Displaying 20 results from an estimated 10000 matches similar to: "SIP call interrupted after 64 seconds"
2006 Mar 27
2
How to disable event_log?
Hi,
how can I disable event_log in order to reduce
hard disk activity?
I can't find any hints in conf files.
Must I hack the source code or even use brutal
methods like creating a dir called event_log in
the log dir, in order to prevent asterisk from
creating an event_log file? (Just chmod a-w event_log does not
work, unfortunately.)
Thanks for any hints!
Roger.
2004 Aug 27
2
how to fetch a call?
Hi,
there is a feature, which I would like to use with asterisk,
and I assume it exists.
Unfortunately I don't know how to say it in english.
In german it's "einen Ruf heranholen".
It means:
The phone set of my collegue is ringing, and I'm hearing
the ringing.
I know, that my collegue is not at his desk, and now
I want to answer the call at my phone (instead of
running to
2005 Jan 03
3
oh323 context for peers
I am experimenting with oh323 channels and h.323 gateways and a Cisco
CallManager. I am not using a gatekeeper at this time. Is it possible to
place calls coming into Asterisk from specific peers into specific
contexts?
In iax.conf eaxh peer has a context in which I can specify the context an
inbound call will be placed in. I don't see anything like this in the
oh323.conf file or the oh323
2006 Mar 24
2
How to nice agi scripts?
Hi,
I have unpleasent short audio gaps when a
perl based agi scripts starts.
Thus, I now started to put all those things in C programmed
daemons for fast-agi.
Anyway I'm looking for another mean, which would help me
more quickly.
I noticed, that all agi scripts are running with system
priority -11, like asterisk does. This is really waste of
priority. I would like to have the AGI scripts
2006 Mar 02
5
Milliwatt Analyzer available
Hi,
some days ago we discused here the need for an analyzer
for the 1000 Hz tone, as opposite application to Milliwatt.
Here it is: Mwanalyze
http://planinternet.net/download/voip/asterisk/app_mwanalyze.c
It performs a Fourier analysis for a fixed frequency
and tells the amplitude.
The frequency is not limited to 1000 Hz, but can be passed
as argument. The periode duration must be a mulitple
2007 Jul 31
2
Connecting GSM Phone to Asterisk Box
Hi All,
I have a telephony project for which I need
to build a prototype to demo for management.
The prototype must work on a GSM phone network.
In the demo system, a call from GSM phone comes
into the demo box. The demo box runs CallWeaver.
Callweaver picks up the GSM call, answers it and
plays a sould file, then dials out to a second GSM
phone somewhere and connects them so they talk.
My
2006 Jun 12
2
No reinvite - reason?
Hi,
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Thus, there should be no reason not to reinvite.
Call (sip, authenticated) comes in and is forward
via SIP (not authenticated) to another asterisk box.
Unfortunately, media path still passes through the asterisk
box in the middle.
Using sip debug I even can't find
2004 Aug 06
1
Problems loading chan_h323 on Opteron 64 bit
Hi,
I compiled asterisk and chan_h323 on an Opteron in 64 bit mode.
In the h323's Makefile I replaced in line 24
CFLAGS += -march=$(shell uname -m)
by
CFLAGS += -march=k8
and also tried
CFLAGS += -m64 -march=k8
Both solutions do compile, but when starting asterisk,
a load error occurs:
undefined symbol:
_ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi
When I grep
2007 Oct 11
3
Distributed FAX - How to best complement asterisk ?
Hi list,
I'm evaluating a private telephony scenario of about 20
locations - 300 phones, 50 FAX machines.
Initial overview points to the installation of asterisk at three
locations connected to the PSTN via ISDN PRI.
All other locations, small by themselves, would get SIP
phones managed by asterisk, since there is good IP
connectivity between all sites.
Now on to the
2009 Apr 20
1
T38 fax failing
Fax over T38 is failing, on the same system it worked with Callweaver.
What do I need to post to be get further assistance please?
2004 Jul 16
2
Offhook tone in channel OSS/dsp
Hi,
I have to develop a phone application using asterisk's
chan_oss.
When the phone is idle, i.e. the last command was a hangup,
one hears a "toot, toot, toot, ..."
But unforuntaly its use is in Germany, where one expects
a continous "toooooooooooooooooooooooooooooooooo ..."
before dialing.
Is there anything to define the tone indicating
"ready to dial"?
2004 Aug 25
1
chan_oh323: __use_ast_pthread_create_instead__ (was: chan_oh323 loading error)
Hi,
> chan_oh323.so: undefined
> symbol: __use_ast_pthread_create_instead__
is not a bug, it's a hint:
use "ast_pthread_create" instead [what your were using]
and means:
replace in asterisk-oh/asterisk-driver/chan_oh323.c
at line 3764
"pthread_create"
by
"ast_pthread_create"
Roger.
2007 Feb 05
2
Howto use PRI lines (E1 or T1) for "data calls"?
Hi,
I'm looking for a mean to send digital data over
an E1 line, just like isdn4linux or Capi via AVM's FritzCard
is able to do it with BRI lines (e.g. for PPP or ISDN raw
connections).
I'm not looking for modulated audio data representing
digital data, like fax or the analogue modems of former
times. I want an interface to the ISDN raw data, with
an outgoing call marked as
2012 Jan 03
4
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi,
Please help me understand the following applications and what are its
advantages if we compare between each of them.
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Regards,
Kaushal
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2010 Apr 08
1
Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
All,
I am looking at a little support on this, as I haven't found it on
google yet. I have had this work on Callweaver, but am moving to
Asterisk for a variety of reasons. My dial plans, and everything else
transferred perfectly, though I am not sure they are 'correct' for
Asterisk 1.6.1, with simple things like SIP users outlined in the
sip.conf file, not in the users file,
2003 May 27
8
[OF] Cable Pinouts
Hi,
Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection?
thanks
Eduardo
2008 Dec 04
2
Packet size limit for HDLC?
Hi,
I'm using app_pppd with a Digium-PRI-card for PPP connections.
I had some strange problems with some IP packets passing
and some not, e.g. ftp login went well, but as soon as
I tried to up- or download a file, noting was transferred.
I finally guessed, it must have to do something with the packet
size. Then I started pppd with the parameters mtu 296 and mru 296
as in further times with
2009 Apr 20
1
Asterisk 'outgoing' directory
Can this be used in the same way as Callweaver works, IE: to invoke Sendfax by
placing (using mv command) a job description file in it?
Michael
2008 May 21
1
T38 fax solution with Asterisk possible?
Hi,
I am looking for a very low cost way of receiving and sending T38 fax
reliably. Is there any possible solution using Asterisk as the PSTN SIP
gateay and Digium E1/T1 card? Is there other open source package that can
help to accomplish this purpose?
Regards,
Mark
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2008 Mar 08
3
replace astdb with a cluster-capable sql database engine
I've been searching the Internet for information
regarding the replacement of astdb with a modern sql
engine.
There are several reasons one would like to do this.
First of all, external applications have a hard time
reading/writing to the now-old astdb format.
Also (and this is what interests me most), the sql
astdb could easily be clustered throughout several
servers (I'm looking for a