Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Tony Mountifield]

In article <20180404133024.kpidrkuiyjoqd...@xorcom.com>,
Tzafrir Cohen  wrote:
> On Wed, Apr 04, 2018 at 11:28:33AM +, Tony Mountifield wrote:
> > In article 
> > ,
> > Richard Mudgett  wrote:
> > > 
> > > The libpri makefile doesn't install things for 64 bit systems in the right
> > > place [1] without your help.  You'll need to specify where to install the
> > > library on the command line for your system:
> > > 
> > > sudo make install libdir=/usr/lib64
> > > 
> > > 
> > > Richard
> > > 
> > > [1] https://issues.asterisk.org/jira/browse/PRI-100
> > 
> > Ah, thanks. I did in fact discover the following 64-bit libraries were
> > installed into /usr/lib instead of /usr/lib64:
> > 
> > 1. From DAHDI, libtonezone.so
> 
> dahdi-tools 2.11 now uses autoconf. It still installs to /usr/lib or is
> it an older version?

I compiled dahdi-linux-complete-2.11.1+2.11.1, by doing:

make
make install
make config
make -C tools install-config

In fact I did all the above with a DESTDIR=$DESTDIR appended to each line,
as I was building a binary bundle for system building.

Before doing so I also did:

mkdir -p $DESTDIR/etc/udev/rules.d $DESTDIR/etc/rc.d/init.d 
$DESTDIR/etc/sysconfig/network-scripts

Maybe that defeated autoconf, and made it default to /usr/lib?
If I had also done a mkdir $DESTDIR/usr/lib64 before building, maybe
autoconf would have found it?

But in any case, running ldconfig made everything get found:

[root@bridge05 ~]# ldconfig -p | fgrep -v lib64
552 libs found in cache `/etc/ld.so.cache'
libtonezone.so.2 (libc6,x86-64) => /usr/lib/libtonezone.so.2
libtonezone.so (libc6,x86-64) => /usr/lib/libtonezone.so
libpri.so.1.4 (libc6,x86-64) => /usr/lib/libpri.so.1.4
libpri.so (libc6,x86-64) => /usr/lib/libpri.so
libasteriskssl.so.1 (libc6,x86-64) => /usr/lib/libasteriskssl.so.1
libasteriskssl.so (libc6,x86-64) => /usr/lib/libasteriskssl.so
[root@bridge05 ~]#

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Tzafrir Cohen]

On Wed, Apr 04, 2018 at 11:28:33AM +, Tony Mountifield wrote:
> In article 
> ,
> Richard Mudgett  wrote:
> > 
> > The libpri makefile doesn't install things for 64 bit systems in the right
> > place [1] without your help.  You'll need to specify where to install the
> > library on the command line for your system:
> > 
> > sudo make install libdir=/usr/lib64
> > 
> > 
> > Richard
> > 
> > [1] https://issues.asterisk.org/jira/browse/PRI-100
> 
> Ah, thanks. I did in fact discover the following 64-bit libraries were
> installed into /usr/lib instead of /usr/lib64:
> 
> 1. From DAHDI, libtonezone.so

dahdi-tools 2.11 now uses autoconf. It still installs to /usr/lib or is
it an older version?

> 
> 2. From LibPRI, libpri.so
> 
> 3. From Asterisk, libasteriskssl.so
> 
> I found that running "ldconfig" caused them all to be discovered:
> 
> [root@bridge05 ~]# ldd /usr/sbin/asterisk
> linux-vdso.so.1 =>  (0x7ffc77ff9000)
> libasteriskssl.so.1 => /usr/lib/libasteriskssl.so.1 
> (0x7efeae1d4000)
> libc.so.6 => /lib64/libc.so.6 (0x7efeade4)
> libxml2.so.2 => /usr/lib64/libxml2.so.2 (0x7efeadaed000)
> libz.so.1 => /lib64/libz.so.1 (0x7efead8d7000)
> libm.so.6 => /lib64/libm.so.6 (0x7efead653000)
> libsqlite3.so.0 => /usr/lib64/libsqlite3.so.0 (0x7efead3c4000)
> libssl.so.10 => /usr/lib64/libssl.so.10 (0x7efead158000)
> libcrypto.so.10 => /usr/lib64/libcrypto.so.10 (0x7efeacd73000)
> libdl.so.2 => /lib64/libdl.so.2 (0x7efeacb6f000)
> libpthread.so.0 => /lib64/libpthread.so.0 (0x7efeac952000)
> libtinfo.so.5 => /lib64/libtinfo.so.5 (0x7efeac731000)
> libresolv.so.2 => /lib64/libresolv.so.2 (0x7efeac517000)
> /lib64/ld-linux-x86-64.so.2 (0x7efeae3d6000)
> libgssapi_krb5.so.2 => /lib64/libgssapi_krb5.so.2 (0x7efeac2d3000)
> libkrb5.so.3 => /lib64/libkrb5.so.3 (0x7efeabfec000)
> libcom_err.so.2 => /lib64/libcom_err.so.2 (0x7efeabde8000)
> libk5crypto.so.3 => /lib64/libk5crypto.so.3 (0x7efeabbbc000)
> libkrb5support.so.0 => /lib64/libkrb5support.so.0 (0x7efeab9b1000)
> libkeyutils.so.1 => /lib64/libkeyutils.so.1 (0x7efeab7ae000)
> libselinux.so.1 => /lib64/libselinux.so.1 (0x7efeab58f000)
> [root@bridge05 ~]# ldd /usr/sbin/dahdi_cfg
> linux-vdso.so.1 =>  (0x7fff6cbaa000)
> libtonezone.so.2 => /usr/lib/libtonezone.so.2 (0x7f862f74a000)
> libpthread.so.0 => /lib64/libpthread.so.0 (0x7f862f52d000)
> libm.so.6 => /lib64/libm.so.6 (0x7f862f2a9000)
> libc.so.6 => /lib64/libc.so.6 (0x7f862ef15000)
> /lib64/ld-linux-x86-64.so.2 (0x7f862f97e000)
> [root@bridge05 ~]# ldd /usr/lib/asterisk/modules/chan_dahdi.so
> linux-vdso.so.1 =>  (0x7ffe8b1df000)
> libtonezone.so.2 => /usr/lib/libtonezone.so.2 (0x7f54adde4000)
> libpri.so.1.4 => /usr/lib/libpri.so.1.4 (0x7f54adb68000)
> libpthread.so.0 => /lib64/libpthread.so.0 (0x7f54ad94b000)
> libc.so.6 => /lib64/libc.so.6 (0x7f54ad5b7000)
> libm.so.6 => /lib64/libm.so.6 (0x7f54ad333000)
> /lib64/ld-linux-x86-64.so.2 (0x7f54ae2d3000)
> [root@bridge05 ~]#
> 
> So I assumed that all should be ok, otherwise the executables would fail to 
> run
> (I initially discovered this when dahdi_cfg couldn't find libtonezone).
> 
> Would there be any subtle issues with the 64-bit libraries being loaded
> from /usr/lib instead of /usr/lib64?
> 
> Should Asterisk and DAHDI builds also be updated to use /usr/lib64 when
> building on a 64-bit OS? Or the build instructions?

dahdi-tools: not AFAIK.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Tony Mountifield]

In article ,
Tony Mountifield  wrote:
> In article 
> 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Tony Mountifield]

In article ,
Richard Mudgett  wrote:
> 
> The libpri makefile doesn't install things for 64 bit systems in the right
> place [1] without your help.  You'll need to specify where to install the
> library on the command line for your system:
> 
> sudo make install libdir=/usr/lib64
> 
> 
> Richard
> 
> [1] https://issues.asterisk.org/jira/browse/PRI-100

Ah, thanks. I did in fact discover the following 64-bit libraries were
installed into /usr/lib instead of /usr/lib64:

1. From DAHDI, libtonezone.so

2. From LibPRI, libpri.so

3. From Asterisk, libasteriskssl.so

I found that running "ldconfig" caused them all to be discovered:

[root@bridge05 ~]# ldd /usr/sbin/asterisk
linux-vdso.so.1 =>  (0x7ffc77ff9000)
libasteriskssl.so.1 => /usr/lib/libasteriskssl.so.1 (0x7efeae1d4000)
libc.so.6 => /lib64/libc.so.6 (0x7efeade4)
libxml2.so.2 => /usr/lib64/libxml2.so.2 (0x7efeadaed000)
libz.so.1 => /lib64/libz.so.1 (0x7efead8d7000)
libm.so.6 => /lib64/libm.so.6 (0x7efead653000)
libsqlite3.so.0 => /usr/lib64/libsqlite3.so.0 (0x7efead3c4000)
libssl.so.10 => /usr/lib64/libssl.so.10 (0x7efead158000)
libcrypto.so.10 => /usr/lib64/libcrypto.so.10 (0x7efeacd73000)
libdl.so.2 => /lib64/libdl.so.2 (0x7efeacb6f000)
libpthread.so.0 => /lib64/libpthread.so.0 (0x7efeac952000)
libtinfo.so.5 => /lib64/libtinfo.so.5 (0x7efeac731000)
libresolv.so.2 => /lib64/libresolv.so.2 (0x7efeac517000)
/lib64/ld-linux-x86-64.so.2 (0x7efeae3d6000)
libgssapi_krb5.so.2 => /lib64/libgssapi_krb5.so.2 (0x7efeac2d3000)
libkrb5.so.3 => /lib64/libkrb5.so.3 (0x7efeabfec000)
libcom_err.so.2 => /lib64/libcom_err.so.2 (0x7efeabde8000)
libk5crypto.so.3 => /lib64/libk5crypto.so.3 (0x7efeabbbc000)
libkrb5support.so.0 => /lib64/libkrb5support.so.0 (0x7efeab9b1000)
libkeyutils.so.1 => /lib64/libkeyutils.so.1 (0x7efeab7ae000)
libselinux.so.1 => /lib64/libselinux.so.1 (0x7efeab58f000)
[root@bridge05 ~]# ldd /usr/sbin/dahdi_cfg
linux-vdso.so.1 =>  (0x7fff6cbaa000)
libtonezone.so.2 => /usr/lib/libtonezone.so.2 (0x7f862f74a000)
libpthread.so.0 => /lib64/libpthread.so.0 (0x7f862f52d000)
libm.so.6 => /lib64/libm.so.6 (0x7f862f2a9000)
libc.so.6 => /lib64/libc.so.6 (0x7f862ef15000)
/lib64/ld-linux-x86-64.so.2 (0x7f862f97e000)
[root@bridge05 ~]# ldd /usr/lib/asterisk/modules/chan_dahdi.so
linux-vdso.so.1 =>  (0x7ffe8b1df000)
libtonezone.so.2 => /usr/lib/libtonezone.so.2 (0x7f54adde4000)
libpri.so.1.4 => /usr/lib/libpri.so.1.4 (0x7f54adb68000)
libpthread.so.0 => /lib64/libpthread.so.0 (0x7f54ad94b000)
libc.so.6 => /lib64/libc.so.6 (0x7f54ad5b7000)
libm.so.6 => /lib64/libm.so.6 (0x7f54ad333000)
/lib64/ld-linux-x86-64.so.2 (0x7f54ae2d3000)
[root@bridge05 ~]#

So I assumed that all should be ok, otherwise the executables would fail to run
(I initially discovered this when dahdi_cfg couldn't find libtonezone).

Would there be any subtle issues with the 64-bit libraries being loaded
from /usr/lib instead of /usr/lib64?

Should Asterisk and DAHDI builds also be updated to use /usr/lib64 when
building on a 64-bit OS? Or the build instructions?

Regards
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Tony Mountifield]

In article 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Antony Stone]

On Wednesday 04 April 2018 at 00:30:00, Richard Mudgett wrote:

> On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson  wrote:
> > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield 
> > > Both the 32-bit and 64-bit were fresh installs of the latest CentOS 6.9
> > > from online repositories using a kickstart build.
> 
> The libpri makefile doesn't install things for 64 bit systems in the right
> place [1] without your help.  You'll need to specify where to install the
> library on the command line for your system:
> 
> sudo make install libdir=/usr/lib64
> 
> [1] https://issues.asterisk.org/jira/browse/PRI-100

Isn't this handled by the CentOS package manager?


Antony.

-- 
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Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Richard Mudgett]

On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson  wrote:

> On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield 
> wrote:
> > In article 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Matt Fredrickson]

On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield  wrote:
> In article 
> 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Tony Mountifield]

In article 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Matt Fredrickson]

On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield  wrote:
> I have some more investigation to do on this, but I wanted to see if anyone
> here had any insight into the issue I've run into.
>
> The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one
> of several systems that have been running without issue since 2010/2011. They
> have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen
> 5 card), libpri 1.2.8 and asterisk 1.2.32.
>
> Having taken this particular system out of production, I updated it to CentOS
> 6.9 32-bit, with DAHDI 2.11.1, LibPRI 1.6.0 and Asterisk 11.25.3 (this version
> of Asterisk is required at the moment due to custom modifications).
> This appears to work fine.
>
> In order to reduce the number of different versions we support, I reinstalled
> the OS using the 64-bit version of CentOS 6.9 instead, and rebuilt, using
> the same versions as above.
>
> However, for reasons I don't understand, the 64-bit version was logging
> frequent PRI errors every few minutes:
>
> [Apr  1 03:40:52] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:40:58] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:44:06] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:46:38] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:47:20] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:47:24] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
>
> This left the PRIs in strange states - trying to make a call failed with 
> cause 101.
>
> So I re-installed the 32-bit OS again, and rebuilt, and the above MDL-ERRORs
> were no longer present, and the system operated normally again.
>
> So my question is: does anyone have any clues why there would be a difference
> in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into
> anything similar?


That does seem quite odd.  If I remember right, those messages would
come up if it looked like the other end hadn't received a message when
it thought it should have.  I can't think of anything that would
particularly impact 64 bit systems versus 32 bit systems in that
domain (ISDN real time message timing, etc).  Are you sure there's
nothing else different (kernel version or something else like that)?
Maybe also run a patlooptest on the spans in question to make sure
that they're running cleanly.

Matthew Fredrickson

>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Strange problem with PRI on 64-bit?

[Tony Mountifield]

I have some more investigation to do on this, but I wanted to see if anyone
here had any insight into the issue I've run into.

The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one
of several systems that have been running without issue since 2010/2011. They
have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen
5 card), libpri 1.2.8 and asterisk 1.2.32.

Having taken this particular system out of production, I updated it to CentOS
6.9 32-bit, with DAHDI 2.11.1, LibPRI 1.6.0 and Asterisk 11.25.3 (this version
of Asterisk is required at the moment due to custom modifications).
This appears to work fine.

In order to reduce the number of different versions we support, I reinstalled
the OS using the 64-bit version of CentOS 6.9 instead, and rebuilt, using
the same versions as above.

However, for reasons I don't understand, the 64-bit version was logging
frequent PRI errors every few minutes:

[Apr  1 03:40:52] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR (A): 
Got supervisory frame with F=1 in state 7(Multi-frame established)
[Apr  1 03:40:58] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR (A): 
Got supervisory frame with F=1 in state 7(Multi-frame established)
[Apr  1 03:44:06] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR (A): 
Got supervisory frame with F=1 in state 7(Multi-frame established)
[Apr  1 03:46:38] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR (A): 
Got supervisory frame with F=1 in state 7(Multi-frame established)
[Apr  1 03:47:20] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR (A): 
Got supervisory frame with F=1 in state 7(Multi-frame established)
[Apr  1 03:47:24] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR (A): 
Got supervisory frame with F=1 in state 7(Multi-frame established)

This left the PRIs in strange states - trying to make a call failed with cause 
101.

So I re-installed the 32-bit OS again, and rebuilt, and the above MDL-ERRORs
were no longer present, and the system operated normally again.

So my question is: does anyone have any clues why there would be a difference
in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into
anything similar?

