Re: [asterisk-users] Polycom IP 4000 Soundstation SIP Conference PhoneQuestion

[Anthony Rodgers]

Hi Matt,

We have one and it works very well - usual Polycom quality, as others
have attested. The only thing we have noticed is a reluctance to
download its config files via FTP when using a VLAN tag.

CP

Matt wrote:
>
> Hi,
> Has anyone here ever used a Polycom IP 4000 Soundstation SIP
> Conference Phone with asterisk?  If so, how well does it work and how
> does it sound?
>
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Re: [asterisk-users] Poor sound quality on incoming calls

[Ryan Parlee]

Matthew,

I've read through the link and I believe my kernel is setup correctly,
however I am still experiencing this problem. 

I do not have a Digium card which this link says may be required, however, I
have no problems with my other Asterisk box running Asterisk 1.2.7.1 and
Linux 2.4.18-3.  That box has neither a Digium card nor the zaptel modules
and it runs fine.  Unfortunately I can no longer use that old box and my
brand-new box isn't working.  

I've really tried to figure this out myself but I'm not making any progress.


To hear what we're experiencing, call 515 333 4030

If anyone can help me out on this and is interesting in seeing my kernel
config or something else please let me know. 

Thanks, 
Ryan
 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew J.
Roth
Sent: Tuesday, July 24, 2007 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Poor sound quality on incoming calls

Ryan Parlee wrote:
> I am experiencing extreme jitter/slowdown on Playback() or Background().
> I've looked thoroughly on voip-info.org and elsewhere for help regarding
> this issue but cannot figure this out.  I can make outgoing calls with no
> problems.
>
> When I run zttest I get the following:
>
> Opened pseudo zap interface, measuring accuracy...
> -34.204102% -20.263672% -24.218750% -26.586914% -23.852539% -13.256836%
> -25.207520%
> -13.366699% -15.441895% -14.550781% -22.351074% -21.765137%
> --- Results after 12 passes ---
> Best: 0.00 -- Worst: -34.204102 -- Average: -21.255493
>   
Ryan,

For some tips on fixing the playback issues, see this post: 
http://lists.digium.com/pipermail/asterisk-users/2007-June/189978.html

First, you may want to investigate why you are getting such poor 
accuracy from zttest.  It seems that you have an underlying problem.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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Re: [asterisk-users] Digium adaper S101I - IAXy Losing connection

[Russell Bryant]

Joseph wrote:
> Thanks, do you know what "iaxy" firmware version is asterisk 1.4 using?
> My asterisk-1.2.21.1 is using  "iaxy version 23"

It is the same version in 1.4, as well.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] What is the best softphone work with Asterisk

[Ryan Stille]

I've had decent luck with PhonerLite, connecting via SIP.  The interface 
is not the best, but I've been able to connect reliably and make calls.

-Ryan

bilal ghayyad wrote:
> Hi List;
>
> I need to configure a softphone to be client and use
> it with Asterisk, which is the recommended one? Is it
> iax2?
>
> Regards
> Bilal
>
>
>
> 
> Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for 
> today's economy) at Yahoo! Games.
> http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  
>
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>   



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Re: [asterisk-users] What is the best softphone work with Asterisk

[Time Bandit]

> I need to configure a softphone to be client and use
> it with Asterisk, which is the recommended one? Is it
> iax2?

You can try my IAX2 softphone for windows :
http://www.marccharbonneau.com/asterisk/mediaxphone.php

Hope it fits your need

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Re: [asterisk-users] What is the best softphone work with Asterisk

[Jim Archer]

I tried several and had very poor luck with each I tried.   These included 
IaxComm, IaxComm Pro, Diax and Firefly II.  Also, One other one from I 
think Germany that had just changed it's name.  All of these had issues.  I 
could not get Firefly configured at all to talk to Asterisk.  Diax, when 
the user places a call, just keeps ringing even when the person answered. 
Both IaxComms would crash.  I'm sure there is one out there but I have not 
found it, although I have not yet tried the SIP soft phones.

