From: Unavailable ID [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Thu, 4 Mar 2004 16:33:12 -0800
Subject: [Asterisk-Users] Asterisk crashed so often
Reply-To: [EMAIL PROTECTED]
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On Thu, Mar 04, 2004 at 04:32:52PM -0500, Clif Jones wrote:
I know a little history on the 3com SIP phones... We have about a dozen
of them
where I work. I'm not familiar with the NBX100 model number but the ones we
have are labeled: P/N: 655005001. The first ones didn't support SIP out
Hello,
IMHO you have a problem with the hardware that Asterisk runs on.
You should really look around because there are a number of companies
selling intel based systems with a cPCI bus fully hot swap capable.
I think the only problem would be getting network adapters compatible with
* but then
Stephen R. Besch wrote:
I have now tested a (previously suggested) method for doing supervised
transfers using the Grandstream SIP phone. It isn't perfect, but it
works and is very functional. Here are the steps:
When I try this, all goes well until, after putting the original caller
on hold
Brian Capouch wrote:
Stephen R. Besch wrote:
I have now tested a (previously suggested) method for doing
supervised transfers using the Grandstream SIP phone. It isn't
perfect, but it works and is very functional. Here are the steps:
When I try this, all goes well until, after putting the
Phillipp,
thank you for your help.
At 00:57 04.03.2004 +0100, Philipp von Klitzing wrote:
Hi!
exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
Use this instead:
exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
this works, in the meantime I do understand also why it did send the
how to disable this DEBUG information...
I am getting this on Asterisk CLI
---
Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22,
channel 4
Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:2765
Hello,
I'm trying to configure asterisk with 2 controller (one is a AVM fxusb,
the other is AVM fcpci). Each controller is bound to a separate BRI ISDN
line.
The two modules are correctly loaded and configured by capinit:
# capiinit status
1 fxusb running fritz-usbA1 3.10-02 2
2
On Fri, 2004-03-05 at 12:18, atif wrote:
how to disable this DEBUG information...
I would have intuitively said 'zap no debug' but apparently the 'no
debug' is not implemented for zap although it exist for sip, iax, h323,
skinny and mgcp. Should we consider this absence as a bug worthy of a
On 2004 Mar 05, at 05:18, atif wrote:
how to disable this DEBUG information...
I am getting this on Asterisk CLI
---
Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception:
Exception on 22, channel 4
Edit
On 2004 Mar 04, at 16:12, Philipp von Klitzing wrote:
#0 0x080570a6 in ast_queue_frame (chan=0x810a770, fin=0x41dfd0cc,
lock=1)
at channel.c:368
368 cur = chan-pvt-readq;
(gdb) bt
Post a 'bt full' to bugs.digium.com.
-Tilghman
___
Greetings All!
I just found Asterisk and am new this list.
I have a home office that I have an old DOS based IVR Telephone System
(Telepro) running on an old 386 pc and Dialogic D41D board. I would like
to upgrade
to something new and came across Asterisk. I have read the documentation
and
Tilghman Lesher wrote:
On 2004 Mar 05, at 05:18, atif wrote:
how to disable this DEBUG information...
I am getting this on Asterisk CLI
---
Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception:
Exception on 22,
Hi,
While terminating calls to Cisco 5300 the called party hears converstion
all OK.
However, calling party hears periodic short bursts of interferance
and/or lost packets noise.
I can see on CLI this:
Mar 5 14:35:53 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference
is 3760, ms is 490
Mar
On 2004 Mar 05, at 08:28, John Fraizer wrote:
Tilghman Lesher wrote:
On 2004 Mar 05, at 05:18, atif wrote:
how to disable this DEBUG information...
I am getting this on Asterisk CLI
---
Mar 5 16:18:17 DEBUG[426001]:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nicolas Bougues
Sent: Friday, March 05, 2004 3:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 3com NBX phones
[...]