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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[asterisk-users] Strange problem with Asterisk 1.8.9.3

[Dereck D]

Hello List.
Last month i started to face a strange issue on an asterisk server
1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
out of the blue UDP stops responding .. and keep getting the following output:

-- Opening message for the problem
--
[Mar 21 09:57:04] ERROR[6748] netsock2.c:
getaddrinfo(pbx2.server.com, (null), ...): Temporary failure in
name resolution
[Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup 'pbx2.server.com'
THEN
-
[Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
UNREACHABLE!  Last qualify: 10
[Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'C1045' is now
UNREACHABLE!  Last qualify: 140
[Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
UNREACHABLE!  Last qualify: 33
[Mar 21 09:57:15] NOTICE[6748] chan_sip.c:-- Registration for
'8885...@pbx2.server.com' timed out, trying again (Attempt #3)
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C3000' is now
UNREACHABLE!  Last qualify: 87
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5005' is now
UNREACHABLE!  Last qualify: 241
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1071' is now
UNREACHABLE!  Last qualify: 117
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1070' is now
UNREACHABLE!  Last qualify: 115
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1077' is now
UNREACHABLE!  Last qualify: 101
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1073' is now
UNREACHABLE!  Last qualify: 96
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1047' is now
UNREACHABLE!  Last qualify: 132
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1072' is now
UNREACHABLE!  Last qualify: 138
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1042' is now
UNREACHABLE!  Last qualify: 158
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5006' is now
UNREACHABLE!  Last qualify: 267
[Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1053' is now
UNREACHABLE!  Last qualify: 136
[Mar 21 09:57:41] NOTICE[6748] chan_sip.c:-- Registration for
'8885...@pbx2.server.com' timed out, trying again (Attempt #3)
[Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'ipkall1' is now
UNREACHABLE!  Last qualify: 168
[Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1078' is now
UNREACHABLE!  Last qualify: 141
[Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1051' is now
UNREACHABLE!  Last qualify: 139
[Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1046' is now
UNREACHABLE!  Last qualify: 157
[Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1076' is now
UNREACHABLE!  Last qualify: 116


this problem kept repeating randomly two or three times a day causing
me losses. and then i migrated to another Centos 6x x86_64 dedicated
server same asterisk version.
everything returns to normal if i reboot the server.  for a moment i
thought i wasn't even able to nslookup but i could wget or yum or do
other stuff..although i have my iptables disabled for a the last week
for testing purposes.
what i did today after the problem is:
bindaddress: set to server IPv4 address instead of defaulting.
disabled IPv6 from the server.


any suggestions to what is causing this issue?

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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Re: [asterisk-users] Strange problem with Asterisk 1.8.9.3

[dotnetdub]

Looks like a DNS issue.


On 21 April 2013 11:05, Dereck D derec...@gmail.com wrote:

 Hello List.
 Last month i started to face a strange issue on an asterisk server
 1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
 out of the blue UDP stops responding .. and keep getting the following
 output:

 -- Opening message for the problem
 --
 [Mar 21 09:57:04] ERROR[6748] netsock2.c:
 getaddrinfo(pbx2.server.com, (null), ...): Temporary failure in
 name resolution
 [Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup 'pbx2.server.com'
 THEN
 -
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
 UNREACHABLE!  Last qualify: 10
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'C1045' is now
 UNREACHABLE!  Last qualify: 140
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
 UNREACHABLE!  Last qualify: 33
 [Mar 21 09:57:15] NOTICE[6748] chan_sip.c:-- Registration for
 '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C3000' is now
 UNREACHABLE!  Last qualify: 87
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5005' is now
 UNREACHABLE!  Last qualify: 241
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1071' is now
 UNREACHABLE!  Last qualify: 117
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1070' is now
 UNREACHABLE!  Last qualify: 115
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1077' is now
 UNREACHABLE!  Last qualify: 101
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1073' is now
 UNREACHABLE!  Last qualify: 96
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1047' is now
 UNREACHABLE!  Last qualify: 132
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1072' is now
 UNREACHABLE!  Last qualify: 138
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1042' is now
 UNREACHABLE!  Last qualify: 158
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5006' is now
 UNREACHABLE!  Last qualify: 267
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1053' is now
 UNREACHABLE!  Last qualify: 136
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c:-- Registration for
 '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'ipkall1' is now
 UNREACHABLE!  Last qualify: 168
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1078' is now
 UNREACHABLE!  Last qualify: 141
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1051' is now
 UNREACHABLE!  Last qualify: 139
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1046' is now
 UNREACHABLE!  Last qualify: 157
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1076' is now
 UNREACHABLE!  Last qualify: 116


 this problem kept repeating randomly two or three times a day causing
 me losses. and then i migrated to another Centos 6x x86_64 dedicated
 server same asterisk version.
 everything returns to normal if i reboot the server.  for a moment i
 thought i wasn't even able to nslookup but i could wget or yum or do
 other stuff..although i have my iptables disabled for a the last week
 for testing purposes.
 what i did today after the problem is:
 bindaddress: set to server IPv4 address instead of defaulting.
 disabled IPv6 from the server.


 any suggestions to what is causing this issue?

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Strange problem with Asterisk 1.8.9.3

[Dereck D]

thank you , but how would it be DNS issue while other users drop
unreachable too?
all operators and SIP Peers go unreachable .. not only unable to register.
oe peer using FQDN the rest are IP addresses.

On Sun, Apr 21, 2013 at 2:18 PM, dotnetdub dotnet...@gmail.com wrote:
 Looks like a DNS issue.


 On 21 April 2013 11:05, Dereck D derec...@gmail.com wrote:

 Hello List.
 Last month i started to face a strange issue on an asterisk server
 1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
 out of the blue UDP stops responding .. and keep getting the following
 output:

 -- Opening message for the problem
 --
 [Mar 21 09:57:04] ERROR[6748] netsock2.c:
 getaddrinfo(pbx2.server.com, (null), ...): Temporary failure in
 name resolution
 [Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup 'pbx2.server.com'
 THEN
 -
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
 UNREACHABLE!  Last qualify: 10
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'C1045' is now
 UNREACHABLE!  Last qualify: 140
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
 UNREACHABLE!  Last qualify: 33
 [Mar 21 09:57:15] NOTICE[6748] chan_sip.c:-- Registration for
 '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C3000' is now
 UNREACHABLE!  Last qualify: 87
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5005' is now
 UNREACHABLE!  Last qualify: 241
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1071' is now
 UNREACHABLE!  Last qualify: 117
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1070' is now
 UNREACHABLE!  Last qualify: 115
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1077' is now
 UNREACHABLE!  Last qualify: 101
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1073' is now
 UNREACHABLE!  Last qualify: 96
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1047' is now
 UNREACHABLE!  Last qualify: 132
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1072' is now
 UNREACHABLE!  Last qualify: 138
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1042' is now
 UNREACHABLE!  Last qualify: 158
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5006' is now
 UNREACHABLE!  Last qualify: 267
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1053' is now
 UNREACHABLE!  Last qualify: 136
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c:-- Registration for
 '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'ipkall1' is now
 UNREACHABLE!  Last qualify: 168
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1078' is now
 UNREACHABLE!  Last qualify: 141
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1051' is now
 UNREACHABLE!  Last qualify: 139
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1046' is now
 UNREACHABLE!  Last qualify: 157
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1076' is now
 UNREACHABLE!  Last qualify: 116


 this problem kept repeating randomly two or three times a day causing
 me losses. and then i migrated to another Centos 6x x86_64 dedicated
 server same asterisk version.
 everything returns to normal if i reboot the server.  for a moment i
 thought i wasn't even able to nslookup but i could wget or yum or do
 other stuff..although i have my iptables disabled for a the last week
 for testing purposes.
 what i did today after the problem is:
 bindaddress: set to server IPv4 address instead of defaulting.
 disabled IPv6 from the server.


 any suggestions to what is causing this issue?

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



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 _
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 asterisk-users mailing list
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Re: [asterisk-users] Strange problem with Asterisk 1.8.9.3

[Dereck D]

ok it seems a bug in asterisk .. work around is to add the FQDNS to
the hosts file and try to setup local DNS

On Sun, Apr 21, 2013 at 2:24 PM, Dereck D derec...@gmail.com wrote:
 thank you , but how would it be DNS issue while other users drop
 unreachable too?
 all operators and SIP Peers go unreachable .. not only unable to register.
 oe peer using FQDN the rest are IP addresses.

 On Sun, Apr 21, 2013 at 2:18 PM, dotnetdub dotnet...@gmail.com wrote:
 Looks like a DNS issue.


 On 21 April 2013 11:05, Dereck D derec...@gmail.com wrote:

 Hello List.
 Last month i started to face a strange issue on an asterisk server
 1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
 out of the blue UDP stops responding .. and keep getting the following
 output:

 -- Opening message for the problem
 --
 [Mar 21 09:57:04] ERROR[6748] netsock2.c:
 getaddrinfo(pbx2.server.com, (null), ...): Temporary failure in
 name resolution
 [Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup 'pbx2.server.com'
 THEN
 -
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
 UNREACHABLE!  Last qualify: 10
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'C1045' is now
 UNREACHABLE!  Last qualify: 140
 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
 UNREACHABLE!  Last qualify: 33
 [Mar 21 09:57:15] NOTICE[6748] chan_sip.c:-- Registration for
 '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C3000' is now
 UNREACHABLE!  Last qualify: 87
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5005' is now
 UNREACHABLE!  Last qualify: 241
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1071' is now
 UNREACHABLE!  Last qualify: 117
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1070' is now
 UNREACHABLE!  Last qualify: 115
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1077' is now
 UNREACHABLE!  Last qualify: 101
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1073' is now
 UNREACHABLE!  Last qualify: 96
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1047' is now
 UNREACHABLE!  Last qualify: 132
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1072' is now
 UNREACHABLE!  Last qualify: 138
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1042' is now
 UNREACHABLE!  Last qualify: 158
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5006' is now
 UNREACHABLE!  Last qualify: 267
 [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1053' is now
 UNREACHABLE!  Last qualify: 136
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c:-- Registration for
 '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'ipkall1' is now
 UNREACHABLE!  Last qualify: 168
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1078' is now
 UNREACHABLE!  Last qualify: 141
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1051' is now
 UNREACHABLE!  Last qualify: 139
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1046' is now
 UNREACHABLE!  Last qualify: 157
 [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1076' is now
 UNREACHABLE!  Last qualify: 116


 this problem kept repeating randomly two or three times a day causing
 me losses. and then i migrated to another Centos 6x x86_64 dedicated
 server same asterisk version.
 everything returns to normal if i reboot the server.  for a moment i
 thought i wasn't even able to nslookup but i could wget or yum or do
 other stuff..although i have my iptables disabled for a the last week
 for testing purposes.
 what i did today after the problem is:
 bindaddress: set to server IPv4 address instead of defaulting.
 disabled IPv6 from the server.


 any suggestions to what is causing this issue?

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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To 

Re: [asterisk-users] Strange problem with Asterisk 1.8.9.3

[dotnetdub]

I don't know with this version but certainly the SIP stack in earlier
version would get blocked when it couldn't do a name resolution and cause
symptoms exactly as you describe. It shouldn't happen but there you go.


On 21 April 2013 12:24, Dereck D derec...@gmail.com wrote:

 thank you , but how would it be DNS issue while other users drop
 unreachable too?
 all operators and SIP Peers go unreachable .. not only unable to register.
 oe peer using FQDN the rest are IP addresses.

 On Sun, Apr 21, 2013 at 2:18 PM, dotnetdub dotnet...@gmail.com wrote:
  Looks like a DNS issue.
 
 
  On 21 April 2013 11:05, Dereck D derec...@gmail.com wrote:
 
  Hello List.
  Last month i started to face a strange issue on an asterisk server
  1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
  out of the blue UDP stops responding .. and keep getting the following
  output:
 
  -- Opening message for the problem
  --
  [Mar 21 09:57:04] ERROR[6748] netsock2.c:
  getaddrinfo(pbx2.server.com, (null), ...): Temporary failure in
  name resolution
  [Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup '
 pbx2.server.com'
  THEN
  -
  [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
  UNREACHABLE!  Last qualify: 10
  [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'C1045' is now
  UNREACHABLE!  Last qualify: 140
  [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer '' is now
  UNREACHABLE!  Last qualify: 33
  [Mar 21 09:57:15] NOTICE[6748] chan_sip.c:-- Registration for
  '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C3000' is now
  UNREACHABLE!  Last qualify: 87
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5005' is now
  UNREACHABLE!  Last qualify: 241
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1071' is now
  UNREACHABLE!  Last qualify: 117
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1070' is now
  UNREACHABLE!  Last qualify: 115
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1077' is now
  UNREACHABLE!  Last qualify: 101
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1073' is now
  UNREACHABLE!  Last qualify: 96
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1047' is now
  UNREACHABLE!  Last qualify: 132
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1072' is now
  UNREACHABLE!  Last qualify: 138
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1042' is now
  UNREACHABLE!  Last qualify: 158
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C5006' is now
  UNREACHABLE!  Last qualify: 267
  [Mar 21 09:57:21] NOTICE[6748] chan_sip.c: Peer 'C1053' is now
  UNREACHABLE!  Last qualify: 136
  [Mar 21 09:57:41] NOTICE[6748] chan_sip.c:-- Registration for
  '8885...@pbx2.server.com' timed out, trying again (Attempt #3)
  [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'ipkall1' is now
  UNREACHABLE!  Last qualify: 168
  [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1078' is now
  UNREACHABLE!  Last qualify: 141
  [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1051' is now
  UNREACHABLE!  Last qualify: 139
  [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1046' is now
  UNREACHABLE!  Last qualify: 157
  [Mar 21 09:57:41] NOTICE[6748] chan_sip.c: Peer 'C1076' is now
  UNREACHABLE!  Last qualify: 116
 
 
  this problem kept repeating randomly two or three times a day causing
  me losses. and then i migrated to another Centos 6x x86_64 dedicated
  server same asterisk version.
  everything returns to normal if i reboot the server.  for a moment i
  thought i wasn't even able to nslookup but i could wget or yum or do
  other stuff..although i have my iptables disabled for a the last week
  for testing purposes.
  what i did today after the problem is:
  bindaddress: set to server IPv4 address instead of defaulting.
  disabled IPv6 from the server.
 
 
  any suggestions to what is causing this issue?
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

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 _
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Re: [asterisk-users] Strange problem on ougoing call

[Olivier CALVANO]

Perfect that's work ;=)

very thanks



Le 25 avril 2012 10:19, Olivier CALVANO o.calv...@gmail.com a écrit :
 Ok thanks i test.

 I put match_auth_username=yes on the two server ?

 And for insecure, into the realtime database ? or into sip.conf of the
 second server ?

 best regards
 olivier



 Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :


 2012/4/25 Olivier CALVANO o.calv...@gmail.com

 Sure, sorry for the Confusion ;=)




 Server A Trader:
       Asterisk Server 1.6.x for call routing only.
       IP Adress: 172.16.0.11
       Use Realtim on MySQL Database
       This server route all call to a lot of VoIP Carrier.


 Server B Ipbx
       Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
       IP Adress: 172.16.0.70
       Second IP: 172.16.1.70 (used for phone lan)
       Use Realtim on MySQL Database
       This server route all call to a lot of VoIP Carrier.


 Linksys SPA942 A:
      IP Adress: 172.16.1.200
      Connected in SIP at Server B IPBX
      use sip.conf (no realtime)
      context: I-User01


 Linksys SPA942 B:
      IP Adress: 172.16.1.220
      Connected in SIP at Server B IPBX
      use sip.conf (no realtime)
      context: I-User02



 On Server A Trader, we have two sip account:
      accountname: USER01 for user in group 1
      accountname: USER02 for user in group 2

 On Server B Ipbx, i use registry:
     register = USER01:1234@172.16.0.11/USER01
     register = USER02:5678@172.16.0.11/USER02
 for two connection to the Trader Server. Registry is good:
 on server A Trader:

 trader*CLI sip show registry
 Host                           dnsmgr Username       Refresh State
          Reg.Time
 172.16.0.11:5060               N      USER01     105 Registered
  Tue, 24 Apr 2012 15:58:58
 172.16.0.11:5060               N      USER02       105 Registered
    Tue, 24 Apr 2012 15:58:59


 On server B Ipbx, i have into my sip.conf after the registry:

 [USER01]
 type=friend
 username=USER01
 secret=1234
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 [USER02]
 type=friend
 username=USER02
 secret=5678
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 and in extensions.conf:

 [I-User01]
 exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)

 [I-User02]
 exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)







 When i call with Linksys SPA942 A, i use the context I-User01 and
 the call are sent
 to SIP account USER01 and  No problems.

 When i call with Linksys SPA942 B, i use the context I-User02 and
 the call are sent
 to SIP account USER02 but Server A Trader reject the call
 immediatly with this error:

 [Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
 mismatch, have USER01, digest has USER02
 [Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
 handle_request_invite: Failed to authenticate device Olivier
 sip:906280@172.16.0.70;tag=as0cd775ab

 Olivier and 906280 is the information that i have on the Linksys
 SPA942 B, 906280 is the username used between




 best ? hihi
 Olivier





 Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
  Hi,
  Lots of mixing and confusing stuff - Can you re-explain the topology you
  are
  trying to achieve with proper IP addresses and declared sip ext. names.
 
  When i call with the phone connected to I-User01, no problems, that's
  work but when i call
  with the second phone (use I-User02) i have a error:
 
 
  Somehow it reminds of the same situation I always face when a peer is
  declared with the same name as of the dialing one on second server -
  only
  Its just not registered there instead registered on server-1.
  So when the call comes in from server-1 to server-2 fromuser=olivier
   which
  is not registered on server-2 but is declared. Server-2 thinks that this
  is
  my valid extension but it is not registered with me and so lets
  authenticate
  this one and here it fails and rejects the call.
 
  BR,
  Sammy.
 
  On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
  wrote:
 
  Hi
 
  i have a strange problems on my asterisk server:
 
  I have two asterisk server.
 
  On the first, i use realtime with a MySQL Database,
  i have two user:
    USER01
    USER02
  exactly the same configuration only username and password has
  different.
 
