2009/1/20 D Tucny d...@tucny.com
2009/1/20 Olivier oza-4...@myamail.com
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
Yes, testing it...
A prospective customer of mine is asking to integrate its iphones with an
Asterisk server and after googling, I still have some
One thing to note about fring, the device establishes a connection using
fring's proprietary protocols to fring servers, fring then establishes SIP
connections from those servers... So, even if connected to the office Wifi
connection, you could experience connectivity issues or high latency
Hi,
Has anyone any return to share about Skype-Digium beta program ?
I would be very curious to know how things are going on this.
Regards
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To
2009/1/20 Olivier oza-4...@myamail.com
One thing to note about fring, the device establishes a connection using
fring's proprietary protocols to fring servers, fring then establishes SIP
connections from those servers... So, even if connected to the office Wifi
connection, you could
Hi Steve;
Do u mean by the Iaxy2 is that IAX digium gateway adaptor?
If yes, then it has a codec limitation and it does not take ddns name (it needs
IP address), also it is gateway and not IP Phone.
Or u mean something else?
Do u have a link for it so I can see it?
Regards
Bilal
Anyone
Laurent a écrit :
Le 19.01.2009 08:50, Benoit a écrit :
Laurent a écrit :
Well, the telcos techs said a straight cable should do the trick, but
since i didn't get any isdn link up
with the straight, i built a crossover like what you described, with no
luck either.
Hi All;
I am facing a problem that always the users confused and connect the telephone
line coming from the telephone service provider to the FXS port and cause it to
be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and
connect the line to fxs while it should be
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
If a user has recorded a new voicemail message (e.g. unavailable
message) then it is prompted with 3 choices.
1. accept recording
2. listen to the recorded message
3. rerecord the message
Isn't it possible to cancel the recording?
Benoit a écrit :
Laurent a écrit :
Did you check (like with a multimeter or something similar) the
connectivity of your cable ? the first E1 crossover cable I made
had a problem (entirely my own fault) and I thought it didn't
work. The way I checked was by connecting the two ports of the
What you need is a so called T38Gateway application.
there is a patch o the tracker which you might want to try:
http://bugs.digium.com/view.php?id=13405
klaus
Steve Gladden schrieb:
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP
hello
When I bridge an incoming and outgoing call (attempting to simulate
call-forwarding) I'm only getting one CDR -- that of the outgoing call.
A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell
phone on PSTN) and bridges the call.
The only CDR created is from B to C.
On 20 Jan 2009, at 10:18, bilal ghayyad wrote:
Hi All;
I am facing a problem that always the users confused and connect the
telephone line coming from the telephone service provider to the FXS
port and cause it to be damaged, specially if the card was 2 fxs and
2 fxo, so they make
Hi!
I have two identical SIP accounts on Asterisk 1.4.22. One account is
registered with eyebeam, the other one is registered with a SNOM phone.
When using the eyebeam client DMTF detection works fine, when using the
SNOM phone many digits are missing in the DTMF detection.
I analyzed with
How are you testing DTMF detection with the Snom UA?
Klaus Darilion wrote:
Hi!
I have two identical SIP accounts on Asterisk 1.4.22. One account is
registered with eyebeam, the other one is registered with a SNOM phone.
When using the eyebeam client DMTF detection works fine, when using
Hi
How do i set up so that everyone can dial, for example *21* to forward all
calls to a cellphone or another extension and how do I enable so that cals can
be transferd between extentions.
I use asterisk 1.6 and have my phones in unistim.conf and my extensions in
extensions.conf.
Regards
On Tue, 20 Jan 2009, Olivier wrote:
GTalk seems to fill the bill of requirements, though, I don't think it's
available on Nokia mobile phones ..
My Nokia mobile phone has a SIP client built-in which uses Wi-Fi... And
while it's not perfect, it's actually very usable and works well with
2009/1/20 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 20 Jan 2009, Olivier wrote:
GTalk seems to fill the bill of requirements, though, I don't think it's
available on Nokia mobile phones ..
My Nokia mobile phone has a SIP client built-in which uses
Hi folks,
I wonder if any of you out there are using Siemens S685IP base station(s) (with
S68H handsets) on Asterisk and experiencing problems with SIP registrations
where the SIP extensions do not ring and peers become unreachable after a
period of time.