Note that the hardware is probably not the same as the
standard
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it
to register, but when I dial into a number as soon as voice is to be
connected I get a Warning 399 SDP body missing message and then a BYE
disconnecting the call. The setup I have works great with Xten's x-pro, but
can't
Greets,
Apologies for the OT post. I'm working with a Snom 105 and can't seem to
find the Administrator's Manual for this phone on Snom's website. Does
anyone know where to find this document? Anyone know how to perform a
factory reset on this device? After upgrading the firmware to 2.03o,
it
The IR device is a 3rd-party piece of hardware from Extended System (now
owned by
iFoundry). The SIP phone looks like all of the other 3com IP phones
that I have seen
and turning it over with the front of the phone facing up the connectors
go from left to
right as follows:
1. Handset connector
On Fri, Mar 05, 2004 at 10:25:49AM -0500, Clif Jones wrote:
The IR device is a 3rd-party piece of hardware from Extended System (now
owned by
iFoundry). The SIP phone looks like all of the other 3com IP phones
that I have seen
and turning it over with the front of the phone facing up the
Hi everyone,
I
am having problems dialing 9 to get an external line with my SIP
phones or SIP clients. I have been looking for months on websites, sitting in
MIRC rooms, and reading * documentation but I cannot seem to find a solution.
My asterisk box is sitting directly on the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks Sven. I think I've made things worse now :-0 Playing around with
some key combinations while booting the phone, I think I've disabled the
display. I see the Snom logo briefly at the bootloader but after that
nothing is displayed on the panel.
I could not find one either, but I did find one for the 200 series
phones, which seemed similar. That is where I found the default admin
password once I locked myself out. Check the 200 series product page.
On Fri, 2004-03-05 at 07:18, Eric Hendrickson wrote:
Greets,
Apologies for the OT
On Thu, 2004-03-04 at 23:21, Steven Critchfield wrote:
On Fri, 2004-03-05 at 02:04, dkwok wrote:
Has anyone using the flash button on GS101 to access call waiting?
My experience is that it does not work. I read in the list that it may
need to tweak the flash duration to under 100msec.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yes, I can get to the webpage. And the phone plays its happy tune after
it boots up. Just no display on the LCD panel.
Sven Fischer wrote:
| Eric,
|
| the phone is working ? Can you access the webpage ?
|
| regards,
|
| Sven
|
| On Friday 05 March
Hi
everyone,
Is there something
that needs to be done with RedHat 9 kernel prior to installing TE410P and
loading wct4xxp zap module? I mean, is there a kernel patch or something else
that must be installed, or is it enough just to compile zaptel
driver?
thanks
Tomica
Hello list,
I'm getting droped calls on an asterisk installation. When on GS phone
dials another one, the call is dropped after some (usually random) time
but most of the tome within 3 to 20 seconds.
I think the cause is stated on the logs, see bellow, and is related with
the message Didn't get
compiling zaptel is enough,
I have one and it loaded up just fine on RH9, make sure you modprobe zaptel,
modprobe wct4xxp and then ztcfg -vvv before you try starting the first time.
MATT---
-Original Message-
From: Tomica Crnek [mailto:[EMAIL PROTECTED]
Sent: Friday, March 05, 2004
Situation:
SIP phone A calls Asterisk.
Asterisk forwards to another SIP agent B.
The SIP agent B forwards A back to an Asterisk extension that is mapped
to a TDM400P channel
Once all of this has transpired, there is no audio channel between the
SIP phone A and the TDM400P.
If A is not registered
From a brief look, it seems you do not have a context= in your sip.conf
file for the extension. If you don't put a contxt in, I don't know what
it assumes, and it will not include the contexts you have set up to
define external access.
From looking at your dialplan, if you put context=local
What sort of phone are you trying to call?
I use SIPPS and * and it works fine, it just wont work when you call
Windows Messenger for some reason. I can call X-lite, POTS, GS phones no
problems.
I use the same config as X-lite, in SIPPS if you click on the spanner or
press F9 to go into
Derek,
Can you use fax with G.729? I know that only ULAW codec can use for fax but
I don't know that if you can use fax with G.729 or not.
BTW, what service provider that you are using? Quality can sometime depend
on provider too.
