009/12/14 Christian Theune c...@gocept.com
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux, dahdi-tools and libpri
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor
On Tuesday 24 November 2009 13:10:43 Tzafrir Cohen wrote:
On Tue, Nov 24, 2009 at 01:07:28PM +0100, Eckhard Jokisch wrote:
Hi,
I have 4 ISDN channels (2 lines) and each line may do calls of up to 360
minutes/month for free.
As I understand asterisk will pick the first available line so
Hi!
Trying to figure out how to rewrite calling number of an incoming call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite shows 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten = 977/733025975,1,Set(CALLERID(number)=0317998975)
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
I would like to do something like:
[myagents]
exten = XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
exten =
The calls itselves doesn't take a lot of CPU resources, even more considering
you're willing to use hardware echo cancelling. The real CPU hogs are apps like
MeetMe() and AGI scripts. Those are no worse than audiotranscoding thought.
You also should design the system in such way there's as few
On Mon, 2009-12-14 at 14:17 -0300, Vinícius Fontes wrote:
I have never used that card myself, but I have never seen an analog board
reporting a RED alarm. Probably there is something incorrect in your
configuration. Please post your /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf.
On Mon, 2009-12-14 at 11:11 -0600, Tilghman Lesher wrote:
On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external
See if it plays back properly.
Running aplay as asterisk user seems to be no problem:
aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
Little Endian, Rate: 48000 Hz, mono
aster...@puppy:~$ aplay -Dpulse
On Wed, Nov 11, 2009 at 8:02 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Jonathan Thurman wrote:
Any chance that 64 bit Linux will be supported?
There is a small chance; I've done some work in the past week while
traveling to attempt solve the 64-bit problems, and I fixed some of them
Your dahdi/system.conf seems fine. But your chan_dahdi.conf is missing some
lines.
[trunkgroups]
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
On 12/14/2009 06:45 PM, Olivier wrote:
2009/12/14 Christian Theune c...@gocept.com mailto:c...@gocept.com
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans
Hi,
does anybody know how I can make this two configuration-settings with
Asterisk-iaxmodem for ourtgoing faxes with Hylafax on an ISDN-line?
1. disconnect-time after dialing without an answer. (is now 30 seconds, but it
must be higher)
2. isdn-service set to 3,1 kHz Audio, (is now speech)
On Mon, 14 Dec 2009 11:02:02 -0800, Dave Platt dpl...@radagast.org
wrote:
See if it plays back properly.
Running aplay as asterisk user seems to be no problem:
aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16
The easiest solution to deal with this is to have one context with
different extensions for the different numbers and route the incoming
calls from there. It should look something like this (not a tested
piece of asterisk script, just an example to give the idea).
Hope it helps :-)
Erik
Hi!
Having two TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.
Chris
2009/12/14 das sandesh sandesh...@gmail.com:
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am
I'm just curious to know if anyone is using a usb 2.0 / ISDN30
(specifically EuroISDN) device. We are looking to purchase another pci
card, but was wondering if anyone has any horror / success stories to
share regarding a usb device.
TIA
Julian
___
--
Hi,
I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here
my settings:
chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
facilityenable=yes
rxwink=300 ; Atlas seems to use long (250ms) winks
;
On Monday 14 December 2009 12:52:50 pm listu...@spamomania.co.uk wrote:
On Mon, 2009-12-14 at 11:11 -0600, Tilghman Lesher wrote:
On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of
On Mon, Dec 14, 2009 at 05:53:40PM +0200, gurel kaynak wrote:
Hello,
I've been trying to setup asterisk with zaptel for the last 3-4 days. I had
a lot of errors and fixed all of them but asterisk still didn't work. Then I
saw that zaptel couldn't be loaded because I was on a vserver and I
On Mon, 14 Dec 2009 11:02:02 -0800, Dave Platt dpl...@radagast.org
wrote:
See if it plays back properly.
Running aplay as asterisk user seems to be no problem:
aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux,
--- On Mon, 12/14/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Dec 14, 2009 at 05:32:21AM
-0800, Vieri wrote:
Hi,
I've noticed that a small but meaningful quota of
calls from my Alcatel PBX to Asterisk are failing.
This does not always happen and it is not easily
On Mon, Dec 14, 2009 at 02:17:39PM -0300, Vinícius Fontes wrote:
I have never used that card myself, but I have never seen an analog
board reporting a RED alarm.
Ahem. Wcfxo always has (AFAIR). Red alarm means that no line is
connected (it gets no current from the remote FXS in the central
Again, more info:
Since I added rtcachefriends=yes this problem went away, but I don't really
want the friends to be cached, because I want changed to be applied ASAP.
Does anyone else have experience of the peers being unregistered before their
time with rtcachefriends=no?
Nic.
From:
Hello,
I would like to connect 2 asterisk boxes together, so this is my scenario:
Asterisk Main: it is connected to many sip providers and its main purpose as
a call termination forwarder.
Asterisk B: its connected to E1, and its purpose to terminate calls. It
will receive SIP
Hello all,
I am trying to use asterisk 1.4.24 so that I can get app_rxfax working, I
installed it, along with the versions of libpri and zaptel that had release
dates closest to the release date of 1.4.24, however, I now have a problem
where outbound dialing now fails, cause 99 on the PRI.