--On Tuesday, July 24, 2007 2:09 PM -0700 bilal ghayyad 
<[EMAIL PROTECTED]> wrote:

> Hi List;
>
> I need to configure a softphone to be client and use
> it with Asterisk, which is the recommended one? Is it
> iax2?
>
> Regards
> Bilal
>
>
>
> _
> ___ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now
> (it's updated for today's economy) at Yahoo! Games.
> http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
>
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Re: [asterisk-users] MySQL components in asterisk-addons not being built

[hugolivude]

Thanks again.  Ive posted the rpm query & configure.log below.  I
noticed that the mysql libraries were in /usr/lib/mysql/ so I tried
./configure --with-mysqlclient=/usr/lib/mysql but end up with the
following:

configure: ***
configure: *** mysql_config was not found in the path you specified:
configure: *** /usr/lib/mysql/bin
configure: *** including --without-mysqlclient

Further pointers welcome!

Thanks
Hugh


rpm query
===
[EMAIL PROTECTED] asterisk-addons-1.4.2]# rpm -ql MySQL-devel | grep client
/usr/lib/mysql/libmysqlclient.a
/usr/lib/mysql/libmysqlclient.la
/usr/lib/mysql/libmysqlclient_r.a
/usr/lib/mysql/libmysqlclient_r.la
/usr/lib/mysql/libndbclient.a
/usr/lib/mysql/libndbclient.la

[EMAIL PROTECTED] asterisk-addons-1.4.2]# rpm -ql MySQL-client | grep client
[EMAIL PROTECTED] asterisk-addons-1.4.2]# rpm -ql MySQL-server | grep client

Doesn't find anything with:
rpm -ql MySQL-client | grep client
or
rpm -ql MySQL-server | grep client


config.log snip
===
configure:6158: checking for mysql_config
configure:6176: found /usr/bin/mysql_config
configure:6188: result: /usr/bin/mysql_config
configure:6220: checking for mysql_init in -lmysqlclient
configure:6255: gcc -o conftest -g -O2   conftest.c -lmysqlclient
-L/usr/lib/mysql -lmysqlclient -lz -lcrypt -lnsl -lm -lc -lnss_files
-lnss_dns -lresolv -lc -lnss_files -lnss_dns -lresolv  >&5
/usr/lib/mysql/libmysqlclient.a(ssl.o)(.gnu.linkonce.d.__vt_Q25yaSSL7Message+0x8):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(ssl.o)(.gnu.linkonce.d.__vt_Q25yaSSL7Message+0xc):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(ssl.o)(.gnu.linkonce.d.__vt_Q25yaSSL7Message+0x10):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(ssl.o)(.gnu.linkonce.d.__vt_Q25yaSSL7Message+0x14):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(ssl.o)(.gnu.linkonce.d.__vt_Q25yaSSL7Message+0x18):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(cert_wrapper.o)(.text+0x77): In
function `yaSSL::x509::~x509(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(cert_wrapper.o)(.text+0x30f): In
function `yaSSL::CertManager::~CertManager(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.t._._Q25mySTLt4list1ZPQ25yaSSL11SSL_SESSION+0x41):
In function `mySTL::list::~list(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.t._._Q25mySTLt4list1ZPQ25yaSSL12input_buffer+0x41):
In function `mySTL::list::~list(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.t._._Q25mySTLt4list1ZPQ25yaSSL13output_buffer+0x41):
In function `mySTL::list::~list(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.t._._Q25mySTLt4list1ZPUc+0x41):
more undefined references to `__builtin_delete' follow
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.d.__vt_Q25yaSSL13HandShakeBase+0x8):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.d.__vt_Q25yaSSL13HandShakeBase+0xc):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.d.__vt_Q25yaSSL13HandShakeBase+0x10):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(template_instnt.o)(.gnu.linkonce.d.__vt_Q25yaSSL13HandShakeBase+0x14):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(yassl_imp.o)(.text+0x2a62): In
function `yaSSL::ServerDHParams::~ServerDHParams(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(yassl_imp.o)(.text+0x4b3b): In
function `yaSSL::Connection::~Connection(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(yassl_int.o)(.text+0x32cd): In
function `yaSSL::SSL_SESSION::~SSL_SESSION(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(yassl_int.o)(.text+0x348b): In
function `yaSSL::Sessions::~Sessions(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(yassl_int.o)(.text+0x37db): In
function `yaSSL::SSL_CTX::~SSL_CTX(void)':
: undefined reference to `__builtin_delete'
/usr/lib/mysql/libmysqlclient.a(yassl_int.o)(.text+0x3c60): more
undefined references to `__builtin_delete' follow
/usr/lib/mysql/libmysqlclient.a(libtaocrypt_la-asn.o)(.gnu.linkonce.d.__vt_Q28TaoCrypt4HASH+0xc):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(libtaocrypt_la-asn.o)(.gnu.linkonce.d.__vt_Q28TaoCrypt4HASH+0x10):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(libtaocrypt_la-asn.o)(.gnu.linkonce.d.__vt_Q28TaoCrypt4HASH+0x14):
undefined reference to `__pure_virtual'
/usr/lib/mysql/libmysqlclient.a(libtaocrypt_la-asn.o)(.gnu.lin

[asterisk-users] Asterisk 1.2.23 and 1.4.9 released

[The Asterisk Development Team]

The Asterisk development team has released Asterisk versions 1.2.23 and
1.4.9.