Thanks.
- Original Message -
From: Derek Samford
There is new firmware that may help http://www.grandstream.com/BETATEST/.
Grandstream acknowledges this problem. They say it is a codec issue with
asterisk. I don't know if this update addresses this problem but it may be
worth a try.
-Original Message-
From: [EMAIL PROTECTED]
I'll try to look over my config again. Not sure I put a realm in, but
everything else seemed fine. I get the acquired message and I see the SIP
messages flowing on the Asterisk server, but once any sound needs to be
sent, it dies. I'm using g711ulaw and I was calling into an announcement
menu
I'll try to look over my config again. Not sure I put a realm in, but
everything else seemed fine. I get the acquired message and I see the SIP
messages flowing on the Asterisk server, but once any sound needs to be
sent, it dies. I'm using g711ulaw and I was calling into an announcement
menu
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it
to register, but when I dial into a number as soon as voice is to be
connected I get a Warning 399 SDP body missing message and then a BYE
disconnecting the call. The setup I have works great with Xten's x-pro, but
can't
Hello,
I'll try that, but why on earth gs phones with the same firmware on
another * server, work with no problem?
I've failed to state I'm using zaprtc, since there is no digium hardware
on the server. Does it matter?
Thanks,
--- Paulo.
On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote:
Hello,
IMHO you have a problem with the hardware that Asterisk runs on.
You should really look around because there are a number of companies
selling intel based systems with a cPCI bus fully hot swap capable.
I think the only problem would be getting network adapters compatible with
* but then
I'm at a little bit of a loss here. I'm going to enclose my SIP output for
this session that hopefully someone knows why I get the SDP not available
message when using SIPPS to Asterisk. It registers great and when I call
SIPPS it rings, but when it answers I get the same problem with the SDP
I have couple of GS phone and CISCO 7960.
The funny thing is that two of that GS phone keep disconnecting and also
CISCO 7960 phone keeps disconnecting.
But the problem appear month ago! This is really strange!
Bart
Hello,
I'll try that, but why on earth gs phones with the same firmware on
Tim,
It looks interesting.. Are you willing to release the source code?
Robert
Tim Sailer said:
On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote:
Since there's not too much out there, I decided to take about 2 hrs and
pound something into shape for a simple status for my * server.
We are interested in deploying some ADSI phones, but we are currently
using Sipura SPA-2000s. Everything I read on ADSI, says that it is
just audio, and you don't need a special channel bank or anything like
that.
But what about through the SPA? I notice that zapata.conf appears to
have an
I've had some small problems when trying to users features like
AbsoluteTimeout with pass thru. Other than that sound quality has been good.
Wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Unavailable
ID
Sent: Thursday, March 04, 2004 9:03 PM
To: [EMAIL
On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote:
Tim,
It looks interesting.. Are you willing to release the source code?
Sure. let me clean it up a bit... OK, a LOT... and finish the comments,
and I'll have a download link for it sometime this weekend. I'll keep
the
Your problem is what I experience with Messenger, when I call it.
Unfortunately I never bothered trying to work out the problem.
I like the SIPPS phone features, but it is ugly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlton J.
O'Riley
Sent: 05
That is pretty cool.
I watched the light bulbs for a while now. This is a useful tool that has
many possibilities. That Tim guy is on the phone more than my teenage
daughter :)
calvis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Nice one thanks for sharing, I look forward to it.
This will be very handy for SIP call transfers. At the moment I blindly
transfer on sip.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: 05 March 2004 19:49
To: [EMAIL PROTECTED]
Is anyone presently using the Sipura SPA 2000 for faxing? I was about
to look into it and just figured that I would ask to see if anyone ran
into any snags, problems, etc. Thanks.
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Fri, 5 Mar 2004, Mark Messmore, Technical Support, University Telcom Inc. wrote:
Is anyone presently using the Sipura SPA 2000 for faxing? I was about
to look into it and just figured that I would ask to see if anyone ran
into any snags, problems, etc. Thanks.