What you are missing is the new state-interface parameter to AddQueueMember.
You can't use functions in a hint exten.
Steve
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying
This may belong on -biz, but does anyone have experience with a decent and
cheap IVR/prompt recording house?
Are decent and cheap mutually exclusive?
A nice *sounding* lady would be nice... you can keep any burly voice studios to
yourself :)
Thanks
Dave
Thanks Victor and Vinícius for the information.
I will not be doing any transcoding but using some AGI scripts, I will
update the status once I configure and start using them.
Thanks
Sandesh
On Mon, Dec 14, 2009 at 1:59 PM, Christian Victor
christ...@victormedia.dewrote:
Hi!
Having two
On Monday 14 December 2009 03:20:11 pm Stephen Davies wrote:
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
But more dynamical, so I would try and look up the actual channel in the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Gibbons wrote:
This may belong on -biz, but does anyone have experience with a decent and
cheap IVR/prompt recording house?
Are decent and cheap mutually exclusive?
A nice *sounding* lady would be nice... you can keep any burly voice
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 .
--
Girgis Rasmy
When agents are on the phone, and the CLI queue show command shows their
status as busy, the queue still tries to send them calls.
Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add
agents. ringinuse is set to no for queue. Agents are using Polycom 430s.
dialplan:
exten =
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote:
Fring, it's free and works perfectly with an Asterisk server..
thanks
On 13 Dec 2009, at 10:15, Alex Samad wrote:
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well with
I personally have not had much luck with these softphones because the
iPhone 3G seems to be underpowered and just doesn't run them well
enough to sustain good voice quality, irrespective of wifi network
conditions. I could be mistaken, though.
It's not going to happen over ATT's 3G network
--- On Mon, 12/14/09, Vieri rentor...@yahoo.com wrote:
From: Vieri rentor...@yahoo.com
Subject: Re: [asterisk-users] Asterisk ZAP/DAHDI reads phantom digit on
overlap PRI
To: asterisk-users@lists.digium.com
Date: Monday, December 14, 2009, 3:26 PM
--- On Mon, 12/14/09, Tzafrir Cohen
I find that Siphone works great on the iTouch. Tried it with my own
asterisk box as well as Callcentric and MagicJack and it was very
clear and stable. Haven't played with it since the last firmware
update though as the update removed support for 3rd party headsets .
On 12/14/09, Alex Balashov
Pat Fleet, the original voice of ATT recorded a free set of the
prompts included in Asterisk and also does custom IVR prompts through
her website at http://patfleet.com/ I'm not sure what the going rates
for IVR prompts is, but she charges $15/phrase.
On 12/14/09, Barry L. Kline
Well I have a 3gs - will tell you how that goes.
decided against siax - have to pay for the base model.
installed fringe, but no voip over 3g, have to wait till i get home, but
it registered with my asterisk server so ..
I am looking for the hacked fring.ipa which allows voip over 3g, just so
I
Hello List.
I have a question regarding connecting two asterisk servers. I'm trying to
learn how asterisk comunicates from server to server. I already have a server
running smoothly now, I'm installing another one to test it along side the
actual one.
I would like to run different scenarios:
this i got from syslog:
puppy:~# grep pulse /var/log/syslog | tail -3
Dec 14 20:32:45 puppy pulseaudio[25967]: main.c: Unable to contact D-Bus:
org.freedesktop.DBus.Error.Spawn.ExecFailed: /usr/bin/dbus-launch
terminated abnormally without any error message
Dec 14 20:32:46 puppy
Joseph,
You may want to try RPA-2E1S1O from www.broad-tel.com from China. It
provides real FXO port that registers with Asterisk.
David
On Sat, Dec 12, 2009 at 1:37 AM, Joseph syscon...@gmail.com wrote:
I'm looking for a reliable ATA FXO/FXS adapter.
Linksys 3102 - a lot of echo problem + two
Hi All;
When using the digest authentication method, so I have to create the realm
domain with its username and passwords to be used for SIP digest
authentication, correct?
Now, how to create this domain? Should be reachable (can be ping) from a remote
device?
In other words, to create this
IAXDIAL is free on app store works great on WiFi even true NATs but seem
blocked for GPRS.
HB
Re: [asterisk-users] iphone client app
From:
Alex Samad a...@samad.com.au
Date:
Tue, 15 Dec 2009 12:08:37 +1100
To:
asterisk-users@lists.digium.com
To:
asterisk-users@lists.digium.com
Well
On Mon, 2009-12-14 at 14:10 -0600, Tilghman Lesher wrote:
I don't want to start a war, but there is a square to that. I'm new to
Asterisk having spent years in analogue telephony. If I can get a test
Asterisk working on a cheap clone card without a hitch, I'm most likely
to expand this
On Mon, 2009-12-14 at 22:27 +0200, Tzafrir Cohen wrote:
On Mon, Dec 14, 2009 at 02:17:39PM -0300, Vinícius Fontes wrote:
I have never used that card myself, but I have never seen an analog
board reporting a RED alarm.
Ahem. Wcfxo always has (AFAIR). Red alarm means that no line is
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