These releases contain bug fixes, including one for a security vulnerability.
The vulnerability is a potential Denial of Service attack when the Asterisk
IAX2 channel driver is configured to allow unauthenticated calls.

We have released an Asterisk Security Advisory for the vulnerability.  The 
current version of the advisory can be downloaded from the ftp site.

http://ftp.digium.com/pub/asa/ASA-2007-018.pdf
 * Affected systems include all Asterisk installations running an affected 
version
that allow unauthenticated IAX2 calls.  Affected open source versions include
1.2.20 through 1.2.22, and 1.4.5 through 1.4.8.

All users that have systems that meet the criteria listed above should 
upgrade as soon as possible.

Thank you very much for your support.


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[asterisk-users] ASA-2007-018: Resource Exhaustion vulnerability in IAX2 channel driver

[The Asterisk Development Team]

  Asterisk Project Security Advisory -

   
++
   |  Product   | 
Asterisk  |
   
|+---|
   |  Summary   | Resource Exhaustion vulnerability in IAX2 
channel |
   || 
driver|
   
|+---|
   | Nature of Advisory | Denial of 
Service |
   
|+---|
   |   Susceptibility   | Remote Unauthenticated 
Sessions   |
   
|+---|
   |  Severity  | 
Moderate  |
   
|+---|
   |   Exploits Known   | 
No|
   
|+---|
   |Reported On | July 19, 
2007 |
   
|+---|
   |Reported By | Russell Bryant, Digium, Inc. 
<[EMAIL PROTECTED]> |
   
|+---|
   | Posted On  | July 23, 
2007 |
   
|+---|
   |  Last Updated On   | July 23, 
2007 |
   
|+---|
   |  Advisory Contact  | Russell Bryant 
<[EMAIL PROTECTED]>   |
   
|+---|
   |  CVE Name  
|   |
   
++

   
++
   | Description | The IAX2 channel driver in Asterisk is vulnerable to 
a   |
   | | Denial of Service attack when configured to 
allow|
   | | unauthenticated calls. An attacker can send a flood 
of   |
   | | NEW packets for valid extensions to the server 
to|
   | | initiate calls as the unauthenticated user. This 
will|
   | | cause resources on the Asterisk system to get 
allocated  |
   | | that will never go away. Furthermore, the IAX2 
channel   |
   | | driver will be stuck trying to 
reschedule|
   | | retransmissions for each of these fake calls 
for |
   | | forever. This can very quickly bring down a system 
and   |
   | | the only way to recover is to restart 
Asterisk.  |
   | 
|  |
   | | Detailed 
Explanation:|
   | 
|  |
   | | Within the last few months, we made some changes 
to  |
   | | chan_iax2 to combat the abuse of this module for 
traffic |
   | | amplification attacks. Unfortunately, this has caused 
an |
   | | unintended side 
effect.  |
   | 
|  |
   | | The summary of the change to combat 
traffic  |
   | | amplification is this. Once you start the PBX on 
the |
   | | Asterisk channel, it will begin receiving frames to 
be   |
   | | sent back out to the network. We delayed this 
from   |
   | | happening until a 3-way handshake has occurred to 
help   |
   | | ensure that we are talking to the IP address 
the |
   | | messages appear to be coming 
from.   |
   | 
|  |
   | | When chan_iax2 accepts an unauthenticated call, 
it   |
   | | immediately creates the ast_channel for the 
call.|
   | | However, since the 3-way handshake has not 
been  |
   | | completed, the PBX is not started on this 
channel.   |
   | 
|  |
   | | Later, when the maximum number of retries have 
been  |
   | | exceeded on responses to this NEW, the code tries 
to |
   | | hang up the call. Now, it has 2 ways to do 
this, |
   | | depending on if there i

Re: [asterisk-users] Answering Machine Beep Detection for *

[Nasir Iqbal]

Hi dave,

you can use AMD application 

for more info please visit
www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD


Regards

Nasir Iqbal

On Tue, 2007-07-24 at 17:22 -0400, dave cantera wrote:
> hi,
> can anyone point me to answering machine beep detection methods or writeups 
> for *?
> thanks,
> daveC
> 
> 
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Re: [asterisk-users] Voicemail .lock- files voicemail box not accessible

[Eric \"ManxPower\" Wieling]

JR Richardson wrote:
> Hi All,
> 
> Strange issue, recently I started getting a lot of .lock files in the
> voicemail /INBOX folder preventing proper access to voicemail.  I can
> delete the .lock files and everything is normal.  After searching
> around, I found some SIP lock file stuff but nothing specific to
> voicemail.
> 
> Can someone point me in the right direction to resolve this?  I'm
> runnning 1.2.9 on Debian Sarge.

The only time I've ever seen this is if you are running out of disk 
space on the partition voicemail is stored on.  I imagine this could 
also happen if Asterisk crashed while someone was leaving voicemail.


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Re: [asterisk-users] MySQL components in asterisk-addons not being built

[James FitzGibbon]

On 7/24/07, hugolivude <[EMAIL PROTECTED]> wrote:


Thanks or all your help!

I've posted the ./configure output below.  I noticed that it says:

   checking for mysql_init in -lmysqlclient... no

Presumably that's a problem, but I don't know how to fix it!!  As I
mentioned, I have MySQL installed and it works fine.  rpm -qa
indicates:

   MySQL-server-5.0.22-0
   MySQL-devel-5.0.22-0
   MySQL-client-5.0.22-0

How do I get mysql_init set up properly, if indeed that is the source
of my problem?



Post the snippet of config.log that deals with mysql, as that will give more
detail as to why it's not finding mysql_init.  For example, mine has this:

configure:6161: checking for mysql_config
configure:6179: found /usr/bin/mysql_config
configure:6191: result: /usr/bin/mysql_config
configure:6223: checking for mysql_init in -lmysqlclient
configure:6258: gcc -o conftest -g -O2   conftest.c -lmysqlclient
-L/usr/lib64/mysql -lmysqlclient -lz -lcrypt -lnsl -lm -L/usr/lib64 -lssl
-lcrypto  >&5
configure:6264: $? = 0
configure:6282: result: yes

Yours will likely have several iterations of trying to find it, indicating
that the script is looking for the mysql libs in several directories and
then when it fails to find it, gives up.  Mine is only one iteration because
it was found in the first place the configure script looked.

Also, post the output of

rpm -ql MySQL-devel | grep client

and

rpm -ql MySQL-client | grep client

from the looks of the RPM package names, you aren't running the same distro
as me (CentOS), but I suspect that the problem is that your RPMs have stuck
the libraries in a non-standard place that the asterisk-addons configure
script doesn't know to look in.

Once you've figured out what that non-standard place is, it should be a
simple matter of passing "--with-mysqlclient=PATH" to ./configure to make it
look for your libs in their actual home.

--
j.
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Re: [asterisk-users] TDM04B & FIOS No Hangups Often

[Nasir Iqbal]

Hi Mike,

I think that callprogress=yes is right.

but Noah Miller also right.

so the solution will be.

callprogress=yes
busydetect=yes
busycount=4 ; suitable values is above then 4 choose minimum 
; as More value more time to wait before hangup.

rxgain=7; you can adjust this value by error & trial 
; method check with different values. and 
; choose what works best for you. 
; note: if you face invalid hangup try to reduce rxgain
; or If you face no-hangup problem try to increase rxgain


I hope it will work for you.


Regards

Nasir Iqbal


On Tue, 2007-07-24 at 16:03 -0500, Eric "ManxPower" Wieling wrote:
> Noah Miller wrote:
> 
> > 2) Set callprogress=yes in zapata.conf (if you haven't already done that).
> 
> If you set callprogress=yes you will have the opposite problem -- active 
> calls will be randomly disconnected.
> 
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Re: [asterisk-users] CallerID from POTS to SIP

[astuser]

Thanks for the reply.  Unfortunately that didn't work.  What's confusing 
is that for the line without any distinctive ring that works correctly 
with callerid, the only thing it does is dial the phones, so here's the 
entire context:

[add-incoming]
exten => s,1,Dial(SIP/ht1&SIP/ht2&SIP/gxp1,20)

The other context, the one with distinctive ring that's not passing 
caller id, actually does a little more:

[main-open]
exten => s,1,Answer
exten => s,n,Wait(3)
exten => s,n,Background(opengreeting)
exten => s,n,Dial(SIP/ht1&SIP/gxp3,20)

But even if I remove those extra bits from that context and make it look 
like the other one, it still doesn't work.