 
  On my second server (phone is connected on this server):
 
  I have in sip.conf:
 
  register = USER01:1234@172.16.0.11/USER01
  register = USER02:5678@172.16.0.11/USER02
 
  [USER01]
  type=friend
  username=USER01
  secret=1234
  host=172.16.0.11
  qualify=yes
  dtmf=rfc2833
  nat=no
  canreinvite=no
  canredirect=no
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  context=I-User01
  musiconhold=default
  insecure=port,invite
 
  

Re: [asterisk-users] Strange problem on ougoing call

[Olivier CALVANO]

Sure, sorry for the Confusion ;=)




Server A Trader:
   Asterisk Server 1.6.x for call routing only.
   IP Adress: 172.16.0.11
   Use Realtim on MySQL Database
   This server route all call to a lot of VoIP Carrier.


Server B Ipbx
   Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
   IP Adress: 172.16.0.70
   Second IP: 172.16.1.70 (used for phone lan)
   Use Realtim on MySQL Database
   This server route all call to a lot of VoIP Carrier.


Linksys SPA942 A:
  IP Adress: 172.16.1.200
  Connected in SIP at Server B IPBX
  use sip.conf (no realtime)
  context: I-User01


Linksys SPA942 B:
  IP Adress: 172.16.1.220
  Connected in SIP at Server B IPBX
  use sip.conf (no realtime)
  context: I-User02



On Server A Trader, we have two sip account:
  accountname: USER01 for user in group 1
  accountname: USER02 for user in group 2

On Server B Ipbx, i use registry:
 register = USER01:1234@172.16.0.11/USER01
 register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:

trader*CLI sip show registry
Host   dnsmgr Username   Refresh State
  Reg.Time
172.16.0.11:5060   N  USER01 105 Registered
  Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060   N  USER02   105 Registered
Tue, 24 Apr 2012 15:58:59


On server B Ipbx, i have into my sip.conf after the registry:

[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite

[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite

and in extensions.conf:

[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)

[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)







When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and  No problems.

When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:

[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab

Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between




best ? hihi
Olivier





Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
 Hi,
 Lots of mixing and confusing stuff - Can you re-explain the topology you are
 trying to achieve with proper IP addresses and declared sip ext. names.

 When i call with the phone connected to I-User01, no problems, that's
 work but when i call
 with the second phone (use I-User02) i have a error:


 Somehow it reminds of the same situation I always face when a peer is
 declared with the same name as of the dialing one on second server - only
 Its just not registered there instead registered on server-1.
 So when the call comes in from server-1 to server-2 fromuser=olivier  which
 is not registered on server-2 but is declared. Server-2 thinks that this is
 my valid extension but it is not registered with me and so lets authenticate
 this one and here it fails and rejects the call.

 BR,
 Sammy.

 On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
 wrote:

 Hi

 i have a strange problems on my asterisk server:

 I have two asterisk server.

 On the first, i use realtime with a MySQL Database,
 i have two user:
   USER01
   USER02
 exactly the same configuration only username and password has different.


 On my second server (phone is connected on this server):

 I have in sip.conf:

 register = USER01:1234@172.16.0.11/USER01
 register = USER02:5678@172.16.0.11/USER02

 [USER01]
 type=friend
 username=USER01
 secret=1234
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 [USER02]
 type=friend
 username=USER02
 secret=5678
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite


 i see the registration:

 ipbx*CLI sip show registry
 Host                           dnsmgr Username       Refresh State
           Reg.Time
 172.16.0.11:5060               N      USER01     105 Registered
   Tue, 24 Apr 2012 15:58:58
 172.16.0.11:5060               N      USER02       105 Registered
     Tue, 24 Apr 

Re: [asterisk-users] Strange problem on ougoing call

[Leandro Dardini]

2012/4/25 Olivier CALVANO o.calv...@gmail.com

 Sure, sorry for the Confusion ;=)




 Server A Trader:
   Asterisk Server 1.6.x for call routing only.
   IP Adress: 172.16.0.11
   Use Realtim on MySQL Database
   This server route all call to a lot of VoIP Carrier.


 Server B Ipbx
   Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
   IP Adress: 172.16.0.70
   Second IP: 172.16.1.70 (used for phone lan)
   Use Realtim on MySQL Database
   This server route all call to a lot of VoIP Carrier.


 Linksys SPA942 A:
  IP Adress: 172.16.1.200
  Connected in SIP at Server B IPBX
  use sip.conf (no realtime)
  context: I-User01


 Linksys SPA942 B:
  IP Adress: 172.16.1.220
  Connected in SIP at Server B IPBX
  use sip.conf (no realtime)
  context: I-User02



 On Server A Trader, we have two sip account:
  accountname: USER01 for user in group 1
  accountname: USER02 for user in group 2

 On Server B Ipbx, i use registry:
  register = USER01:1234@172.16.0.11/USER01
 register = USER02:5678@172.16.0.11/USER02
 for two connection to the Trader Server. Registry is good:
 on server A Trader:

 trader*CLI sip show registry
 Host   dnsmgr Username   Refresh State
  Reg.Time
 172.16.0.11:5060   N  USER01 105 Registered
  Tue, 24 Apr 2012 15:58:58
 172.16.0.11:5060   N  USER02   105 Registered
Tue, 24 Apr 2012 15:58:59


 On server B Ipbx, i have into my sip.conf after the registry:

 [USER01]
 type=friend
 username=USER01
 secret=1234
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 [USER02]
 type=friend
 username=USER02
 secret=5678
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 and in extensions.conf:

 [I-User01]
 exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)

 [I-User02]
 exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)







 When i call with Linksys SPA942 A, i use the context I-User01 and
 the call are sent
 to SIP account USER01 and  No problems.

 When i call with Linksys SPA942 B, i use the context I-User02 and
 the call are sent
 to SIP account USER02 but Server A Trader reject the call
 immediatly with this error:

 [Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
 mismatch, have USER01, digest has USER02
 [Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
 handle_request_invite: Failed to authenticate device Olivier
 sip:906280@172.16.0.70;tag=as0cd775ab

 Olivier and 906280 is the information that i have on the Linksys
 SPA942 B, 906280 is the username used between




 best ? hihi
 Olivier





 Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
  Hi,
  Lots of mixing and confusing stuff - Can you re-explain the topology you
 are
  trying to achieve with proper IP addresses and declared sip ext. names.
 
  When i call with the phone connected to I-User01, no problems, that's
  work but when i call
  with the second phone (use I-User02) i have a error:
 
 
  Somehow it reminds of the same situation I always face when a peer is
  declared with the same name as of the dialing one on second server - only
  Its just not registered there instead registered on server-1.
  So when the call comes in from server-1 to server-2 fromuser=olivier
  which
  is not registered on server-2 but is declared. Server-2 thinks that this
 is
  my valid extension but it is not registered with me and so lets
 authenticate
  this one and here it fails and rejects the call.
 
  BR,
  Sammy.
 
  On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
  wrote:
 
  Hi
 
  i have a strange problems on my asterisk server:
 
  I have two asterisk server.
 
  On the first, i use realtime with a MySQL Database,
  i have two user:
USER01
USER02
  exactly the same configuration only username and password has different.
 
 
  On my second server (phone is connected on this server):
 
  I have in sip.conf:
 
  register = USER01:1234@172.16.0.11/USER01
  register = USER02:5678@172.16.0.11/USER02
 
  [USER01]
  type=friend
  username=USER01
  secret=1234
  host=172.16.0.11
  qualify=yes
  dtmf=rfc2833
  nat=no
  canreinvite=no
  canredirect=no
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  context=I-User01
  musiconhold=default
  insecure=port,invite
 
  [USER02]
  type=friend
  username=USER02
  secret=5678
  host=172.16.0.11
  qualify=yes
  dtmf=rfc2833
  nat=no
  canreinvite=no
  canredirect=no
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  context=I-User01
  musiconhold=default
  insecure=port,invite
 
 
  i see the registration:
 
  ipbx*CLI sip show registry
  Host   dnsmgr Username   Refresh 

Re: [asterisk-users] Strange problem on ougoing call

[Olivier CALVANO]

Ok thanks i test.

I put match_auth_username=yes on the two server ?

And for insecure, into the realtime database ? or into sip.conf of the
second server ?

best regards
olivier



Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :


 2012/4/25 Olivier CALVANO o.calv...@gmail.com

 Sure, sorry for the Confusion ;=)




 Server A Trader:
       Asterisk Server 1.6.x for call routing only.
       IP Adress: 172.16.0.11
       Use Realtim on MySQL Database
       This server route all call to a lot of VoIP Carrier.


 Server B Ipbx
       Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
       IP Adress: 172.16.0.70
       Second IP: 172.16.1.70 (used for phone lan)
       Use Realtim on MySQL Database
       This server route all call to a lot of VoIP Carrier.


 Linksys SPA942 A:
      IP Adress: 172.16.1.200
      Connected in SIP at Server B IPBX
      use sip.conf (no realtime)
      context: I-User01


 Linksys SPA942 B:
      IP Adress: 172.16.1.220
      Connected in SIP at Server B IPBX
      use sip.conf (no realtime)
      context: I-User02



 On Server A Trader, we have two sip account:
      accountname: USER01 for user in group 1
      accountname: USER02 for user in group 2

 On Server B Ipbx, i use registry:
     register = USER01:1234@172.16.0.11/USER01
     register = USER02:5678@172.16.0.11/USER02
 for two connection to the Trader Server. Registry is good:
 on server A Trader:

 trader*CLI sip show registry
 Host                           dnsmgr Username       Refresh State
          Reg.Time
 172.16.0.11:5060               N      USER01     105 Registered
  Tue, 24 Apr 2012 15:58:58
 172.16.0.11:5060               N      USER02       105 Registered
    Tue, 24 Apr 2012 15:58:59


 On server B Ipbx, i have into my sip.conf after the registry:

 [USER01]
 type=friend
 username=USER01
 secret=1234
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 [USER02]
 type=friend
 username=USER02
 secret=5678
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 and in extensions.conf:

 [I-User01]
 exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)

 [I-User02]
 exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)







 When i call with Linksys SPA942 A, i use the context I-User01 and
 the call are sent
 to SIP account USER01 and  No problems.

 When i call with Linksys SPA942 B, i use the context I-User02 and
 the call are sent
 to SIP account USER02 but Server A Trader reject the call
 immediatly with this error:

 [Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
 mismatch, have USER01, digest has USER02
 [Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
 handle_request_invite: Failed to authenticate device Olivier
 sip:906280@172.16.0.70;tag=as0cd775ab

 Olivier and 906280 is the information that i have on the Linksys
 SPA942 B, 906280 is the username used between




 best ? hihi
 Olivier





 Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
  Hi,
  Lots of mixing and confusing stuff - Can you re-explain the topology you
  are
  trying to achieve with proper IP addresses and declared sip ext. names.
 
  When i call with the phone connected to I-User01, no problems, that's
  work but when i call
  with the second phone (use I-User02) i have a error:
 
 
  Somehow it reminds of the same situation I always face when a peer is
  declared with the same name as of the dialing one on second server -
  only
  Its just not registered there instead registered on server-1.
  So when the call comes in from server-1 to server-2 fromuser=olivier
   which
  is not registered on server-2 but is declared. Server-2 thinks that this
  is
  my valid extension but it is not registered with me and so lets
  authenticate
  this one and here it fails and rejects the call.
 
  BR,
  Sammy.
 
  On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
  wrote:
 
  Hi
 
  i have a strange problems on my asterisk server:
 
  I have two asterisk server.
 
  On the first, i use realtime with a MySQL Database,
  i have two user:
    USER01
    USER02
  exactly the same configuration only username and password has
  different.
 
 
  On my second server (phone is connected on this server):
 
  I have in sip.conf:
 
  register = USER01:1234@172.16.0.11/USER01
  register = USER02:5678@172.16.0.11/USER02
 
  [USER01]
  type=friend
  username=USER01
  secret=1234
  host=172.16.0.11
  qualify=yes
  dtmf=rfc2833
  nat=no
  canreinvite=no
  canredirect=no
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  context=I-User01
  musiconhold=default
  insecure=port,invite
 
  [USER02]
  type=friend
  username=USER02
  secret=5678
  host=172.16.0.11
  qualify=yes
  dtmf=rfc2833
  nat=no
  

[asterisk-users] Strange problem on ougoing call

[Olivier CALVANO]

Hi

i have a strange problems on my asterisk server:

I have two asterisk server.

On the first, i use realtime with a MySQL Database,
i have two user:
   USER01
   USER02
exactly the same configuration only username and password has different.


On my second server (phone is connected on this server):

I have in sip.conf:

register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02

[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite

[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite


i see the registration:

ipbx*CLI sip show registry
Host   dnsmgr Username   Refresh State
   Reg.Time
172.16.0.11:5060   N  USER01 105 Registered
   Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060   N  USER02   105 Registered
 Tue, 24 Apr 2012 15:58:59




i have one phone connected to the context I-User01 and another
connected to I-User02

When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:


On the first server:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab


The exten:

On I-User01: exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
On I-User02: exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)



i i change on the I-User02:
 Dial(SIP/USER02/${EXTEN:1},90,r)
in
 Dial(SIP/USER01/${EXTEN:1},90,r)
all call work's.


anyone have a idea ? i think's that i have a error but don't see where

best regards
Olivier

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Re: [asterisk-users] Strange problem on ougoing call

[Olivier CALVANO]

Hi

No idea ?

thanks
Olivier


Le 24 avril 2012 16:06, Olivier CALVANO o.calv...@gmail.com a écrit :
 Hi

 i have a strange problems on my asterisk server:

 I have two asterisk server.

 On the first, i use realtime with a MySQL Database,
 i have two user:
   USER01
   USER02
 exactly the same configuration only username and password has different.


 On my second server (phone is connected on this server):

 I have in sip.conf:

 register = USER01:1234@172.16.0.11/USER01
 register = USER02:5678@172.16.0.11/USER02

 [USER01]
 type=friend
 username=USER01
 secret=1234
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 [USER02]
 type=friend
 username=USER02
 secret=5678
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite


 i see the registration:

 ipbx*CLI sip show registry
 Host                           dnsmgr Username       Refresh State
           Reg.Time
 172.16.0.11:5060               N      USER01     105 Registered
   Tue, 24 Apr 2012 15:58:58
 172.16.0.11:5060               N      USER02       105 Registered
     Tue, 24 Apr 2012 15:58:59




 i have one phone connected to the context I-User01 and another
 connected to I-User02

 When i call with the phone connected to I-User01, no problems, that's
 work but when i call
 with the second phone (use I-User02) i have a error:


 On the first server:
 [Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
 mismatch, have USER01, digest has USER02
 [Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
 handle_request_invite: Failed to authenticate device Olivier
 sip:906280@172.16.0.70;tag=as0cd775ab


 The exten:

 On I-User01: exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
 On I-User02: exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)



 i i change on the I-User02:
     Dial(SIP/USER02/${EXTEN:1},90,r)
 in
     Dial(SIP/USER01/${EXTEN:1},90,r)
 all call work's.


 anyone have a idea ? i think's that i have a error but don't see where

 best regards
 Olivier

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Re: [asterisk-users] Strange problem on ougoing call

[SamyGo]

Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are trying to achieve with proper IP addresses and declared sip ext. names.

When i call with the phone connected to I-User01, no problems, that's
 work but when i call
 with the second phone (use I-User02) i have a error:


Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server - only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier  which
is not registered on server-2 but is declared. Server-2 thinks that this is
my valid extension but it is not registered with me and so lets
authenticate this one and here it fails and rejects the call.

BR,
Sammy.

On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.comwrote:

 Hi

 i have a strange problems on my asterisk server:

 I have two asterisk server.

 On the first, i use realtime with a MySQL Database,
 i have two user:
   USER01
   USER02
 exactly the same configuration only username and password has different.


 On my second server (phone is connected on this server):

 I have in sip.conf:

 register = USER01:1234@172.16.0.11/USER01
 register = USER02:5678@172.16.0.11/USER02

 [USER01]
 type=friend
 username=USER01
 secret=1234
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite

 [USER02]
 type=friend
 username=USER02
 secret=5678
 host=172.16.0.11
 qualify=yes
 dtmf=rfc2833
 nat=no
 canreinvite=no
 canredirect=no
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 context=I-User01
 musiconhold=default
 insecure=port,invite


 i see the registration:

 ipbx*CLI sip show registry
 Host   dnsmgr Username   Refresh State
   Reg.Time
 172.16.0.11:5060   N  USER01 105 Registered
   Tue, 24 Apr 2012 15:58:58
 172.16.0.11:5060   N  USER02   105 Registered
 Tue, 24 Apr 2012 15:58:59




 i have one phone connected to the context I-User01 and another
 connected to I-User02

 When i call with the phone connected to I-User01, no problems, that's
 work but when i call
 with the second phone (use I-User02) i have a error:


 On the first server:
 [Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
 mismatch, have USER01, digest has USER02
 [Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
 handle_request_invite: Failed to authenticate device Olivier
 sip:906280@172.16.0.70;tag=as0cd775ab


 The exten:

 On I-User01: exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
 On I-User02: exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)



 i i change on the I-User02:
 Dial(SIP/USER02/${EXTEN:1},90,r)
 in
 Dial(SIP/USER01/${EXTEN:1},90,r)
 all call work's.


 anyone have a idea ? i think's that i have a error but don't see where

 best regards
 Olivier

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[asterisk-users] Strange problem with zap channel.