I'm really trying. :)
It
hi
It works, sort of. Basically, about 1 in 4 faxes are going out without
errors. Of course, that's to an IAX peer, so I'm not sure if it's a
problem with the IAX peer or with the Siupra.
check you IAX connection.
perhaps is using gsm and that could explain the failure
Faxes must be sent
Hi Wes,
Do you need to buy license when you are using pass thru. How does it work?
I'm thinking about using pass thru for voip since the service provider has
g.279 codec. Can you setup your * box connects to telco termination with
pass thru?
PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP
I am also looking forward to it.
It looks really nice!
bart
- Original Message -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 05, 2004 4:57 PM
Subject: RE: [Asterisk-Users] Simple * status
Nice one thanks for sharing, I look forward to it.
This
On Fri, 5 Mar 2004, Brancaleoni Matteo wrote:
check you IAX connection. perhaps is using gsm and that could explain
the failure Faxes must be sent uncompressed, ie with [u-a]law as codecs.
Forgot to mention that - I've tried both ulaw and alaw all the way through
(sipura - asterisk, and
yes be sure you are using ULAW and I found that 9600 was the baud rate to go
with. 14400 seemed to be unreliable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: Friday, March 05, 2004 2:15 PM
I'm new to asterisk and quite impressed by the feature list. I have a
D/4PCI already in hand. Is there any reason NOT to use this and buy a
digium card instead?
I basically want to set up a couple line analog system to check it out and
probably use as a a Soho setup for VM, access to a postgres
Yes, I do something like that.
MediatrixFXO(1204)-Asterisk-MediatrixFXO(1204), I have bought license from
diguim for G.729. I do not really have a telco provider just an ISP. I use
if for a private network.
Wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
On Fri, 5 Mar 2004, Justin Carlson wrote:
yes be sure you are using ULAW and I found that 9600 was the baud rate
to go with. 14400 seemed to be unreliable.
Are you actually faxing over the 'net?
I'm using ULAW both from the Sipura - Asterisk and from Asterisk -
NuFone (IAX), and have the
Alfred,
I am in similar position. I took the route of going with Digium boards for
two primary reasons - they have excellent quality and offer the best in
customer and technical support. I am starting with an X100P just for testing
everything out (using IAXClient and Asterisk).
We have two
Hello *,
my setup of an Asterisk box with a TDM400P and an E100P went just
fine except that I cannot manage to place outgoing calls via the E1
interface (while incoming calls _do_ work). The CLI says:
-8-
Mar 6 00:42:25 DEBUG[-1147995216]: chan_sip.c:3593 build_route:
build_route:
When an NBX100 is upgraded a .tar file is uploaded and installed on the box.
Inside that tar file is the firmware for the phones which is downloaded when
the phone boots. If someone can provide the last SIP firmware I will
replace the phone firmware in the tar file with the SIP code and see if
Has anyone implemented distinctive ring for SIP devices in Asterisk? My
searches revealed that there was a patch created at one time but I can't
tell if it was accepted or not.
Basically I have a Sipura analog adapter that I would like to have ring
differently for internal calls vs external
On Fri, 2004-03-05 at 21:04, Matt McIntyre wrote:
Has anyone implemented distinctive ring for SIP devices in Asterisk? My
searches revealed that there was a patch created at one time but I can't
tell if it was accepted or not.
Basically I have a Sipura analog adapter that I would like to
Hi Alfred-
I'd like to echo Kaydon's very positive comments regarding asterisk and
Digium with one or two caveats. I've had lots of experience building
systems with Dialogic boards (analog and E1), and more recently a few
systems built with Digium's quad E1 boards (the E400P and now the TE410P),
Hey, all. I'm new to this Asterisk stuff and have a general concept
question about making calls and whatnot over the net.
I have a 4-port TDM card and a 1-port x100p card for incoming. All is
configured and working fine. I have a _very_ simple configuration (start
simple, add bells and whistles
Andrew McRory wrote:
I can offer some links that helped me...
[...]
If anyone has other links I'd appreciate them!
Don't forget http://www.asteriskdocs.org/ !