Symptoms are rather sporadic,
On Tuesday, January 20, 2009, bilal ghayyad wrote:
What is the solution for this disaster?
I live in UK, where we don't use RJ11 for telephones and so need to
use adapters, which I just leave hanging out of the FXO ports. With
the adapters in place, it's difficult to plug the phones into the
Hi,
I have a question...
With the variable ${CHANNEL} I can get the channel whose made the call, or the
caller. How can I get the channel of the called side?
Veja quais são os assuntos do momento no Yahoo! +Buscados
http://br.maisbuscados.yahoo.com
features.conf
2009/1/20 Ralf Träskman r...@adlibris.com
Hi
How do i set up so that everyone can dial, for example **21** to forward
all calls to a cellphone or another extension and how do I enable so that
cals can be transferd between extentions.
I use asterisk 1.6 and have my
Message: 18 Date: Mon, 19 Jan 2009 19:56:14 +0200 From: Tzafrir Cohen
tzafrir.co...@xorcom.com
Subject: Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms To:
asterisk-users@lists.digium.com Message-ID:
On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote:
I am missing any
2009/1/20 Olivier oza-4...@myamail.com
2009/1/20 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 20 Jan 2009, Olivier wrote:
GTalk seems to fill the bill of requirements, though, I don't think it's
available on Nokia mobile phones ..
My Nokia
2009/1/20 Olivier oza-4...@myamail.com
One thing to note about fring, the device establishes a connection using
fring's proprietary protocols to fring servers, fring then establishes SIP
connections from those servers... So, even if connected to the office Wifi
connection, you could
Mark Michelson wrote:
If you are using gsm prompts and gcc version 4.2 or higher, then you may be
experiencing the optimizer bug that gcc has with gsm audio. The workarounds
for
this are to use a different format for sounds or to set the DONT_OPTIMIZE
flag
in menuselect. If you want an
On Tue, 20 Jan 2009, Geoff Lane wrote:
On Tuesday, January 20, 2009, bilal ghayyad wrote:
What is the solution for this disaster?
I live in UK, where we don't use RJ11 for telephones and so need to
use adapters, which I just leave hanging out of the FXO ports. With
the adapters in place,
I have a similar problem with Snom. Since I've upgraded from version 6 to
version 7 I cannot call IVR systems. The first DTMF goes ok, but after that
others are not accepted nor I am heard by the operator at the other side.
Since I am the only one who has Snom here I didn't bother to debug it...
2009/1/20 Simon Dixey simon_...@hotmail.co.uk
Hi folks,
I wonder if any of you out there are using Siemens S685IP base station(s)
(with S68H handsets) on Asterisk and experiencing problems with SIP
registrations where the SIP extensions do not ring and peers become
unreachable after a
Dear Steve;
But it is not logical to keep having damaging ports, till when these mistakes
will be able to keep replace it?
Regards
Bilal
Hi All;
I am facing a problem that always the users confused
and connect the
telephone line coming from the telephone service
provider to the FXS
Hi,
I am running Asterisk 1.4.22. If in X-Lite I have set to hang-up after 0
seconds of RTP the call gets cut off between 1 1/2 to 2 minutes. I have tried
to connect to another server and the call stayed up. If I take out the (RTP)
setting then it works fine. What would cause the RTP to stop
I have a Snom 320 and both inband and RFC2833 OOB work fine for me.
Yehavi Bourvine wrote:
I have a similar problem with Snom. Since I've upgraded from version 6
to version 7 I cannot call IVR systems. The first DTMF goes ok, but
after that others are not accepted nor I am heard by the
Dear Gordon;
I did not understand the idea of the following:
If you're carefull, you can clip the tang of the RJ11 plug on the end of the
BT adapter so that it needs a small screwdriver to remove it from the socket on
the board...
Do u have any link?
What do u mean by clibing the tang of the
hi
i have a E1 connected to the PSTN when the remote site hang up asterisk dont
detect it
i found this in the pri debug
Ext: 1 Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]
asterisk version is 1.4
any ideas?
thanks!
--
On Tuesday, January 20, 2009, bilal ghayyad wrote:
What do u mean by clibing the tang of the RJ11 plug on the end of
the BT adaptor?