[Tim Uckun]

I am trying to help a guy out with his Atcom IP04.  He has set it up like this.

He has a handful of IP phones all connecting via SIP. He has two phone
lines connected to the FXO ports one from telecom, another from
vodaphone.  He has set up the dialplan so that one of the trunks fails
over to the other trunk.  Everything seems to be working OK except for
outgoing calls.  He can call from extension to extension without
problems. If I call in on either of the trunk lines we can have a
normal conversation.

If he calls out to me he can hear me but I can't hear him.  The status
on GUI shows the phone as still ringing even though I picked up and he
can hear me.

Here is a log of one of the calls.   If anybody can offer a clue as to
what the problem might be I'd be grateful.  I looked at the port
definitions and they are set up for NZ signaling (kewl loop).

[Jun  6 13:24:29] DEBUG[4667]: app_macro.c:337 _macro_exec: Executed
application: Dial
-- Executing [1-d...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/6006-015d0004, 16  0 ?1-BUSY|1:1-out|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-BUSY,1)
[Jun  6 13:24:29] DEBUG[4667]: app_macro.c:337 _macro_exec: Executed
application: Gotoif
  == Auto fallthrough, channel 'SIP/6006-015d0004' status is 'BUSY'
-- Executing [9075763...@dlpn_dialplan1:1]
Macro(SIP/6006-015d0004,
trunkdial-failover-0.3|Zap/g1/075763441|Zap/g2/075763441|trunk_1|trunk_2)
in new stack
-- Executing [...@macro-trunkdial-failover-0.3:1]
Set(SIP/6006-015d0004, CALLERID(num)=6498287700) in new stack
[Jun  6 13:31:41] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed
application: Set
-- Executing [...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/6006-015d0004, 1?1-dial|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
[Jun  6 13:31:41] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed
application: GotoIf
-- Executing [1-d...@macro-trunkdial-failover-0.3:1]
Dial(SIP/6006-015d0004, Zap/g1/075763441) in new stack
[Jun  6 13:31:41] DEBUG[4825]: dsp.c:1787 ast_dsp_set_busy_pattern:
dsp busy pattern set to 0,0
[Jun  6 13:31:41] DEBUG[4825]: chan_zap.c:1952 zt_call: Dialing '075763441'
[Jun  6 13:31:41] DEBUG[4825]: chan_zap.c:2028 zt_call: Deferring dialing...
-- Called g1/075763441
[Jun  6 13:31:42] DEBUG[4825]: chan_zap.c: zt_handle_event: Sent
deferred digit string: T075763441w
[Jun  6 13:31:44] DEBUG[4825]: chan_zap.c:3788 zt_handle_event: Done
dialing, but waiting for progress detection before doing more...


At this point I have picked up the phone and am speaking, he can hear
me but I can't hear him.

After I hang up I get this.


[Jun  6 13:32:02] DEBUG[4825]: dsp.c:1445 ast_dsp_busydetect:
ast_dsp_busydetect detected busy, avgtone: 255, avgsilence 240
-- Zap/1-1 is busy
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:1/0/0)
[Jun  6 13:32:02] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed
application: Dial
-- Executing [1-d...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/6006-015d0004, 16  0 ?1-BUSY|1:1-out|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-BUSY,1)
[Jun  6 13:32:02] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed
application: Gotoif
  == Auto fallthrough, channel 'SIP/6006-015d0004' status is 'BUSY'

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[asterisk-users] Strange Problem

[Alexandru Oniciuc]

Hello list!

I've run into a strange problem today and I was hoping that someone here has 
seen this before and maybe can give me a hand:

I'm using asterisk 1.6.0.22 in this config:

(A)PATTON ISDN -(B) ASTERISK - (C)PATTON PRI - PSTN - (D)OTHER PBX

Strange Problem:

USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the 
user makes a selection and gets his call passed to an extension of that PBX 
(USER D), USER D has no sound while USER A hears the voice just fine.

 If USER A makes a direct call to USER D, calling directly his extension, the 
call has audio both ways and its all working fine.
The same thing if USER A calls directly mobile phones or numbers that aren't 
managed by IVRs.

I've verified this with a few PBXs(different manufacturers), and the problem is 
there every time an IVR gets the control of the call.

A sip debug in asterisk confirmed that the SIP Session is not renegotiated when 
the call exits USER's D IVR and ends up to his extension.

Any idea what might be causing this?

Thank you in advance!

Alex
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Re: [asterisk-users] Strange Problem

[Danny Nicholas]

Monitor the successful and failing calls from a CLI session with core set
verbose 5.  This should show you what is different between the two calls.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandru
Oniciuc
Sent: Monday, February 08, 2010 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Strange Problem

 

Hello list!

 

I've run into a strange problem today and I was hoping that someone here has
seen this before and maybe can give me a hand:

 

I'm using asterisk 1.6.0.22 in this config:

 

(A)PATTON ISDN -(B) ASTERISK - (C)PATTON PRI - PSTN - (D)OTHER PBX

 

Strange Problem:

 

USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When
the user makes a selection and gets his call passed to an extension of that
PBX (USER D), USER D has no sound while USER A hears the voice just fine.

 

 If USER A makes a direct call to USER D, calling directly his extension,
the call has audio both ways and its all working fine.

The same thing if USER A calls directly mobile phones or numbers that aren't
managed by IVRs.

 

I've verified this with a few PBXs(different manufacturers), and the problem
is there every time an IVR gets the control of the call.

 

A sip debug in asterisk confirmed that the SIP Session is not renegotiated
when the call exits USER's D IVR and ends up to his extension.

 

Any idea what might be causing this?

 

Thank you in advance!

 

Alex

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[asterisk-users] Strange problem

[Carlos Eduardo Langoni]

Hi folks!

I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0.
Sometimes twice a day, sometimes after 3 days, all sip devices looses
registry, but asterisk doesn't show nothing strange, no error log, and
all calls in E1 trunk continue running, but sending to voicemail.

What could be this problem? What should I research do find a solution?

Thanks a lot!

Carlos

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Re: [asterisk-users] Strange problem

[Danny Nicholas]

It sounds like your SIP devices aren't set up to periodically(frequently)
re-register themselves.  You can resolve this on the device level or have
asterisk poll them for re-registration.  It could also be some sort of
firewall/NAT problem that chops the connections at some interval.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo
Langoni
Sent: Monday, August 31, 2009 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Strange problem

Hi folks!

I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0.
Sometimes twice a day, sometimes after 3 days, all sip devices looses
registry, but asterisk doesn't show nothing strange, no error log, and
all calls in E1 trunk continue running, but sending to voicemail.

What could be this problem? What should I research do find a solution?

Thanks a lot!

Carlos

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Re: [asterisk-users] Strange problem

[Danny Nicholas]

I set mine at 300 (5 minutes).  You might want a higher value if you have
lots of phones, but since I only have 8 at my shop, this causes no
noticeable downside.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo
Langoni
Sent: Monday, August 31, 2009 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Strange problem

Danny,

Thank for your reply.
I'm sure that is not firewall/nat because all sip devices are using a
private class of IP and asterisk has a network adapter with an IP from
the same class/network.

How muchi is a good value for re-register?

Thanks a lot



2009/8/31 Danny Nicholas da...@debsinc.com:
 It sounds like your SIP devices aren't set up to periodically(frequently)
 re-register themselves.  You can resolve this on the device level or have
 asterisk poll them for re-registration.  It could also be some sort of
 firewall/NAT problem that chops the connections at some interval.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos
Eduardo
 Langoni
 Sent: Monday, August 31, 2009 10:33 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Strange problem

 Hi folks!

 I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2
2.1.1.0.
 Sometimes twice a day, sometimes after 3 days, all sip devices looses
 registry, but asterisk doesn't show nothing strange, no error log, and
 all calls in E1 trunk continue running, but sending to voicemail.

 What could be this problem? What should I research do find a solution?

 Thanks a lot!

 Carlos

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Re: [asterisk-users] Strange problem

[Carlos Eduardo Langoni]

Danny,

Thank for your reply.
I'm sure that is not firewall/nat because all sip devices are using a
private class of IP and asterisk has a network adapter with an IP from
the same class/network.

How muchi is a good value for re-register?

Thanks a lot



2009/8/31 Danny Nicholas da...@debsinc.com:
 It sounds like your SIP devices aren't set up to periodically(frequently)
 re-register themselves.  You can resolve this on the device level or have
 asterisk poll them for re-registration.  It could also be some sort of
 firewall/NAT problem that chops the connections at some interval.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo
 Langoni
 Sent: Monday, August 31, 2009 10:33 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Strange problem

 Hi folks!

 I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0.
 Sometimes twice a day, sometimes after 3 days, all sip devices looses
 registry, but asterisk doesn't show nothing strange, no error log, and
 all calls in E1 trunk continue running, but sending to voicemail.

 What could be this problem? What should I research do find a solution?

 Thanks a lot!

 Carlos

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[asterisk-users] Strange problem with VoicemailMain

[mark morreny]

Dear all,

I am having a very strange problem with VoicemailMain.  When using this
application to record unavail, greet, and busy, I an see the corresponding
file gets created in the ../default/SIP # directory.  When pressing 1
to confirm the recorded message, the *.wav file gets deleted from the file
system.

How can this happen?  I can't figure out why.  Is there any option I need to
turned on to enable the recording of greeting?

Thanks alot for all your help.

Regards,
Mark
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Re: [asterisk-users] Strange problem Solved

[Accursio Avona]

Sorry, i had a mistake in my dialplan
  - Original Message - 
  From: Accursio Avona 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, March 10, 2008 6:42 PM
  Subject: [asterisk-users] Strange problem



  Hi All,
  i'm experiencing a strange problem on sip channel. 
  Sometime appens that the sip client ring as if it recieves 3 calls at the 
same time from the same number, even if thre is only a single call.

  I'm experiencing that both on the softphone sjphone and on the sip phone 
Grandstream GXP2000
  an on two asterisk box
  one asterisk v 1.0.7
  the second asterisk 1.2.16
  I have not idea where to start for debug this 
  Someone can help me?
  thank's in advance
  Accursio


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12.17
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[asterisk-users] Strange problem

[Accursio Avona]

Hi All,
i'm experiencing a strange problem on sip channel. 
Sometime appens that the sip client ring as if it recieves 3 calls at the same 
time from the same number, even if thre is only a single call.

I'm experiencing that both on the softphone sjphone and on the sip phone 
Grandstream GXP2000
an on two asterisk box
one asterisk v 1.0.7
the second asterisk 1.2.16
I have not idea where to start for debug this 
Someone can help me?
thank's in advance
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[asterisk-users] Strange problem with latest Asterisk

[Christian]

Hi all,
I'm having a problem with latest version of Asterisk.
When I put someone on hold or if I dial an extension with music on hold the 
call hangs up after a few seconds when MUOH has changed file to play. Any 
thoughts?
Many thanks,
Christian

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[asterisk-users] strange problem in asterisk + mediant2k setup

[satish patel]

Dear all

I have asterisk 1.2 with mediant2k i have create SIP Trunk from 
asterisk 2 mediant and my PRI terminated on Mediant 2000 E1 port now everthing 
is fine but problem is when i call to somebody outside and he/she disconnect my 
phone but my asterisk counitine ringing my SIP Snom phone why ??? if i call to 
outside and mobile or phone would be busy but my IP SNOM Phone give me ruinging 
means i dont understand problem on mediant side or asterisk outgoing call 
working fine but only when some one disconnect call i dont get any message like 
phone is busy or something else but my asterisk phone continue rining 





   
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[asterisk-users] Strange problem with channel allocation

[Jonson Player]

Hello I just settup a realtime mysql table for sip_peers. All peers
(friends) is autenticateing but when i want to initiate a call between them
i got the following error. Someone have some ideea? Thank you.


---Cut Here---

pbx*CLIconsole dial 1014
 == Console is full duplex
   -- Executing [EMAIL PROTECTED]:1] Dial(OSS/dsp, SIP/1014|40|t) in new
stack
[2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:650 mysql_reconnect:
MySQL RealTime: Everything is fine.
[2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014'
   -- Called 1014
[2007-06-03 20:16:10] WARNING[27424]: channel.c:3222
ast_channel_make_compatible: No path to translate from
SIP/1014-081e93c0(256) to OSS/dsp(64)
[2007-06-03 20:16:10] WARNING[27424]: channel.c:3222
ast_channel_make_compatible: No path to translate from
SIP/1014-081e93c0(256) to OSS/dsp(64)

^ ??

[2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758 auto_congest:
Auto-congesting SIP/1014-081e93c0
[2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758 auto_congest:
Auto-congesting SIP/1014-081e93c0
   -- SIP/1014-081e93c0 is circuit-busy
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:210 mysql_log:
cdr_mysql: inserting a CDR record.
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:226 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield)
VALUES ('2007-06-03 20:16:10','','','s','default',
'SIP/1014-081e93c0','','','',8,0,'NO ANSWER',3,'','')
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Executing [EMAIL PROTECTED]:2] VoiceMail(OSS/dsp, u1014) in new stack
 Console call has been answered 
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6798 vm_exec:
Prefixing the mailbox with an option is deprecated ('u1014').
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6798 vm_exec:
Prefixing the mailbox with an option is deprecated ('u1014').
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6799 vm_exec: Please
move all leading options to the second argument.
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6799 vm_exec: Please
move all leading options to the second argument.
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:2854 leave_voicemail:
No entry in voicemail config file for '1014'
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:2854 leave_voicemail:
No entry in voicemail config file for '1014'
   -- Executing [EMAIL PROTECTED]:3] Hangup(OSS/dsp, ) in new stack
 == Spawn extension (default, 1014, 3) exited non-zero on 'OSS/dsp'
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:210 mysql_log:
cdr_mysql: inserting a CDR record.
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:226 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield)
VALUES ('2007-06-03 20:16:10','','','1014','default',
'OSS/dsp','SIP/1014-081e93c0','Hangup','',8,0,'ANSWERED',3,'','')
 Hangup on console 
[2007-06-03 20:16:18] DEBUG[27370]: res_config_mysql.c:650 mysql_reconnect:
MySQL RealTime: Everything is fine.
[2007-06-03 20:16:18] DEBUG[27370]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014'

---And Here---
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[asterisk-users] Strange problem with asterisk

[Vitaly Oborsky]

Situation such. There is an asterisk working as office pbx. 6 fxo - 18
fxs ports. All works perfectly, but some times in a week something
occurs. Could not catch what exactly yet. But symptoms such. The
asterisk infinitely writes the message of a type to broad gullies:
WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers
of channels can change. Because of that that broad gullies get
littered fairly promptly, I have not time to see that occured in an
instant of the beginning of this event. When the asterisk is in such
condition, the appropriating channel does not work, in this case 8.
What can it be?

asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x
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Re: [asterisk-users] Strange problem with asterisk

[Stephen Bosch]

Hi, Vitaly:

Vitaly Oborsky wrote:
 Situation such. There is an asterisk working as office pbx. 6 fxo - 18
 fxs ports. All works perfectly, but some times in a week something
 occurs. Could not catch what exactly yet. But symptoms such. The
 asterisk infinitely writes the message of a type to broad gullies:
 WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers
 of channels can change. Because of that that broad gullies get
 littered fairly promptly, I have not time to see that occured in an
 instant of the beginning of this event. When the asterisk is in such
 condition, the appropriating channel does not work, in this case 8.
 What can it be?
 
 asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x

This looks suspiciously like a Babelfish translation... and I have to
admit it's a bit confusing.

Can you try rewording it? :\

-Stephen-
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Re: [asterisk-users] Strange problem with asterisk

[Tzafrir Cohen]

On Fri, May 11, 2007 at 05:32:33PM +0300, Vitaly Oborsky wrote:
 Situation such. There is an asterisk working as office pbx. 6 fxo - 18
 fxs ports. All works perfectly, but some times in a week something
 occurs. Could not catch what exactly yet. But symptoms such. The
 asterisk infinitely writes the message of a type to broad gullies:
 WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers
 of channels can change. Because of that that broad gullies get
 littered fairly promptly, I have not time to see that occured in an
 instant of the beginning of this event. When the asterisk is in such
 condition, the appropriating channel does not work, in this case 8.
 What can it be?
 
 asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x

Could you try a later verssion of bristuff?

I seem to recall a bug report for chan_zap (or is it Zaptel) with 
exactly those symptoms. I cannot find it right now.
Anybody?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Strange problem

[Dovid B]

Although I dont have an answer I would say to look at the defualt ports and 
see if they are opend on all sides and if NAT is used that it is set 
properly.


- Original Message - 
From: Frederico Madeira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, January 30, 2007 2:58 PM
Subject: [asterisk-users] Strange problem



Hi guys.

I'm working on a VOIP service provider.

We have two customers running asterisk. Customer A and B.