Nick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have another question for the group. I'm trying to make the following
happen on my Cisco phone:
I have two lines configured, 2001 and 3001. If I'm talking on 2001 and
someone tries to call me on 2001 I'd like the call to roll over to 3001 and
then if I don't answer, it goes to Voice mail.
Hermann Wecke wrote:
After trying and trying to compile and make Asterisk run on a Debian
box, I gave up and picked another HD with RH 9 on it. No headaches. Only 1
build was necessary to build and run *.
The problem I found with debian is how they decided how to do the linux
header files for
I'm having a problem with transferring a call that comes in a Zap
channel and is connected with a SIP channel (on a GS HT-286).
The call is answered automatically, then the user enters an extension.
Dial() is called with both T and t flags. When the bridge is made
between the channels, the
Hi Greg,
Welcome to * world :-)
Your connection is slow '128k and upload speed of 32k' so you probably need
the G.729 codec ($$$ - $10/channel/call from Digium).
The X100P is only for dial-out from your phones that connect to TDM card.
This should use to dial local number in Costa Rica.
To
It works on grandstream handytone 286 also, i just tested both directions and it
worked perfectly first time going into a fax
machine and into a windows xp machine all of this over my flaky wireless link too!
Panny
- Original Message -
From: Brancaleoni Matteo [EMAIL PROTECTED]
To:
At 05:04 PM 3/5/2004, you wrote:
I have another question for the group. I'm trying to make the following
happen on my Cisco phone:
I have two lines configured, 2001 and 3001. If I'm talking on 2001 and
snip
Try this
exten = 2001,1,Dial(SIP/2001SIP/3001,20)
This will ring them both at the same
On Fri, Mar 05, 2004 at 06:45:32PM -0800, Unavailable ID wrote:
Your connection is slow '128k and upload speed of 32k' so you probably need
the G.729 codec ($$$ - $10/channel/call from Digium).
Yeah, I know... it's slow. But, I am in a developing country, and I
I'm a tightwad (don't wanna
Maybe you are using inband DTMF with a compressed codec. DTMF on a call
with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or
INFO.
On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
I'm having a problem with transferring a call that comes in a Zap
channel and is connected
I'm using SIP INFO and ulaw. It seems that the same thing happens
from SIP to SIP as well, regardless of what the DTMF setting is. The
actual problem is that the calling user can transfer, but the called
user cannot. I just tried the latest CVS snapshot and the v1.0 stable
branch and they both
What is your ACTUAL Dial line?
On Fri, 2004-03-05 at 21:19, Barton Hodges wrote:
I'm using SIP INFO and ulaw. It seems that the same thing happens
from SIP to SIP as well, regardless of what the DTMF setting is. The
actual problem is that the calling user can transfer, but the called
user
exten = s,10,Dial(${ARG1}/${DIALED},19,Ttm) which translates to
Dial(SIP/210-80f2, SIP/280|19|Ttm)
I believe the problem is related to the Grandstream HandyTone-286. A
caller can transfer, but a callee cannot. The problem does not exist
with a BT101 (1.0.4.23). I just tried all of the
Scott Stingel wrote:
If you'll be running commercial apps, I would recommend that you do a lot of
testing, especially load testing, with the types of applications you'll be
running. Dialogic boards, although incredibly expensive, do have lots of
horsepower built in for the purposes of encoding
Hi,
I installed 'gnophone' on my notePC(RHL9 linux-2.4.20-30.9)
with asterisk CVS-02/05/04. I have three unsolved problems:
(1)call from gnophone to sip phone is OK, but gnophone's
speaker volume is very low even though setting highest
volume with gmix, the speaker volume is very high.
Anyone been able to get the conference feature on the 7960's to work without
using meetme ?
I get - warning, chan_sip.c:2103 process_sdp: No compatible codecs!
Thanks,
Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Howdy -
I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck.
Right now, the setup is
Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover - TE410P
Right now, the CSU/DSU-ish WorldDSL box has a green light indicating E1
sync, but the TE410P shows a red alarm. I checked
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