On an RJ11 plug, the casing includes a springy piece that locks the
plug into an RJ11 socket. When plugged in, the end of the springy
piece sticks out of the
On Tue, 20 Jan 2009, bilal ghayyad wrote:
Dear Gordon;
I did not understand the idea of the following:
If you're carefull, you can clip the tang of the RJ11 plug on the end of the
BT adapter so that it needs a small screwdriver to remove it from the socket
on the board...
Do u have any
Geoff Lane wrote:
If you take care, it's possible to cut off the end of the tang that
sticks out of the socket while leaving enough of the tang to lock the
plug in place.
You may not even need to clip the end off, with the last lot of RJ-11
plugs I ordered the tangs were short enough to snap
On Tue, 20 Jan 2009, bilal ghayyad wrote:
Do u mean by the Iaxy2 is that IAX digium gateway adaptor?
Yes. Your request was for a hard phone, but I was replying to the reply
about the Iaxy2.
Thanks in advance,
Steve
Hello,
We're having some issues with CallerID and I thought someone here might
be able to shed some light as none of our carriers seem to know what I'm
talking about.
The issues is this:
A client of ours uses an after-hours voicemail service as mandated by
their corporate office. We have a
Most voicemail/answering service dont' care about callerid or ani,
they instead use the DID that the call comes in on to decide how to
answer the call.
Get a different voicemail/answering service.
On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote:
Hello,
We're having
Hi all,
Does asterisk (I'm using 1.4.19) sets any channel variable with the holded
chan when it does an atxfer? I tried to see if it does on the source, but i
didn't find any clue, neither enabling console debug.
Thanks,
Gabriel
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Alex Balashov schrieb:
How are you testing DTMF detection with the Snom UA?
The Voicemail(u...@context) application asks the user for the voicemail
password.
Using eyebeam everyhing works fine. Using SNOM (320 FW 7.3.10a) it works
almost never.
regards
klaus
Klaus Darilion wrote:
Not a possibility I'm afraid. Our client is an insurance agent and the
voicemail/answering service is mandated by corporate.
There also are not various DID's to call in on. All voicemail calls go
to an 800 number
Thanks for your advice though.
Most voicemail/answering service dont' care
Hi!
I have the following scenario:
Asterisk
INVITE- |
--200,ACK-- |
Playback(Foo)
|
Dial(..)
| -INVITE-
| -404. ACK--
|
As my extension configuration stops after the Dial command I
On Tue, 2009-01-20 at 16:08 +0100, Klaus Darilion wrote:
As my extension configuration stops after the Dial command I expect
Asterisk to hang up the call. Instead I see on the console:
|
|
== Auto fallthrough, channel 'SIP/81.16.153.183-0883dea0' status is
Hi List,
i've set up an Asterisk 1.6.0.3 Server equipped with an Xorcom Astribank BRI
XR0013 (2 Port BRI) and Dahdi 2.1.0. In- and outgoing calls are no problem. But
I can't get Asterisk/DAHDI to use the second Number of the BRI. All calls set
the primary number as callerid. It wouldn't be a
On Tuesday 20 January 2009 06:31:51 Kevin P. Fleming wrote:
Mark Michelson wrote:
If you are using gsm prompts and gcc version 4.2 or higher, then you may
be experiencing the optimizer bug that gcc has with gsm audio. The
workarounds for this are to use a different format for sounds or to
with {$BRIDGEPEER}
2009/1/20 Jose Enes Mateus jemat...@yahoo.com.br
Hi,
I have a question...
With the variable ${CHANNEL} I can get the channel whose made the call, or
the caller. How can I get the channel of the called side?
Veja quais são os assuntos do momento no Yahoo!
John Todd schrieb:
So here are some projects you might look into:
- open-source ZRTP implementation
libzrtpcpp is open source. GPL.
http://www.gnu.org/software/ccrtp/
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk:
I've noticed on a few installations that the very first audio played after a
call in answered (eg: Greeting), the first part of the audio is
cutoff/stuttered.
Is this because Asterisk needs some RTP to create a sync for audio - and the
first 1 second is lost? Should one play 1 sec of silence
At 12:48 PM 1/19/2009, you wrote:
I do not know if they use analog or digital signals for the phones but
if we use the cell phone system as an example they took down all analog
towers because they could service more phones on the same bandwidth with
digital. I would assume that would hold true
features.conf for transfers
for call forwardin you need some application logic.
e.g.