When A call to B everything is ok.
When B call to A the call ring but sip messages didn't arrive on
asterisk A. In my softswitch i see the invite sip message sended to A.
When every other numbers(TDM and SIP) call do A everything is ok.

Have any issue in asterisk that can resolve this problem ?
I'm figuring out with our link provider to see if he has some firewall
rules that can cause this problem

I'm very very confuse becouse the invite message in every time come
from my softswitch with  ip of my softswitch so, why only invite
originate on B side has this problem ?

Thanks.

--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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[asterisk-users] Strange problem

[Frederico Madeira]

Hi guys.

I'm working on a VOIP service provider.

We have two customers running asterisk. Customer A and B.

When A call to B everything is ok.
When B call to A the call ring but sip messages didn't arrive on
asterisk A. In my softswitch i see the invite sip message sended to A.
When every other numbers(TDM and SIP) call do A everything is ok.

Have any issue in asterisk that can resolve this problem ?
I'm figuring out with our link provider to see if he has some firewall
rules that can cause this problem

I'm very very confuse becouse the invite message in every time come
from my softswitch with  ip of my softswitch so, why only invite
originate on B side has this problem ?

Thanks.

--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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Re: [asterisk-users] Strange problem(Munin-node-1.2.4-7)

[Steve Glaus]

Crazy Boy wrote:

Hi,

Sorry to post this in this forum.

I am new to Trixbox. When I am trying to install Trixbox, I am facing 
this problem. First I have installed Trixbox ISO image file from a CD. 
When its rebooting and Asterisk is installing, it is got stucked near 
this below point:


Munin-1.2.4-7
Preparing package for installation...
0:group munin already present
0:user munin already present
Munin-node-1.2.4-7

and stopped at this moment. Why this is happening? I tried to 
installed Trixbox 3 times. But, I faced this problem evertime. Please 
tell me the problem. Looking forward to your response. Thank you.


Regards,
Chandra.

  http://lists.digium.com/mailman/listinfo/asterisk-users
 
http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com


Chandra,

You might have more luck asking this in the trixbox forum.

I received the same problem. I think all I did was power off the box and 
reboot and it went all the way to the end of the install. I don't know 
why this happens, sorry I'm not of more help.


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[asterisk-users] Strange problem(Munin-node-1.2.4-7)

[Crazy Boy]

Hi,Sorry to post this in this forum.I am new to Trixbox. When I am trying to install Trixbox, I am facing this problem. First I have installed Trixbox ISO image file from a CD. When its rebooting and Asterisk is installing, it is got stucked near this below point:Munin-1.2.4-7Preparing package for installation...0:group munin already present0:user munin already presentMunin-node-1.2.4-7and stopped at this moment. Why this is happening? I tried to installed Trixbox 3 times. But, I faced this problem evertime. Please tell me the problem. Looking forward to your response. Thank you.Regards,Chandra. 
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[asterisk-users] strange problem with calls between MGCP and SIP clients(ATA's)

[Andy Kuo]

Hi,

We have experience problems with calls between MGCP ATA's and SIP
ATA's (Linksys PAP2-NA).
A call from MGCP or SIP to the other connects normally and the
conversation can usually last around 30 seconds and it becomes one-way
audio.

What I don't understand is how the calls can be set up and talk for a
few seconds without problems and suddenlly go wrong.  If there are
problems, such as misconfiguration, the call should not even be
connected, or at least the on-way audio problem should start right
from the beginning, shouldn't it?

I know MGCP is not very popular here, but we have quite a few of them
on hand that we would really like to use.
Any comments/suggestions are greatly appreciated.

Thanks.
Andy
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[Asterisk-Users] Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!

[Yoja Asterisk]

I've got a strange situation with musiconhold.

It works if I dial my extension 6000:

From extensions.conf:

exten = 6000,1,Answer
exten = 6000,2,MusicOnHold()


Debug output if I call 6000:
-- Executing Answer(SIP/gs1-b6ee, ) in new stack
-- Executing MusicOnHold(SIP/gs1-b6ee, ) in new stack
-- Started music on hold, class 'default', on SIP/gs1-b6ee
-- Stopped music on hold on SIP/gs1-b6ee
server*CLI


If I dial out and put a call on hold the other party hears the musiconhold:

Debug output when I do an outgoing call:
-- Executing SetCallerID(SIP/gs1-cb7a, Anonymous 0031x)
in new stack
-- Executing Dial(SIP/gs1-cb7a, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/voipbuster-ac66 is making progress passing it to SIP/gs1-cb7a
-- SIP/voipbuster-ac66 answered SIP/gs1-cb7a
-- Attempting native bridge of SIP/gs1-cb7a and SIP/voipbuster-ac66
-- Started music on hold, class 'default', on SIP/voipbuster-ac66
-- Stopped music on hold on SIP/voipbuster-ac66


But If somebody rings me and I put him on hold he hears nothing:

Debug output for incoming call:
-- Executing SetCallerID(SIP/gw02-mci.budgetphone.nl-42ba1908,
prive xx) in new stack
-- Executing Dial(SIP/gw02-mci.budgetphone.nl-42ba1908,
SIP/sipuraSIP/gs4) in new stack
-- Called sipura
-- Called gs4
-- SIP/sipura-7685 is ringing
-- SIP/gs4-4a86 is ringing
-- SIP/gs4-4a86 answered SIP/gw02-mci.budgetphone.nl-42ba1908
-- Attempting native bridge of SIP/gw02-mci.budgetphone.nl-42ba1908 and
SIP/gs4-4a86
-- Started music on hold, class 'default', on
SIP/gw02-mci.budgetphone.nl-42ba1908

According to the logs it starts the music on hold, the same way as in the
other calls but it stays quiet! I've tried everything but I don't know what
else to check.

I've got Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian machine.

In my sip.conf:
[general]
musicclass=default
musiconhold=default

(I tried it with only miscclass, only musiconhold, and without both, nothing
changes)

In musiconhold.conf
[classes]
default = quietmp3:/usr/share/asterisk/mohmp3

What can be wrong, what else can I check?

Kind regards,

De Boer

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[Asterisk-Users] strange problem with Telasip DID, please help

[Xin Li]

I have configured telasip DID with following entried in 
sip_custom.conf [telasip] username= (fake) type=peer 
secret=x quality=yes nat=yes insecure=very fromuser= 
host=gw4.telasip.com #disallow=all #allow=ulaw #allow=alaw 
fromdomain=gw4.telasip.com context=from-telasip and Register 
string in sip.conf under general and extensions.conf has following entries 
[from-telasip] exten = 1134817097,1,Answer exten = 
1134817097,2,Wait(1) exten = 1134817097,3,Background(pls-hold-while-try) 
exten = 1134817097,4,NoOp(Incoming call for Suzie on TelaSIP 
#8431234567) exten = 1134817097,5,Dial(SIP/71469,20,m) exten = 
1134817097,6,VoiceMail([EMAIL PROTECTED]) exten = 1134817097,7,Hangup 
The problem is I can receive one incoming call to this DID successfully. 
Then I tried to call this DID, it say it is not avaiable. SO in Asterisk CLI I 
type reload to reload Asterisk. Then incoming call works again, then next one is 
not, then reload, it works, so and so. What could be the problem? Please 
help.
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[Asterisk-Users] strange problem with asterisk in media proxy mode

[VoIP Linux]

HiI am facing very strange problem when i try to use asterisk in media proxy mode by using canreinvite=no i receive no voice at both ends. and when i use canreinvite=yes voice is OK at both endpoints. i tried to use play back application to check if asterisk is communicating well with UA and play back works fineanyone ever faced this problem pls help mehere is the declaration of my UAs in sip.conf[5000500]type=peerhost=dynamicdtmfmode=rfc2833context=defaultcanreinvite=yesallow=all[5000600]type=peerhost=dynamicdtmfmode=rfc2833context=defaultcanreinvite=yesallow=all  thanks allot  
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[Asterisk-Users] Strange Problem

[Gulzar Hussain]

Hi All

I am having a strange problem when I call from 1 RTC
Client to another without Asterisk in between
everything use to be fine but when asterisk is there
as a Registrar a problem use to occur in many calls,
Caller can hear the voice of the receiving side
but the receiver cant be able to hear the caller for
exact 8 seconds, conversation will become two
way after 8 seconds but this problem is a big
hurdle in proper establishment of a call

Does anybody ever had this problem ?
Any suggestions will be higly apreciated

i have tried capturing packets but dont find anything
abnormal

Thanx in Advance






__ 
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Re: [Asterisk-Users] Strange Problem

[Rehan Ahmed]

Try Nat=1 in your general and user settings.

Rehan

On 12/23/05, Gulzar Hussain [EMAIL PROTECTED] wrote:
Hi AllI am having a strange problem when I call from 1 RTCClient to another without Asterisk in between
everything use to be fine but when asterisk is thereas a Registrar a problem use to occur in many calls,Caller can hear the voice of the receiving sidebut the receiver cant be able to hear the caller for
exact 8 seconds, conversation will become twoway after 8 seconds but this problem is a bighurdle in proper establishment of a callDoes anybody ever had this problem ?Any suggestions will be higly apreciated
i have tried capturing packets but dont find anythingabnormalThanx in Advance__Yahoo! for Good - Make a difference this year.
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-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service.

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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Evil Skymarshal]

Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
If you do not have QOS assigned to the SIP protocol it is quite possiblethat there are packets time outs and the packets are discarded. Is itpossible to test the network during the evening or at a time whentraffic is at it lowest?
It took me some 2 days to debug that biest. Finaly I found it: The firewall rules allowed UDP from 10,000 to 12,000. So I had a fair chance of getting a connection from time to time. ;-)cu
 ES
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[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Evil Skymarshal]

Hi,I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For testing reasons I but the following in extensions.conf---cut---[from-sip]exten = 2000,1,Answer()exten = 2000,2,Wait(1)
exten = 2000,3,SayDigits(123)exten = 2000,4,Hangup()---cut---When ever I call the 2000 asterisk -vc says:---cut--- -- Executing Answer(SIP/2303-1ae1, ) in new stack
 -- Executing Wait(SIP/2303-1ae1, 1) in new stack -- Executing SayDigits(SIP/2303-1ae1, 123) in new stack -- Playing 'digits/1' (language 'en') -- Playing 'digits/2' (language 'en')
 -- Playing 'digits/3' (language 'en') -- Executing Hangup(SIP/2303-1ae1, ) in new stack---cut---BUT I don't hear it everytime! Why? Sometimes I can hear it and most time I can not. I redial 20 times and I can hear it only 3-4 times. Does anybody have an idea what kind of strange problem that could be?
Thanx ES
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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

What are you codec and dmtfmode settings in sip.conf and in the sip 
phone settings. If you dmtfmode is set to 'inband' and you are using 
anything other than ulaw or alaw codec it wont work. Also since your 
hear the phone sometimes you may be experiencing QOS issues on your 
network. Doe you have QOS set up on your switches in the points between 
the server running asterisk and the sip client?


Hope this helps

Evil Skymarshal wrote:


Hi,

I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For 
testing reasons I but the following in extensions.conf


---cut---
[from-sip]
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,SayDigits(123)
exten = 2000,4,Hangup()
---cut---

When ever I call the 2000 asterisk -vc says:

---cut---
-- Executing Answer(SIP/2303-1ae1, ) in new stack
-- Executing Wait(SIP/2303-1ae1, 1) in new stack
-- Executing SayDigits(SIP/2303-1ae1, 123) in new stack
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Executing Hangup(SIP/2303-1ae1, ) in new stack
---cut---

BUT I don't hear it everytime! Why? Sometimes I can hear it and most 
time I can not. I redial 20 times and I can hear it only 3-4 times. 
Does anybody have an idea what kind of strange problem that could be?


Thanx
  ES



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
 



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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Evil Skymarshal]

Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
What are you codec and dmtfmode settings in sip.conf and in the sipphone settings.I use gsm.
 If you dmtfmode is set to 'inband' and you are usinganything other than ulaw or alaw codec it wont work.I changed the settings and tried:---cut---exten = 2000,1,Answer()exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)exten = 2000,4,Hangup()---cut---Same problem. Sometimes it works but most of the times it doesn't.
 Also since yourhear the phone sometimes you may be experiencing QOS issues on yournetwork.Of course it could be a QOS problem. But should I hear at least something?cu ES

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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

If you do not have QOS assigned to the SIP protocol it is quite possible 
that there are packets time outs and the packets are discarded. Is it 
possible to test the network during the evening or at a time when 
traffic is at it lowest? Also try several traceroutes and see if there 
is a wide variation in return times (widely varying treceroutes could 
indicate network saturation). You are using gsm are you using 
dmtfmode=rfc2833 or something else (this must be set in the sip.conf and 
on the sip soft phone and they must match!)


Thanks

Evil Skymarshal wrote:


Hi Chuck,

2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings.


I use gsm.

If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work.


I changed the settings and tried:
---cut---
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)
exten = 2000,4,Hangup()
---cut---

Same problem. Sometimes it works but most of the times it doesn't.
 


Also since your
hear the phone sometimes you may be experiencing QOS issues on your
network.


Of course it could be a QOS problem. But should I hear at least something?

cu
  ES



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
 



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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

Something else I should mention. Sip uses UDP and TCP packets. TCP 
packets are used if there is congestion on the network. I am unclear 
about what mechanism causes sip to switch between UDP and TCP but I 
believe it is controllable - I believe It would be easier to use QOS 
though. If UDP is used that packets could time out and you would never 
know it since UDP is dumb and has no packet loss recovery mechanism. 
What is the topology of your network. Is the Asterisk box and the client 
on the same backbone and switch?


Thanks

Evil Skymarshal wrote:


Hi Chuck,

2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings.


I use gsm.

If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work.


I changed the settings and tried:
---cut---
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)
exten = 2000,4,Hangup()
---cut---

Same problem. Sometimes it works but most of the times it doesn't.
 


Also since your
hear the phone sometimes you may be experiencing QOS issues on your
network.


Of course it could be a QOS problem. But should I hear at least something?

cu
  ES



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
 



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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Rich Adamson]

I don't believe asterisk has any sip tcp support. Its all udp.


 Hi,
 
 Something else I should mention. Sip uses UDP and TCP packets. TCP 
 packets are used if there is congestion on the network. I am unclear 
 about what mechanism causes sip to switch between UDP and TCP but I 
 believe it is controllable - I believe It would be easier to use QOS 
 though. If UDP is used that packets could time out and you would never 
 know it since UDP is dumb and has no packet loss recovery mechanism. 
 What is the topology of your network. Is the Asterisk box and the client 
 on the same backbone and switch?
 
 Thanks
 
 Evil Skymarshal wrote:
 
  Hi Chuck,
 
  2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED]:
 
  What are you codec and dmtfmode settings in sip.conf and in the sip
  phone settings.
 
 
  I use gsm.
 
  If you dmtfmode is set to 'inband' and you are using
  anything other than ulaw or alaw codec it wont work.
 
 
  I changed the settings and tried:
  ---cut---
  exten = 2000,1,Answer()
  exten = 2000,2,Wait(1)
  exten = 2000,3,Playback(hello-world)
  exten = 2000,4,Hangup()
  ---cut---
 
  Same problem. Sometimes it works but most of the times it doesn't.
   
 
  Also since your
  hear the phone sometimes you may be experiencing QOS issues on your
  network.
 
 
  Of course it could be a QOS problem. But should I hear at least something?
 
  cu
ES
 
 
 
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 No virus found in this incoming message.
 Checked by AVG Free Edition.
 Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
   
 
 
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---End of Original Message-


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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

Rich I stand corrected you are absolutely right - see 
http://www.voip-info.org/wiki-Asterisk+config+sip.conf


The following appears on the page:


   Please note

   * Asterisk does not yet support SIP over TCP. It only supports SIP
 http://www.voip-info.org/wiki/view/SIP over UDP.
   * For Grandstream http://www.voip-info.org/wiki/view/Grandstream
 phones: set *dtmfmode=info*
   * Asterisk uses the incoming RTP
 http://www.voip-info.org/wiki/view/RTP Stream as a timing source
 for sending its outgoing Stream. If the incoming stream is
 interrupted due to silence suppression then musiconhold will be
 choppy. So in conclusion, you cannot use silence suppression.
 *Make sure ALL SIP phones have disabled silence suppression.*
 There is a solution for the silence suppression problem, see bug
 5374 http://bugs.digium.com/view.php?id=5374 for details.

Thanks


Rich Adamson wrote:


I don't believe asterisk has any sip tcp support. Its all udp.


 


Hi,

Something else I should mention. Sip uses UDP and TCP packets. TCP 
packets are used if there is congestion on the network. I am unclear 
about what mechanism causes sip to switch between UDP and TCP but I 
believe it is controllable - I believe It would be easier to use QOS 
though. If UDP is used that packets could time out and you would never 
know it since UDP is dumb and has no packet loss recovery mechanism. 
What is the topology of your network. Is the Asterisk box and the client 
on the same backbone and switch?


Thanks

Evil Skymarshal wrote:

   


Hi Chuck,

2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


   What are you codec and dmtfmode settings in sip.conf and in the sip
   phone settings.