Any more suggestions?  Anyone else have experience with distinctive ring 
and caller id?  Is my syntax even right for getting callerid through 
distinctive ring for asterisk from SVN?

Steve

On Mon, Jul 23, 2007 at 11:26:45PM -0700, Ira wrote:
> At 08:47 PM 7/23/2007, you wrote:
> 
> >For both lines I can definitely see the callerid being reported correctly
> >in /var/log/asterisk/cdr-custom/Master.csv.  It's just not getting passed
> >through to the sip clients for the incoming line with distinctive ring.
> >For those lines it shows up as "asterisk".
> 
> Long ago when I was having CID problems I added a wait(3) before the 
> dial() in my incoming context on my ZAP lines and most of the CID 
> problems went away.

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Re: [asterisk-users] MySQL components in asterisk-addons not being built

[hugolivude]

Thanks or all your help!

I've posted the ./configure output below.  I noticed that it says:

   checking for mysql_init in -lmysqlclient... no

Presumably that's a problem, but I don't know how to fix it!!  As I
mentioned, I have MySQL installed and it works fine.  rpm -qa
indicates:

   MySQL-server-5.0.22-0
   MySQL-devel-5.0.22-0
   MySQL-client-5.0.22-0

How do I get mysql_init set up properly, if indeed that is the source
of my problem?

Thanks,
Hugh

checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking target system type... i686-pc-linux-gnu
checking for uname... /bin/uname
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking for g++... g++
checking whether we are using the GNU C++ compiler... yes
checking whether g++ accepts -g... yes
checking for a BSD-compatible install... /usr/bin/install -c
checking whether ln -s works... yes
checking for GNU make... make
checking for grep... /bin/grep
checking for basename... /bin/basename
checking for dirname... /usr/bin/dirname
checking for sh... /bin/sh
checking for ln... /bin/ln
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... (cached) /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking for initscr in -lcurses... yes
checking curses.h usability... yes
checking curses.h presence... yes
checking for curses.h... yes
checking for initscr in -lncurses... yes
checking for curses.h... (cached) yes
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... no
checking for asterisk.h... yes
configure: creating ./config.status
config.status: creating build_tools/menuselect-deps
config.status: creating makeopts

   .$$$=..
.$7$7..  .7$$7:.
  .$$:. ,$7.7
.$7. 7   .$$77
 ..$$.   $.$$$7
..7$   .?.   $   .?.   7$$$.
   $.$.   .$$$7. 7 .7$$$.  .$$$.
 .777.   .$$77$$$77$7.  $$$,
 $$$~  .7$7.   .$$$.
.$$7  .7$$$7:  ?$$$.
$$$  ?7$$I.$$$7
$$$   .7  :$$$.
$$$   $$7.$$$.
$$$$$$   7$$$7  .$$$.$$$.
 7 .$$$.
7$$$777$$$
 $$$$
  7.   $$  (TM)
   $$$.   .7$$  $$
 7$.$$
   .

configure: Package configured for:
configure: OS type  : Linux
configure: Host CPU : i686

On 7/24/07, Nasir Iqbal <[EMAIL PROTECTED]> wrote:
> Hi,
>
> please see your ./configure output especially few last lines.
>
> and note missing thins.
>
> Regards
>
> Nasir Iqbal
>
> On Tue, 2007-07-24 at 10:45 -0400, hugolivude wrote:
> > Thanks Tahir.  I already got the asterisk-addons though - that's what
> > I'm having trouble with!   BTW - asterisk-addons also provides a
> > menuselect now.  The problem is that the MySQL components all show up
> > XXX even though I have MySQL installed.
> >
> > Hugh
> >
> > On 7/24/07, Tahir Almas <[EMAIL PROTECTED]> wrote:
> > > Hi Hugh,
> > >
> > > MySQL CDR is not included in default asterisk distribution. so there is
> > > no entry for MySQL CDR in "make menuselect". you must install additional
> > > Addons after asterisk installation. you can download from
> > >
> > > http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.2.tar.gz
> > >
> > > 1. Extract them
> > > 2. Make
> > > 3. Make Install
> > > 4. cdr_addon_mysql.so will be installed including all other modules.
> > >
> > > Regards
> > >
> > > Nasir Iqbal
> > >
> > >
> > > On Tue, 2007-07-24 at 08:17 -0400, hugolivude wrote:
> > > > I'm trying to add MySQL CDR recording in Asterisk 1.4.6.  I'm
> > > > following the instructions posted here:
> > > >
> > > > http://www.voip-info.org/wiki-Asterisk+cdr+mysql
> > > >
> > > > I have MySQL installed and it works fine - starts on stratup, I can
> > > > create DBs, tables and so on and I can connect through php.  rpm -qa
> > > > indicates:
> > > >
> > > > MySQL-server-5.0.22-0
> > > > MySQL-devel-5.0.22-0
> > > > MySQL-client-5.0.22-0
> > > >
> > > > However I still get XXX for all of the MySQL add ons when I do:
> > > >
> > > > make menuselect
> > >