_**21**. = {
Set(NUM=${EXTEN:6}); // contains the new target
// now store this number somewhere, e.g. astdb, odbc ...
...
}
context fromPstn {
1234 = {
// check if user has actived forwarding
Jared Smith schrieb:
On Tue, 2009-01-20 at 16:08 +0100, Klaus Darilion wrote:
As my extension configuration stops after the Dial command I expect
Asterisk to hang up the call. Instead I see on the console:
|
|
== Auto fallthrough, channel
Tilghman Lesher wrote:
Actually, this problem should already be fixed. It was a case of optimization
with assembly code. Adding the noclobber option to the assembly should have
fixed this.
Well, people are still running into it as evidenced by this thread, and
it's a very subtle issue to
Hi All,
I'm confused! My Asterisk system has a Zap trunk and three SIP trunks.
I'd like to configure the dialplan to route via the first trunk in a
list and if that's not available or it's busy, fall over to the
second, then to the third, etc.
AIUI Dial(Zap/1SIP/out1SIP/out2/${EXTEN}) rings all
I would use the ${DIALSTATUS} variable. In your dialplan dial the first trunk
you wish, then afterwards examine the ${DIALSTATUS} variable. If that is not
equal to ANSWER then dial your second trunk and so on.
For example:
exten = s,1,Dial(ZAP/1/${EXTEN})
exten = s,n,ExecIf($[${DIALSTATUS} !=
I have ran into a case using 1.4.22 where a SIP call to an asterisk
client (running a slow PC) to ALSA
does not hangup the call when it is done. The server is using call files
to initiate the call, the client answers on
the ALSA port, the server plays the message and hangs up.
I found that
I contacted Christophorus directly and he was able to point me to the
correct solution. Apparently, not all TFTP servers are created equal.
I uninstalled tftpd (running Ubuntu server Hardy Heron) and installed
atftpd instead. As soon as this was configured, the phone quit
searching for the
Hi All,
A long time ago I posted about an issue where calls on one of our Asterisk
boxes were being dropped in Voicemail (and only in voicemail) after about 20
seconds with the error logged [Jan 19 14:33:26] WARNING[27458]:
chan_sip.c:1980 retrans_pkt: Hanging up call
hi,
try to set the rtptimeout value in sip.conf to a resonable value - so
asterisk will kill the channels if it does not receive rtp traffic for
the specified time
regards,
Wolfgang
Jerry Geis schrieb:
I have ran into a case using 1.4.22 where a SIP call to an asterisk
client (running a
hi,
try to set the rtptimeout value in sip.conf to a resonable value - so
asterisk will kill the channels if it does not receive rtp traffic for
the specified time
regards,
Wolfgang
I uncommeted the rtptimeout=60 value in sip.conf and did a reload.
It still hasnt dropped the dead
Jerry Geis wrote:
hi,
try to set the rtptimeout value in sip.conf to a resonable value - so
asterisk will kill the channels if it does not receive rtp traffic for
the specified time
regards,
Wolfgang
I uncommeted the rtptimeout=60 value in sip.conf and did a reload.
It still hasnt
On Tuesday 20 January 2009 11:13:43 Kevin P. Fleming wrote:
Tilghman Lesher wrote:
Actually, this problem should already be fixed. It was a case of
optimization with assembly code. Adding the noclobber option to the
assembly should have fixed this.
Well, people are still running into it
Dear List,
I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3.
When I try making a call with a .call file, the call goes straight to the
dialplan and start executing the dialplan even before the called party has
pick up. Anybody knows why by any chance?
Any help would be
Using centos 5.2,
I want to use a kickstart file to select packages in order to have an
unattended install onto a bare-metal server.
Is there any way of finding out which packages I need to compile, build
and run asterisk ?
I generally want to build all modules in asterisk and the wct4xxp
On 20 Jan 2009, at 20:58, Julian Lyndon-Smith wrote:
Using centos 5.2,
I want to use a kickstart file to select packages in order to have an
unattended install onto a bare-metal server.
Is there any way of finding out which packages I need to compile,
build
and run asterisk ?