I use gsm.

   If you dmtfmode is set to 'inband' and you are using
   anything other than ulaw or alaw codec it wont work.


I changed the settings and tried:
---cut---
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)
exten = 2000,4,Hangup()
---cut---

Same problem. Sometimes it works but most of the times it doesn't.


   Also since your
   hear the phone sometimes you may be experiencing QOS issues on your
   network.


Of course it could be a QOS problem. But should I hear at least something?

cu
 ES



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005


 


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---End of Original Message-






 



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[Asterisk-Users] Strange problem with dtmf pound and star

[Michael Kleinhenz]

Hi,

I've got a strange problem here: I am using Asterisk 1.0.7 (the Debian
Testing version asterisk-1.0.7.dfsg.1-2). The dtmfmode is set to rfc2833
and I'm making a call to Asterisk via SIP. If I press star in this call,
it is recognized as a pound and the star is not available. The physical
pound key on the keypad has no function.

Anyone heard of this problem? Could it be that european phones are
different? I've tried it with two different phones on the same SIP
router (DeTeWe 31lan SIP).

Thanks,
Michael

--
Michael Kleinhenz
[EMAIL PROTECTED]   
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[Asterisk-Users] Strange problem with Bristuff

[tonini . massimo]

Hi all,
I have a strange problem with a quadbri
card and my asterisk box with installed verson 1.0.7 of asterisk Bristuffed.
I have connected to the card 3 isdn
in ptp mode configured in selection passing (I don't know if is exact the
english traduction but I have 3 isdn with 99 numbers and asterisk forward
the extensions)
The problem is this: if I call from
a cellular to asterisk all is Ok but when I try to call from a fixed line
the extension (the last part of the number) is not sent but only the first
part.

Someone can help me ?

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[Asterisk-Users] strange problem

[Christoph Eicke]

Hi!

I have a strange problem with Asterisk (Asterisk 1.0.8-BRIstuffed-0.2.0-RC7k) 
on a VIA Samuel x86. When I make a call from the CLI either over IAX2 or SIP, 
my first call after the initial start of Asterisk works fine, even though 
upons starting Asterisk tells me Read error on sound device: Resource 
temporarily unavailable. I hear the call over the loudspeaker. When I hang 
up (using CLI hangup command) and try to place another call, I get a busy 
signal over the loudspeaker and I get Error reading from sound device (If 
you're running 'artsd' then kill it): Resource temporarily unavailable. If I 
issue a hangup again, I can dial out fine again. 
I don't have artsd running and nothing else besides Asterisk is using /dev/dsp 
(according to lsof). I have read somewhere that this might have to do with 
the sound card chip that I'm running (VIA Technologies, Inc. VT82C686 AC97 
Audio Controller (rev 50)). 
Unfortunately I don't have the luxury of getting to see the debug output 
of /var/log/asterisk/debug on that machine, because it runs on a 128MB read 
only file system. 
I'm not loading chan_alsa.so, only chan_oss.so as I think this might have 
something to do with the problem.

Any help would be great, or any hints into a possible direction.

Thanks,
Christoph
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RE: [Asterisk-Users] Strange problem with SIP and CAPI

[Cyrille Demaret]

Hi,

New version of chan_capi-cm 0.5.4 has fixed my problem.

Regards,

Cyrille

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Cyrille
Demaret
Envoyé : vendredi 15 juillet 2005 11:13
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Strange problem with SIP and CAPI

Hi,

I’ve strange problem when I’m making a call from SIP (Cisco 7960) to capi
(Fritz PCI). When I call a national number, I’m hearing the ringtone when
the called party is ringing but when I call an international number, I don’t
hear the ringtone and I’ve a silence until the called party answers. Both
call are going through the same extension.

Here’s 2 log files, one with a national number and one with an
international.

National :

--- 
-- Executing Dial(SIP/200-837b, CAPI/010xx:b0478xx|30) in
new stack
-- data = 010xx:b0478xx
-- capi request omsn = 010xx
  == found capi with omsn = 010xx
  == CAPI Call CAPI[contr1/010xx]/80 with B3-- creating pipe for
PLCI=-1
-- CONNECT_CONF ID=003 #0x0844 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- CONNECT_CONF ID=003 #0x0844 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

  == received CONNECT_CONF PLCI = 0x101 INFO = 0
sent CONNECT_REQ MN =0x844
-- Called 010xx:b0478xx
-- INFO_IND ID=003 #0x811c LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8002
  InfoElement = default

-- INFO_IND ID=003 #0x811c LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8002
  InfoElement = default

sent INFO_RESP (PLCI=0x101)
-- INFO_IND ID=003 #0x811d LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- INFO_IND ID=003 #0x811d LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

sent INFO_RESP (PLCI=0x101)
-- INFO_IND ID=003 #0x811e LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8001
  InfoElement = default

-- INFO_IND ID=003 #0x811e LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8001
  InfoElement = default

sent INFO_RESP (PLCI=0x101)
-- CAPI[contr1/010xx]/80 is ringing
 I hear the ringtone here
-- INFO_IND ID=003 #0x811f LEN=0020
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x29
  InfoElement = 05 07 0f 0a6

-- INFO_IND ID=003 #0x811f LEN=0020
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x29
  InfoElement = 05 07 0f 0a6

sent INFO_RESP (PLCI=0x101)
-- CONNECT_ACTIVE_IND ID=003 #0x8120 LEN=0028
  Controller/PLCI/NCCI= 0x101
  ConnectedNumber = 11 8332478540441
  ConnectedSubaddress = default
  LLC = default

-- CONNECT_ACTIVE_IND ID=003 #0x8120 LEN=0028
  Controller/PLCI/NCCI= 0x101
  ConnectedNumber = 11 8332478540441
  ConnectedSubaddress = default
  LLC = default

sent CONNECT_ACTIVE_RESP (PLCI=0x101)
-- sent CONNECT_B3_REQ (PLCI=0x101)
-- CONNECT_B3_CONF ID=003 #0x0845 LEN=0014
  Controller/PLCI/NCCI= 0x10101
  Info= 0x0

-- CONNECT_B3_CONF ID=003 #0x0845 LEN=0014
  Controller/PLCI/NCCI= 0x10101
  Info= 0x0

-- CONNECT_B3_ACTIVE_IND ID=003 #0x8121 LEN=0013
  Controller/PLCI/NCCI= 0x10101
  NCPI= default

-- CONNECT_B3_ACTIVE_IND ID=003 #0x8121 LEN=0013
  Controller/PLCI/NCCI= 0x10101
  NCPI= default

sent CONNECT_B3_ACTIVE_RESP (NCCI=0x10101)
-- CAPI[contr1/010xx]/80 answered SIP/200-837b
-- DATA_B3_IND ID=003 #0x8122 LEN=0030
  Controller/PLCI/NCCI= 0x10101
  Data32  = 0xb79f39fe
  DataLength  = 0xa0
  DataHandle  = 0x0
  Flags   = 0x0
  Data64  = 0x0

.
sent DATA_B3_RESP (NCCI=0x10101)
-- DATA_B3_IND ID=003 #0x8143 LEN=0030
  Controller/PLCI/NCCI= 0x10101
  Data32  = 0xb79f39fe
  DataLength  = 0xa0
  DataHandle  = 0x1
  Flags   = 0x0
  Data64  = 0x0

sent

[Asterisk-Users] Strange problem with SIP and CAPI

[Cyrille Demaret]

Hi,

I’ve strange problem when I’m making a call from SIP (Cisco 7960) to capi
(Fritz PCI). When I call a national number, I’m hearing the ringtone when
the called party is ringing but when I call an international number, I don’t
hear the ringtone and I’ve a silence until the called party answers. Both
call are going through the same extension.

Here’s 2 log files, one with a national number and one with an
international.

National :

--- 
-- Executing Dial(SIP/200-837b, CAPI/010xx:b0478xx|30) in
new stack
-- data = 010xx:b0478xx
-- capi request omsn = 010xx
  == found capi with omsn = 010xx
  == CAPI Call CAPI[contr1/010xx]/80 with B3-- creating pipe for
PLCI=-1
-- CONNECT_CONF ID=003 #0x0844 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- CONNECT_CONF ID=003 #0x0844 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

  == received CONNECT_CONF PLCI = 0x101 INFO = 0
sent CONNECT_REQ MN =0x844
-- Called 010xx:b0478xx
-- INFO_IND ID=003 #0x811c LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8002
  InfoElement = default

-- INFO_IND ID=003 #0x811c LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8002
  InfoElement = default

sent INFO_RESP (PLCI=0x101)
-- INFO_IND ID=003 #0x811d LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- INFO_IND ID=003 #0x811d LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

sent INFO_RESP (PLCI=0x101)
-- INFO_IND ID=003 #0x811e LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8001
  InfoElement = default

-- INFO_IND ID=003 #0x811e LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8001
  InfoElement = default

sent INFO_RESP (PLCI=0x101)
-- CAPI[contr1/010xx]/80 is ringing
 I hear the ringtone here
-- INFO_IND ID=003 #0x811f LEN=0020
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x29
  InfoElement = 05 07 0f 0a6

-- INFO_IND ID=003 #0x811f LEN=0020
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x29
  InfoElement = 05 07 0f 0a6

sent INFO_RESP (PLCI=0x101)
-- CONNECT_ACTIVE_IND ID=003 #0x8120 LEN=0028
  Controller/PLCI/NCCI= 0x101
  ConnectedNumber = 11 8332478540441
  ConnectedSubaddress = default
  LLC = default

-- CONNECT_ACTIVE_IND ID=003 #0x8120 LEN=0028
  Controller/PLCI/NCCI= 0x101
  ConnectedNumber = 11 8332478540441
  ConnectedSubaddress = default
  LLC = default

sent CONNECT_ACTIVE_RESP (PLCI=0x101)
-- sent CONNECT_B3_REQ (PLCI=0x101)
-- CONNECT_B3_CONF ID=003 #0x0845 LEN=0014
  Controller/PLCI/NCCI= 0x10101
  Info= 0x0

-- CONNECT_B3_CONF ID=003 #0x0845 LEN=0014
  Controller/PLCI/NCCI= 0x10101
  Info= 0x0

-- CONNECT_B3_ACTIVE_IND ID=003 #0x8121 LEN=0013
  Controller/PLCI/NCCI= 0x10101
  NCPI= default

-- CONNECT_B3_ACTIVE_IND ID=003 #0x8121 LEN=0013
  Controller/PLCI/NCCI= 0x10101
  NCPI= default

sent CONNECT_B3_ACTIVE_RESP (NCCI=0x10101)
-- CAPI[contr1/010xx]/80 answered SIP/200-837b
-- DATA_B3_IND ID=003 #0x8122 LEN=0030
  Controller/PLCI/NCCI= 0x10101
  Data32  = 0xb79f39fe
  DataLength  = 0xa0
  DataHandle  = 0x0
  Flags   = 0x0
  Data64  = 0x0

.
sent DATA_B3_RESP (NCCI=0x10101)
-- DATA_B3_IND ID=003 #0x8143 LEN=0030
  Controller/PLCI/NCCI= 0x10101
  Data32  = 0xb79f39fe
  DataLength  = 0xa0
  DataHandle  = 0x1
  Flags   = 0x0
  Data64  = 0x0

sent DATA_B3_RESP (NCCI=0x10101)
-- DATA_B3_CONF ID=003 #0x0861 LEN=0016
  Controller/PLCI/NCCI= 0x10101
  DataHandle  = 0x6fe
  Info= 0x0

-- DATA_B3_CONF ID=003 #0x0861 LEN=0016
  Controller/PLCI/NCCI= 0x10101
  DataHandle  = 0x6fe
  

Re: [Asterisk-Users] Strange problem with SIP and CAPI

[Armin Schindler]

On Fri, 15 Jul 2005, Cyrille Demaret wrote:
 Hi,
 
 I’ve strange problem when I’m making a call from SIP (Cisco 7960) to capi
 (Fritz PCI). When I call a national number, I’m hearing the ringtone when
 the called party is ringing but when I call an international number, I don’t
 hear the ringtone and I’ve a silence until the called party answers. Both
 call are going through the same extension.
 
 Here’s 2 log files, one with a national number and one with an
 international.
... 
 Does anyone have an idea about what's wrong?

I cannot see any difference in the logs. Even the first one does not seem
to have early-B3.
But the logs look incomplete, you are using chan_capi-0.3.5, right?
Maybe you want to try out chan_capi-cm from sourceforge...

Armin
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Re: [Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream

[Jason Williams]

 But when BT-100 calls 7960 the following is happening:
 
-- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack
-- Called 1707
-- SIP/1707-e96a is ringing
-- SIP/1707-e96a answered SIP/3710-8f2b
-- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a
 
 May  4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is
 not codec1 = 4, cannot native bridge.
 
 sipsrv1*CLI sip show channels
 
 Peer User/ANRCall ID  Seq (Tx/Rx)   Format  Last Msg
 192.168.128.171  170702fff7f7169  00102/0   ulawTx: ACK
 67.126.23.2513710b5d3f977ea1  00101/52181   g729Rx: ACK
 
 When this bug is gonna be fixed?
 

Change the codec order in the phone configuration and place g729
higher it is not asterisk doing this
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[Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream

[Irakli Natsvlishvili]

Hello everybody,

Further interesting details about BT-100, * and Cisco 7960.

Asterisk has G729 installed, on BT-100 there is g729 selected on all codec
selections. On Cisco 7960 preferred codec is g711.

Form sip.conf


[1707]
;- Cisco 7960
context=default
type= friend
username=1707
host = dynamic
dtmfmode=rfc2833
qualify=2000
disallow=all
allow=g729
allow=ulaw

[3710]
; - GrandStream Bt-100
context=default
type=friend
username=3710
user=phone
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
qualify=2000
disallow=all
allow=g729
allow=ulaw

When 7960 calls BT-100 there is g729 used on both ends. 

sipsrv1*CLI sip show channels

Peer User/ANRCall ID  Seq (Tx/Rx)   Format  Last Msg
67.126.23.2513710118e46ce79a  00103/0   g729Tx: ACK
192.168.128.171  170700070ef7-36  00102/00101   g729Tx: ACK

But when BT-100 calls 7960 the following is happening:

-- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack
-- Called 1707
-- SIP/1707-e96a is ringing
-- SIP/1707-e96a answered SIP/3710-8f2b
-- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a

May  4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is
not codec1 = 4, cannot native bridge.

sipsrv1*CLI sip show channels

Peer User/ANRCall ID  Seq (Tx/Rx)   Format  Last Msg
192.168.128.171  170702fff7f7169  00102/0   ulawTx: ACK
67.126.23.2513710b5d3f977ea1  00101/52181   g729Rx: ACK

When this bug is gonna be fixed?

I.N. 

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[Asterisk-Users] Strange problem with h323

[David J Carter]

All,

I have downloaded and installed openh323 as per the documentation.

When the machine now reboots safe_asterisk just keeps restarting.

If I start another session and just load asterisk -vvvgc asterisk loads.

If I enter noload chan_h323.so in the modules.conf then safe_asterisk will
kick in.

Not 100% on Linux yet but I have added the environment variables info into
/etc/profile so they would load each time a reboot takes place, (thought
this is the right place).

If I do export, the list doesn't show the environment variables, so I assume
I have added them in the wrong place. This I assume is why h323 is failing.

Anyone point me in the right direction as to where to load these variables,
so they load every time?

Thanks

Dave

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[Asterisk-Users] Strange problem with incoming call.

[C F]

When someone calls in on a zap channel with FXO and presses an
extension, and another user picks up using (*8) I changed it to 888,
after a few minutes ( I think 2), the call gets dissconected. The
users all use Cisco 7960.
I didn't yet have a chance to test it when not using Call Pickup (*8)888.
Please help.
Here is the screen shot in asterisk:
+++
===
-- Executing Macro(Zap/1-1, rollbusy|102) in new stack ;macro
to ring next available line on cisco phone
-- Executing Dial(Zap/1-1, SIP/1021|15|tm) in new stack
-- Called 1021
-- Started music on hold, class 'default', on Zap/1-1
-- SIP/1021-eaad is ringing
-- SIP/1011-4c98 answered Zap/1-1 ;another phone picked up pressing 888
-- Stopped music on hold on Zap/1-1
pbx*CLI sip debug ; i enabled debug here
SIP Debugging Enabled
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for
address/port to send to
set_destination: set destination to 192.168.123.60, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.123.50:5060;branch=z9hG4bK01879857
From: sip:[EMAIL PROTECTED];tag=as4fb177da
To: JJ Fried sip:[EMAIL PROTECTED];tag=003094c29e4902aa56de3e78-1e1782d1
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.123.60:5060
  == Spawn extension (macro-rollbusy, s, 1) exited non-zero on
'Zap/1-1' in macro 'rollbusy'
  == Spawn extension (macro-stdext, s, 2) exited non-zero on 'Zap/1-1'
in macro 'stdext'
  == Spawn extension (macro-ccs, s, 5) exited non-zero on 'Zap/1-1' in
macro 'ccs'
  == Spawn extension (macro-handleexten, s, 4) exited non-zero on
'Zap/1-1' in macro 'handleexten'
  == Spawn extension (closed, 102, 1) exited non-zero on 'Zap/1-1'
-- Executing Playback(Zap/1-1, goodbye) in new stack
-- Playing 'goodbye' (language 'en')
pbx*CLI

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.123.50:5060;branch=z9hG4bK01879857
From: sip:[EMAIL PROTECTED];tag=as4fb177da
To: JJ Fried sip:[EMAIL PROTECTED];tag=003094c29e4902aa56de3e78-1e1782d1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 06 Jan 2005 23:26:08 GMT
CSeq: 102 BYE
Server: CSCO/6
Content-Length: 0


9 headers, 0 lines
Message is BYE
Destroying call '[EMAIL PROTECTED]'
-- Executing Hangup(Zap/1-1, ) in new stack
  == Spawn extension (closed, T, 2) exited non-zero on 'Zap/1-1'
-- Executing System(Zap/1-1, /bin/rm .gsm) in new stack
-- Hungup 'Zap/1-1'
pbx*CLI

+
Please help.
Thanks
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[Asterisk-Users] Strange problem with TDM400 FXO in UK

[Ian D. Wlloughby]

Hi Folks,
I have a Rev H TDM01B board which seems to be working pretty well. However sometimes when I dial out on the Zap channel I get the Congestion signal on my SIP phone and a message logged in the log file saying :-

Unable to create channel of type 'Zap'

followed by :-

Dial argument takes format (technology1/[device:]numbetechnology2/[device:]number2...|optional timeout)

If I then dial in my SIP phone rings and I am again able to dial out.

I am guessing the Polarity reversal on the dial in is reseting the channel but why is it getting in this state in the first place, any idea's?

Regards
Ian

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Re: [Asterisk-Users] strange problem PBX-Asterisk

[Kurt Bauer]

Thanks for the hints, 'overlapdial=yes' did the trick.
br,
kurt
--On Tuesday, August 24, 2004 10:08:08 PM +0200 Peter Svensson 
[EMAIL PROTECTED] wrote:

On Tue, 24 Aug 2004, Christian Victor wrote:
 maybe I oversee somth. very obvious, but I'm a little puzzled about
 the  following 'error':

 When I make a call from the PBX to * I get number not available, but
 on  debug I see, that asterisk is searching just for the first digit
 in the  extension, which of course doesn't exist, eg:
I seems that you PBX uses Overlap Dial and transmits the extensien
digit by digit and Asterisk expects the extension to be en bloc. So
when it receives anything from the PBX (wich is in this case the first
digit) Asterisk thinks that this is the whole block of extension.
Don't know how to fix it though. ;-)
This in case you are on a PRI: see overlapdial at
  http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
It needs to be set on pri links where ovarlap dialing is used, even
incoming towards asterisk.
Without more information on the connections between the systems and the
configuration it is hard to figure out what is wrong.
Peter
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[Asterisk-Users] strange problem PBX-Asterisk

[Kurt Bauer]

Hi,
maybe I oversee somth. very obvious, but I'm a little puzzled about the 
following 'error':

When I make a call from the PBX to * I get number not available, but on 
debug I see, that asterisk is searching just for the first digit in the 
extension, which of course doesn't exist, eg:

I dial 77 (for conn to *) and 12345 (valid extension) on the console I see:
-- Extension '1' in context 'sip-local' from '+ 14070' does not 
exist.  Rejecting call on channel 0/1, span 1

Everything worked fine before an update on Friday, but I haven't changed 
any config files then.
I 'downgraded' to 1.0RC2 today, but then same problem.

If any of you has any hints, please let me know.
Thanks a lot,
br,
Kurt

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Re: [Asterisk-Users] strange problem PBX-Asterisk

[Christian Victor]

Hi Kurt!
maybe I oversee somth. very obvious, but I'm a little puzzled about the 
following 'error':

When I make a call from the PBX to * I get number not available, but on 
debug I see, that asterisk is searching just for the first digit in the 
extension, which of course doesn't exist, eg:
I seems that you PBX uses Overlap Dial and transmits the extensien 
digit by digit and Asterisk expects the extension to be en bloc. So 
when it receives anything from the PBX (wich is in this case the first 
digit) Asterisk thinks that this is the whole block of extension.

Don't know how to fix it though. ;-)
Christian
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Re: [Asterisk-Users] strange problem PBX-Asterisk

[Peter Svensson]

On Tue, 24 Aug 2004, Christian Victor wrote:

  maybe I oversee somth. very obvious, but I'm a little puzzled about the 
  following 'error':
  
  When I make a call from the PBX to * I get number not available, but on 
  debug I see, that asterisk is searching just for the first digit in the 
  extension, which of course doesn't exist, eg:
 
 I seems that you PBX uses Overlap Dial and transmits the extensien 
 digit by digit and Asterisk expects the extension to be en bloc. So 
 when it receives anything from the PBX (wich is in this case the first 
 digit) Asterisk thinks that this is the whole block of extension.
 
 Don't know how to fix it though. ;-)

This in case you are on a PRI: see overlapdial at
  http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
It needs to be set on pri links where ovarlap dialing is used, even 
incoming towards asterisk.

Without more information on the connections between the systems and the 
configuration it is hard to figure out what is wrong.

Peter


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Re: [Asterisk-Users] Strange problem with Dial

[Michael George]

On Fri, Aug 20, 2004 at 07:14:06PM -0400, Michael George wrote:
 I'm trying to add an emergency dial to my context.  However, when I try to
 dial it, I get caught in an endless loop.
 
 For debugging, I have pared out nearly all the control flow and just have
 ChanIsAvail() and Dial() called.  Using two different extensions to call teh
 same number, I get two different actions by *.
 
 Here is the vvverbose output:
 
 -- Starting simple switch on 'Zap/3-1'
 -- Executing Dial(Zap/3-1, Zap/1/5932336||T) in new stack
 -- Called 1/5932336
 -- Zap/1-1 answered Zap/3-1
 -- Attempting native bridge of Zap/3-1 and Zap/1-1
 -- Hungup 'Zap/1-1'
   == Spawn extension (internal, 95932336, 1) exited non-zero on 'Zap/3-1'
 -- Hungup 'Zap/3-1'
 -- Starting simple switch on 'Zap/3-1'
 -- Executing ChanIsAvail(Zap/3-1, Zap/1) in new stack
 -- Hungup 'Zap/1-1'
 -- Executing NoOp(Zap/3-1, avail: Zap/1-1) in new stack
 -- Executing Dial(Zap/3-1, Zap/1/5932336||T) in new stack
 Aug 20 19:09:27 NOTICE[294926]: app_dial.c:714 dial_exec: Unable to create
 channel of type 'Zap'
   == Everyone is busy/congested at this time
 -- Executing NoOp(Zap/3-1, busy) in new stack
 -- Hungup 'Zap/3-1'
 
 the first way, I'm matching this context:
 exten = _9NXX,1,Dial(${TrunkLocal}/${EXTEN:${TrunkMSD}},,T)
 exten = _9NXX,2,Congestion
 exten = _9517XXX,1,Dial(${TrunkLocal}/${EXTEN},,T})
 exten = _9517XXX,2,Congestion
 
 The second way I'm mathing this one:
 exten = 911,1,ChanIsAvail(Zap/1)
 exten = 911,2,NoOp(avail: ${AVAILCHAN})
 exten = 911,3,Dial(Zap/1/5932336,,T)
 exten = 911,102,NoOp(None Avail)
 exten = 911,104,NoOp(busy)
 
 Why does the latter fail at the Dial()?

I am still having a problem with this call flow.  I just updated my * source
and rebuilt and reinstalled.

I want to implement the feature I saw in Tips  Tricks where before calling an
emergency number, the outgoing channel(s) are checked for availability so one
can be cleared before trying to dial.  The example code is this:

exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,Dial(Zap/1/911)
exten = 911,3,Hangup()
exten = 911,102,SoftHangup(Zap/1-1)
exten = 911,103,Wait(1)
exten = 911,104,Goto(1)

However, every time I try this flow, the Dial() called immediately after the
ChanIsAvail() will fail as busy (return to prio+101).  I know the channel is
available because I see that ChanIsAvail() went to prio+1.

Has there been a change in the code that might cause this?  Perhaps an issue
in the zaptel driver?

Thanks!

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Strange problem with Dial

[Michael George]

I'm trying to add an emergency dial to my context.  However, when I try to
dial it, I get caught in an endless loop.

For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called.  Using two different extensions to call teh
same number, I get two different actions by *.

Here is the vvverbose output:

-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, Zap/1/5932336||T) in new stack
-- Called 1/5932336
-- Zap/1-1 answered Zap/3-1
-- Attempting native bridge of Zap/3-1 and Zap/1-1
-- Hungup 'Zap/1-1'
  == Spawn extension (internal, 95932336, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
-- Starting simple switch on 'Zap/3-1'
-- Executing ChanIsAvail(Zap/3-1, Zap/1) in new stack
-- Hungup 'Zap/1-1'
-- Executing NoOp(Zap/3-1, avail: Zap/1-1) in new stack
-- Executing Dial(Zap/3-1, Zap/1/5932336||T) in new stack
Aug 20 19:09:27 NOTICE[294926]: app_dial.c:714 dial_exec: Unable to create
channel of type 'Zap'
  == Everyone is busy/congested at this time
-- Executing NoOp(Zap/3-1, busy) in new stack
-- Hungup 'Zap/3-1'

the first way, I'm matching this context:
exten = _9NXX,1,Dial(${TrunkLocal}/${EXTEN:${TrunkMSD}},,T)
exten = _9NXX,2,Congestion
exten = _9517XXX,1,Dial(${TrunkLocal}/${EXTEN},,T})
exten = _9517XXX,2,Congestion

The second way I'm mathing this one:
exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,NoOp(avail: ${AVAILCHAN})
exten = 911,3,Dial(Zap/1/5932336,,T)
exten = 911,102,NoOp(None Avail)
exten = 911,104,NoOp(busy)

Why does the latter fail at the Dial()?

- 


There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Strange problem with Dial

[Michael George]

On Fri, Aug 20, 2004 at 07:14:06PM -0400, Michael George wrote:
 I'm trying to add an emergency dial to my context.  However, when I try to
 dial it, I get caught in an endless loop.
 
 For debugging, I have pared out nearly all the control flow and just have
 ChanIsAvail() and Dial() called.  Using two different extensions to call teh
 same number, I get two different actions by *.

For some reason ChanIsAvail() is causing the channel to appear busy after the
return.  When I take the call to ChanIsAvail() out, the Dial() works just
fine.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] strange problem with oh323 loaded!

[Michael Manousos]

What exactly is the problem with v0.6.3(a)?
Michael.
Anthony Law wrote:
I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323
to 0.6.2a and it seems fine.

Regards,

Anthony
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***
* *
*G R E E C E  *
* *
* EUROPEAN CHAMPION EURO 2004 *
* *
***
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[Asterisk-Users] strange problem with oh323 loaded!

[Anthony Law]

Hi,

As explained in my original post on June 30. When I used CVS 2004-06-16 with
oh323-0.6.3a.  I can compile and install without problem but when I am in
the asterisk console whenever I issue stop now or restart now or
extension reload I got stuck on the console and asterisk did not response
to either shutting down or restarting.

It stucked on

Executing last minute cleanups
  == Cleaning up OpenH323 channel driver.
  == Unregistered channel type 'OH323'
  == Destroying any remaining musiconhold processes

 The same thing will not happen if I do not load the oh323-0.6.3.a module.
Since I have this problem I have gone back to oh323-0.6.3 and it acts the
same, finally yesterday I revert it back to oh323-0.6.2a and the above did
not happen. Do you happen to know why?



Regards,



Anthony


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RE: [Asterisk-Users] strange problem with oh323 loaded!

[T. Chan]

Hi,

I am the unlucky one, I have similar problem, but I am mostly using
safe_asterisk, and this stop now...restart now never works, with neither
0.6.3 nor 0.6.2

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anthony Law
Sent: Thursday, July 08, 2004 3:33 PM
To: Mailing List Asterisk
Subject: [Asterisk-Users] strange problem with oh323 loaded!


Hi,

As explained in my original post on June 30. When I used CVS 2004-06-16 with
oh323-0.6.3a.  I can compile and install without problem but when I am in
the asterisk console whenever I issue stop now or restart now or
extension reload I got stuck on the console and asterisk did not response
to either shutting down or restarting.

It stucked on

Executing last minute cleanups
  == Cleaning up OpenH323 channel driver.
  == Unregistered channel type 'OH323'
  == Destroying any remaining musiconhold processes

 The same thing will not happen if I do not load the oh323-0.6.3.a module.
Since I have this problem I have gone back to oh323-0.6.3 and it acts the
same, finally yesterday I revert it back to oh323-0.6.2a and the above did
not happen. Do you happen to know why?



Regards,



Anthony


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[Asterisk-Users] strange problem with oh323 loaded!

[Anthony Law]

I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323
to 0.6.2a and it seems fine.



Regards,



Anthony


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Re: [Asterisk-Users] strange problem with oh323 loaded!

[Michael Manousos]

OK, I'll look at it.
Michael.
T. Chan wrote:
Dear All,
I don't know but I tried all 0.6.x version of OH323 and normally I use
safe_asterisk to start asterisk, and everytime when I use 'stop now' to
terminate asterisk, it does not do anything, and you are rite, I have to use
kill -9 to kill the PIDs and threads. However, if I use asterisk -vvvgc to
start, 'stop now' works.
Thanks all
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: Friday, July 02, 2004 4:37 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] strange problem with oh323 loaded!
Same problem here - with latest 0.6.3a oh323.  Locks up on exit.  Had to
kill -9
This didn't happen with 0.6.2a, but that's on a different machine.  Maybe
you could try this older version which worked fine (same PwLib and OpenH323)
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law
Sent: Friday, July 02, 2004 1:15 PM
To: Mailing List Asterisk
Subject: [Asterisk-Users] strange problem with oh323 loaded!
Hi,
Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk
source contrib/init.d/rc.redhat.asterisk It started without problem and when
i issue stop now It freezes, please see below,

tai*CLI
add debug   dontdumpextensions  helpiax2
include init
loadlocal   logger  mgcpno  oh323
reload  remove  save
set showsip skinny  softunload
  == Registered channel type 'OH323' (OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.6.3)
  == Parsing '/etc/asterisk/enum.conf':   == Parsing
'/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf':   == Parsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.
tai*CLI stop now
tai*CLI
It freezes right here and does nothing else
-
If I do it with safe_asterisk , it died and loops
[EMAIL PROTECTED] init.d]# /usr/sbin/safe_asterisk -vvc [EMAIL PROTECTED] init.d]#
Asterisk ended with exit status 127 Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127

As I have mentioned, if I noload oh323 this won't happen
*CLI stop now
Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying any remaining musiconhold processes Asterisk cleanly ending
(0).
Any ideas?
Regards,

Anthony
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[Asterisk-Users] strange problem with oh323 loaded!

[Anthony Law]

Hi,

Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk
source contrib/init.d/rc.redhat.asterisk
It started without problem and when i issue stop now It freezes, please
see below,


tai*CLI
add debug   dontdumpextensions  helpiax2
include init
loadlocal   logger  mgcpno  oh323
reload  remove  save
set showsip skinny  softunload
  == Registered channel type 'OH323' (OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.6.3)
  == Parsing '/etc/asterisk/enum.conf':   == Parsing
'/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf':   == Parsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.

tai*CLI stop now
tai*CLI

It freezes right here and does nothing else

-
If I do it with safe_asterisk , it died and loops

[EMAIL PROTECTED] init.d]# /usr/sbin/safe_asterisk -vvc
[EMAIL PROTECTED] init.d]# Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127



As I have mentioned, if I noload oh323 this won't happen

*CLI stop now
Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying any remaining musiconhold processes
Asterisk cleanly ending (0).

Any ideas?


Regards,



Anthony

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RE: [Asterisk-Users] strange problem with oh323 loaded!

[Scott Stingel]

Same problem here - with latest 0.6.3a oh323.  Locks up on exit.  Had to
kill -9

This didn't happen with 0.6.2a, but that's on a different machine.  Maybe
you could try this older version which worked fine (same PwLib and OpenH323)

Regards
Scott 


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law
Sent: Friday, July 02, 2004 1:15 PM
To: Mailing List Asterisk
Subject: [Asterisk-Users] strange problem with oh323 loaded!

Hi,

Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk
source contrib/init.d/rc.redhat.asterisk It started without problem and when
i issue stop now It freezes, please see below,


tai*CLI
add debug   dontdumpextensions  helpiax2
include init
loadlocal   logger  mgcpno  oh323
reload  remove  save
set showsip skinny  softunload
  == Registered channel type 'OH323' (OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.6.3)
  == Parsing '/etc/asterisk/enum.conf':   == Parsing
'/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf':   == Parsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.

tai*CLI stop now
tai*CLI

It freezes right here and does nothing else

-
If I do it with safe_asterisk , it died and loops

[EMAIL PROTECTED] init.d]# /usr/sbin/safe_asterisk -vvc [EMAIL PROTECTED] init.d]#
Asterisk ended with exit status 127 Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127



As I have mentioned, if I noload oh323 this won't happen

*CLI stop now
Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying any remaining musiconhold processes Asterisk cleanly ending
(0).