I
Brian Alexander wrote:
Mark,
Thanks - that was the problem I was having. Is there somewhere I could
have looked to have discovered the problem on my own? I would never have
guessed that on my own and my searches had not found it either.
Thanks again,
-Brian
In this particular case,
running asterisk 1.4.13...
I've noticed with SOME email clients that the timestamp reported from
voicemail is 5 hours off (difference of EST vs UTC). That is, a voicemail
received at 15:17:59 EST is sent via email with a Date: header of 10:17:59
- 0500. An email sent through normal means (mail
Klaus Darilion wrote:
Ok.
Just for the info to others: the 10 seconds are hardcoded in pbx.c
What line in which version?
--
Sean Bright
sean.bri...@gmail.com
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On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
I have just installed a Digium TDM808 (8 fxo port) on an Asterisk
1.6.3. When I try making a call with a .call file, the call goes
straight to the dialplan and start executing the dialplan even before
the called party has pick up.
Steve Howes wrote:
On 20 Jan 2009, at 20:58, Julian Lyndon-Smith wrote:
Using centos 5.2,
I want to use a kickstart file to select packages in order to have an
unattended install onto a bare-metal server.
Is there any way of finding out which packages I need to compile,
build
and
Is there any way of going around this??? Any tricks, configuration hacks??
On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote:
On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
I have just installed a Digium TDM808 (8 fxo port) on an Asterisk
1.6.3. When I try
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the
It seems to me that $FAXFILE lives in the 6403 context, not the n, so it
would still be useable. If this isn't the case, I'd call an AGI script in
the I context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Tue, 20 Jan 2009, Julian Lyndon-Smith wrote:
Using centos 5.2,
I want to use a kickstart file to select packages in order to have an
unattended install onto a bare-metal server.
I started (but have not finished) a live pen drive install. I've snipped out
some bits specific to this
On Tue, 20 Jan 2009, Ken D'Ambrosio wrote:
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems
On Tue, 20 Jan 2009, Danny Nicholas wrote:
It seems to me that $FAXFILE lives in the 6403 context, not the n, so it
would still be useable. If this isn't the case, I'd call an AGI script
in the I context.
$FAXFILE is a channel variable. It is useable in any context, extension
and priority
Anyone have an example XML file for the PAP2T?
Cheers,
j
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If your provider provides any signalling to indicate answer, such as a
polarity reversal, this could be detected easily...
; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party.
;
;answeronpolarityswitch=yes
This isn't very common though... alternatively, there
I'm not sure if this trick will work with this device, but I was able to
pull down a spa8000's config by connecting to:
http://ipaddress/admin/spacfg.xml
Tom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Jan 20, 2009, at 8:58 AM, Philipp Kempgen wrote:
John Todd schrieb:
So here are some projects you might look into:
- open-source ZRTP implementation
libzrtpcpp is open source. GPL.
http://www.gnu.org/software/ccrtp/
Philipp Kempgen
Excellent! I didn't know that existed.
With:
callprogress=yes
and
progzone=us
it works fine for not, not 100% percent because in some calls,it takes like
3-4 seconds before executing dialplan, which is not bad not to say normal.
But most calls are ok.
And when I tried with
answeronpolarityswitch=yes
it doesnt do dialplan at all,
Hello,
We're using Asterisk to manage call queues. Queue members are
connected via IAX2 using the Zoiper softphone, and Zoiper is
configured with 2 lines.
We're finding that calls are routed to queue members even when they
are on the phone, on their softphone's other line. For example, if a
Hi,
take a look at the rininuse setting - if set to yes than you have the
behaviour you described. If set to no - then you will get nearly the
behaviour you want.
Try upgrading to 1.4.22 if using queues - some concerns regarding the
member status have been fixed there.
regards,
Wolfgang
JOB TITLE: VOIP SOFTWARE DEVELOPER
LOCATION: Zurich/Switzerland
JOB TYPE: Full time
COMPANY: Peoplefone AG is a Pan-European VoIP provider
present in 4 countries and headquartered in Zurich/Switzerland (
Sean Bright schrieb:
Klaus Darilion wrote:
Ok.
Just for the info to others: the 10 seconds are hardcoded in pbx.c
What line in which version?
at least in 1.4.22:
grep -r -A 10 -B 10 Auto fallthrough *
klaus
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