Any ideas?


Regards,



Anthony

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RE: [Asterisk-Users] strange problem with oh323 loaded!

[T. Chan]

Dear All,

I don't know but I tried all 0.6.x version of OH323 and normally I use
safe_asterisk to start asterisk, and everytime when I use 'stop now' to
terminate asterisk, it does not do anything, and you are rite, I have to use
kill -9 to kill the PIDs and threads. However, if I use asterisk -vvvgc to
start, 'stop now' works.

Thanks all

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: Friday, July 02, 2004 4:37 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] strange problem with oh323 loaded!


Same problem here - with latest 0.6.3a oh323.  Locks up on exit.  Had to
kill -9

This didn't happen with 0.6.2a, but that's on a different machine.  Maybe
you could try this older version which worked fine (same PwLib and OpenH323)

Regards
Scott


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law
Sent: Friday, July 02, 2004 1:15 PM
To: Mailing List Asterisk
Subject: [Asterisk-Users] strange problem with oh323 loaded!

Hi,

Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk
source contrib/init.d/rc.redhat.asterisk It started without problem and when
i issue stop now It freezes, please see below,


tai*CLI
add debug   dontdumpextensions  helpiax2
include init
loadlocal   logger  mgcpno  oh323
reload  remove  save
set showsip skinny  softunload
  == Registered channel type 'OH323' (OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.6.3)
  == Parsing '/etc/asterisk/enum.conf':   == Parsing
'/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf':   == Parsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.

tai*CLI stop now
tai*CLI

It freezes right here and does nothing else

-
If I do it with safe_asterisk , it died and loops

[EMAIL PROTECTED] init.d]# /usr/sbin/safe_asterisk -vvc [EMAIL PROTECTED] init.d]#
Asterisk ended with exit status 127 Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127



As I have mentioned, if I noload oh323 this won't happen

*CLI stop now
Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying any remaining musiconhold processes Asterisk cleanly ending
(0).

Any ideas?


Regards,



Anthony

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RE: [Asterisk-Users] strange problem with oh323 loaded!

[T. Chan]

Hi, Anthony, can you try issuing stop now on safe_asterisk and see if it
works please? I am used to using safe_asterisk and with this new version and
when I tried issuing stop now, it did not do it.

Thanks



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anthony Law
Sent: Wednesday, June 30, 2004 4:17 PM
To: Mailing List Asterisk
Subject: [Asterisk-Users] strange problem with oh323 loaded!


Hi,

I am using asterisk CVS 2004-06-16 with oh323-0.6.3a
I have a strange problem if I start asterisk with oh323 loaded

/usr/sbin/asterisk -vc

once I am in the console and issue restart now or reload asterisk hangs
and it not stoping or restarting at all, below is the console logging when
it happens, as you can see it stucks on Destroying any remaining
musiconhold processes

 [chan_oh323.so] = (OpenH323 Channel Driver)
  == Parsing '/etc/asterisk/rtp.conf': Found
  == Parsing '/etc/asterisk/oh323.conf': Found
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5, PWlib v1.6.6
  == Registered channel type 'OH323' (OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.6.3)
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.
*CLI restart now
Beginning asterisk restart
Executing last minute cleanups
  == Cleaning up OpenH323 channel driver.
  == Unregistered channel type 'OH323'
  == Destroying any remaining musiconhold processes

If I do not load oh323 the above will not happen. Does anyone knows how to
why or how to fix? Even if I use safe_asterisk it acts the same. Is this a
problem with oh323 or asterisk itself?



Regards,



Anthony


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[Asterisk-Users] strange problem with oh323 loaded!

[Anthony Law]

Hi,

I am using asterisk CVS 2004-06-16 with oh323-0.6.3a
I have a strange problem if I start asterisk with oh323 loaded

/usr/sbin/asterisk -vc

once I am in the console and issue restart now or reload asterisk hangs
and it not stoping or restarting at all, below is the console logging when
it happens, as you can see it stucks on Destroying any remaining
musiconhold processes

 [chan_oh323.so] = (OpenH323 Channel Driver)
  == Parsing '/etc/asterisk/rtp.conf': Found
  == Parsing '/etc/asterisk/oh323.conf': Found
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5, PWlib v1.6.6
  == Registered channel type 'OH323' (OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.6.3)
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.
*CLI restart now
Beginning asterisk restart
Executing last minute cleanups
  == Cleaning up OpenH323 channel driver.
  == Unregistered channel type 'OH323'
  == Destroying any remaining musiconhold processes

If I do not load oh323 the above will not happen. Does anyone knows how to
why or how to fix? Even if I use safe_asterisk it acts the same. Is this a
problem with oh323 or asterisk itself?



Regards,



Anthony


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[Asterisk-Users] strange problem with SIP/voicemail

[Matthew Simpson]

I'm having a very strange problem I've been fighting with all day.  It's
2:30am, and I'm stuck.  I think the problem may lie with one of my SIP
providers, but I'm not sure.

I have two ways to call into my test Grandstream.  I can call a PSTN 360
area code number that will forward to my FWD number, which in turn is
registered with my * box on extension 2030.  If I call the 360 number,
everything works, my Grandstream rings, and if I don't answer, it goes to
voicemail and voicemail works.

I also have a PSTN 972 area code number that forwards directly to my * box.
If I call the 972 number, my Grandstream will ring, but if I don't answer,
it will give me silence for a bit, then I hear a click, my CLI interface
says that it is recording a message, but then it says:

Apr 19 02:21:20 WARNING[15373]: app_voicemail.c:1261 play_and_record: No
audio available on SIP/66.147.170.34-0811abe8??

Here is my exten map [actual phone number munged].  I have removed the
Grandstream from the exten for this example.  It makes no difference whether
the Grandstream gets rang or not:

exten = 9725551212,1,Answer
exten = 9725551212,2,Voicemail2(u1000)
exten = 9725551212,3,Hangup

Also, just for testing, I have added this extension:

exten = 2501,1,Voicemail2(u1000)
exten = 2501,2,Hangup

If I dial 2501 from my grandstream, voicemail works that way, too.

My questions:

1) Should I have the Answer in there or not?  It doesn't help to add or
remove it.  On the FWD number, I do not have an Answer.

2) I can get voicemail to work on the incoming 972 number if I change the
dialplan around and then do a restart gracefully.  Example:

exten = 9725551212,1,Answer
exten = 9725551212,2,Playback(transfer)
exten = 9725551212,3,Voicemail2(u1000)
exten = 9725551212,4,Hangup

It will work once, maybe twice, and then it won't work any more after that
until I fiddle with the dialplan again and do another restart.  On Saturday
when I thought I had all of this working, I dialed in at least ten times and
had no problems.

I originally was running a CVS from 03-14-04 now I am running 04-19-04, and
still have the same issue.

Anyone?

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[Asterisk-Users] Strange Problem

[Asterisk Learner]

I am experiencing a strange problem and wanted to know if someone
has faced any similar issues or could provide me with a way to counter this
problem. I am in the process of experimenting with asterisk and trying to setup
a basic functional system. I have one TDM400P (single port) and one X100P. I am
using one analog phone connected to the TDM400P and I also have a couple of
Xlite SIP phones configured. I can make calls out to the PSTN and I can also
receive calls. 



The problem happens when someone from the outside (PSTN)
calls the Asterisk box. I have asterisk configured to forward the call to Zap/2
(analog phone). Zap/2 rings and I can talk to the person on the other end but
if I hang up first then the other end does not see as the call being hung up.
Asterisk CLI shows that Zap/1-1 (FXO) hungup but for some reason the other end
thinks that the call is still up and does not disconnect unless the person
hangs up himself. The confusing part is that if I initiate the call then this
problem does not happen. 



Can someone tell me what is happening and how to resolve
this issue?



Thanks

[Asterisk-Users] Strange problem

[Matt Riddell]

One of my winblows machines died last week (too much adware, spyware, and a
bad windows update etc) and I needed to reinstall.

Before it died, IAXcomm and Firefly both worked (with the exception that
firefly only rang for 30ms and then passed me the audio from the remote end
of the call).

Since reinstalling, audio from IAXcomm has become incredibly choppy (I am
just dialing in to Voicemail).  Firefly is still the same.

I'm not using anywhere near my full bandwidth with either softphone.

I have enabled DMA etc on hard disks and have optimised the system back to
what it was.

It would appear they are both using GSM codec, and I am at a loss as to what
may have caused this to happen.

If anyone has any ideas they would be much appreciated as long as it isn't
install linux, as I would, but this machine needs to be running some music
creation software which is not available under linux (buzztracker).

All other system software runs as before including the cpu/memory intensive
java development environment.

Has anyone seen anything similar? I realise that it may be hard to detect as
most would assume bandwidth to be the problem.

The PC is running the latest updates to WIN98SE on 1000Mhz with 296Mb RAM.

Any ideas would be greatly appreciated.

Kind regards,

Matt Riddell

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Re: [Asterisk-Users] Strange problem

[Matt Riddell]

Having reread my post I guess it could be the successful download of windows
updates...

maybe?

Can anyone confirm/deny?

Matt

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[Asterisk-Users] strange problem with grandstream software 1.0.4.39

[Roy Sigurd Karlsbakk]

hi all

I have a strange problem that started right after an upgrade from 
1.0.3.81: Every now and then the display flashes 484 when the phone is 
idle, on hook. Early Dial is disabled, and I don't understand anything. 
Everything works fine apart from this annoying flashing...

Anyone that knows what this might be?

regards

roy

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[Asterisk-Users] Strange problem with call hangup on Budgetone 102 Phones

[Jon Fautley]

Hi,

I've got Asterisk configured and working (sort of) 
with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). 
This * box is on a 'live', non-nat IP address.
I also have a couple of budgetone phones, one 
behind NAT and one not. When I place an outgoing call, I get the following 
messages:

-- Executing Dial("SIP/filbert-9876", 
"CAPI/288:333") in new stack -- creating pipe for 
PLCI=-1  sent CONNECT_REQ MN 
=0x5 -- Called 288:333 -- Setting up 
echo canceller (PLCI=0x201, function=1, options=2, 
tail=64)  sent FACILITY_REQ 
(PLCI=0x201) -- CAPI[contr1/288]/0 answered 
SIP/filbert-9876 -- Echo canceller successfully set up 
(PLCI=0x201)WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request) -- CAPI 
Hangingup  sent DISCONNECT_B3_REQ 
NCCI=0xa0201  sent DISCONNECT_REQ 
PLCI=0x201 -- removed pipe for PLCI = 0x201 == 
Spawn extension (sip, 9333, 1) exited non-zero on 
'SIP/filbert-9876'WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): 
Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 103 (Request)
I can hear the voicemail service (extn. 333) answer 
correctly, but then after about 5 seconds i'll get the WARNING message and the 
system will hangup.

Here's a snippet from my sip.conf 
file:

---
[general]port = 5060bindaddr = 
0.0.0.0
context = 
sip-incomingsrvlookup=noqualify=yesdisallow=allallow=alawallow=ulaw

[filbert]type=friendhost=dynamicdtmfmode=infocontext=sipcallerid="Jon 
Fautley" 
200nat=yespickupgroup=1reinvite=nocanreinvite=nodisallow=allallow=ulaw


Any ideas?

Many thanks,

Jon

[Asterisk-Users] Strange problem with * and festival

[Alexandru Coseru]

I'm trying to use festival with * and for an unknown reason , it fails..

Here is a small debug:

*CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing Festival(H323:20231, just a test) in new stack
  == Parsing '/etc/asterisk/festival.conf': Found
  == Spawn extension (default, 500, 1) exited non-zero on 'H323:20231'
ClearCallThread::ClearCallThread: Object initialized.
-- Hungup 'H323:20231'
WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
ClearCallThread::ClearCallThread: Object deleted.

I'm using the lastest * from CVS.
My festival is patched and compiled and it is working just fine..
* even creates the file used for cache and that file is ok too..


I've done some debugging and found out that the exit point is located in
app_festival.c  near line 161.
Here is the code:
 if (res  1) {
res = -1;
break;
}


Can somebody tell me what's wrong ?

Thanks a lot
Alex

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Re: [Asterisk-Users] Strange problem with * and festival

[Brian West]

Do you answer the channel first?

exten = s,1,Answer
exten = s,2,Festival,Asterisk rocks!!


bkw
On Thu, 13 Nov 2003, Alexandru Coseru wrote:

 I'm trying to use festival with * and for an unknown reason , it fails..

 Here is a small debug:

 *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created.
 -- Executing Festival(H323:20231, just a test) in new stack
   == Parsing '/etc/asterisk/festival.conf': Found
   == Spawn extension (default, 500, 1) exited non-zero on 'H323:20231'
 ClearCallThread::ClearCallThread: Object initialized.
 -- Hungup 'H323:20231'
 WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
 ClearCallThread::ClearCallThread: Object deleted.

 I'm using the lastest * from CVS.
 My festival is patched and compiled and it is working just fine..
 * even creates the file used for cache and that file is ok too..


 I've done some debugging and found out that the exit point is located in
 app_festival.c  near line 161.
 Here is the code:
  if (res  1) {
 res = -1;
 break;
 }


 Can somebody tell me what's wrong ?

 Thanks a lot
 Alex

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[Asterisk-Users] Strange Problem with Asterisk....

[Andy Hester]

Wondered if anybody might have some ideas about what could be causing
this

I have a T100p hooked up to an Adit 600 with 12 channels of voice off of a
T-1 coming in.
I have a t100p connected to a zhone z-plex with 24 fxs going to my stations.

Some of the station are 2 line phones. These have 2 zap channels that are
dialed when the extension is matched.
Also, some extension require ringing 2 phones at the same time. For example
the boss wants his assistants phone to ring also when someone calls him.

Then the calls roll to a receptionist if not answered.
Finally the call goes to voicemail

This has been working fine for weeks now, but today the customer told me
that one of the phones configured this way was hearing 3 to 4 other
conversations on their phone.  The phone did not give dialtone, but in
picking up the handset you could hear someone shuffling paper working at
their desk etc.  When certain other extensions were on calls, you could hear
clearly the person they were connected to, but not the person in the office.
I checked all the wiring and couldn't find any problem.  I rebooted the
channelbank thinking maybe somehow it had gotten screwy.  No change.  Then,
just for grins, I stopped and restarted *.  Voila, the problem was gone.

Can anyone think of a scenario that could cause * to do this?

Thanks,
Andy

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Re: [Asterisk-Users] Strange Problem with Asterisk....

[Mark Spencer]

I think there are issues with combining flash-hook supervised transfers
with meetme conference bridges.  Can you find out if that took place, i.e.
someone tries to transfer into a meetme conference?

Mark

On Thu, 6 Nov 2003, Andy Hester wrote:

 Wondered if anybody might have some ideas about what could be causing
 this

 I have a T100p hooked up to an Adit 600 with 12 channels of voice off of a
 T-1 coming in.
 I have a t100p connected to a zhone z-plex with 24 fxs going to my stations.

 Some of the station are 2 line phones. These have 2 zap channels that are
 dialed when the extension is matched.
 Also, some extension require ringing 2 phones at the same time. For example
 the boss wants his assistants phone to ring also when someone calls him.

 Then the calls roll to a receptionist if not answered.
 Finally the call goes to voicemail

 This has been working fine for weeks now, but today the customer told me
 that one of the phones configured this way was hearing 3 to 4 other
 conversations on their phone.  The phone did not give dialtone, but in
 picking up the handset you could hear someone shuffling paper working at
 their desk etc.  When certain other extensions were on calls, you could hear
 clearly the person they were connected to, but not the person in the office.
 I checked all the wiring and couldn't find any problem.  I rebooted the
 channelbank thinking maybe somehow it had gotten screwy.  No change.  Then,
 just for grins, I stopped and restarted *.  Voila, the problem was gone.

 Can anyone think of a scenario that could cause * to do this?

 Thanks,
 Andy

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RE: [Asterisk-Users] Strange Problem with Asterisk....

[Mark Spencer]

 No, I have Meetme on an extension, but the users don't even know about it
 and the ext# for it is in a completely different range.

Hrm okay.

 I don't know if I explained it adequately looking back on my post.  The
 situation persisted over 8 to 10 hours and through numerous calls.  what
 seems weird to me is that the other extensions functioned correctly, but the
 one didn't. I could pick up the handset and be listening to the sounds of
 someone working quietly at their desk, and then if they made a call, I could
 hear the ringing, the person answer, but not the caller.  this happened over
 and over.  But it didn't pick up the audio of all the other zap channels,
 just 4 or 5.

You can do zap show channel foo to see what the state of a particular
zap channel is.  The only situation i've ever heard of anything like this
in is related to MeetMe and flashhook transfer being combined.  If it
happens to get in the state again, let me know.  Also, check the call
detail records to see if there were any unusual calls at the time.

Mark

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