[asterisk-users] Answering Machine Beep Detection for *

[dave cantera]

hi,
can anyone point me to answering machine beep detection methods or writeups for 
*?
thanks,
daveC


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[asterisk-users] How I can configure asterisk to register as gatekeeper server with another gatekeeper

[bilal ghayyad]

Hi List;

Is there a link that help me to configure asterisk to
register to another gatekeeper as client?

Regards
Bilal


   
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[asterisk-users] Testers needed for VoIP router solution

[Robert Augustyn]

Hi all,
We have put together a firmware for a range of inexpensive routers.
It has been configured to provide optimum VoIP performance.
We have internally tested it for number of months and it looks very good.
You should be able to run it easily with 20+  phones on local network ( we
still did not hit the upper limit ) assuming that you have bandwidth.
Your VoIP will get prioritized over other types of traffic.
You should be able to talk, download files and run torrents at the same time
with no visible degradation of the VoIP voice quality.
It will be delivered ready to upload with all your configurations, which you
will have to provide to us.
We will custom build firmware for your configuration.
We just ask you to upload it, test it and provide feedback.
 
If you are interested ( sorry only first 10 will be accepted ) please
contact me at firmware at linqone dot com and we will send you the set of
questions we need you to answer before we can build a solution for you.
Thanks,
 
 This firmware will work on: 

*   Linksys WRT54G v1-v4, WRT54GS v1-v4,
 WRT54GL v1.x,
 WRTSL54GS (no USB
support) 

*   Buffalo
 WHR-G54S,
 WHR-HP-G54, WZR-G54,
WBR2-G54 

*   Asus
 WL500G Premium
(no USB support) 

This will not work on Linksys WRT54G/GS v5-v7 or newer WRT54G/GS routers.
 
If you do not have any of the above routers you can get one for UNDER $40
shipped at:
 
 

http://www.circuitcity.com/ccd/Search.do?c=1&context=&keyword=Buffalo+WHR-G5
4S&searchSection=All&go.x=11&go.y=10 
 
How do I find my Linksys WRT54G/WRT54GS/WRT54GL's version?
Look at the bottom side of the router to check for the version number, or
compare the first 4 characters of the serial number with the following list:

CDF0/CDF1 = WRT54G v1.0
CDF2/CDF3 = WRT54G v1.1
CDF5 = WRT54G v2.0
CDF7 = WRT54G v2.2
CDF8 = WRT54G v3.0
CDF9 = WRT54G v3.1
CDFA = WRT54G v4.0

CGN0/CGN1 = WRT54GS v1.0
CGN2 = WRT54GS v1.1
CGN3 = WRT54GS v2.0
CGN4 = WRT54GS v2.1
CGN5 = WRT54GS v3.0
CGN6 = WRT54GS v4.0

CL7A = WRT54GL v1.0
CL7B = WRT54GL v1.1


If it's not listed above, and it's not a WRT54GL, it's not supported. 
 
 
 
Sincerely,
Robert Augustyn
 
This firmware is provided as-is without any warranty. I will NOT be
responsible for damages that occur due to the use of this firmware. USE AT
YOUR OWN RISK.
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Re: [asterisk-users] Dial out through multiple Zap groups

[C F]

This should do what you want:

You can call it like this:
exten => 
12125551212,1,Macro(dialoutbound,${EXTEN},8005551212,3-Zap/G1/-Zap/G2/-Sip/Nufone/)

The above using the macro below will try zap/g1 first if it's in use
or otherwise unavailable, ti will go to zap/g2 and then sip/nufone.

[macro-dialoutbound]
;arg1 number to dial
;arg2 callerid
;arg3 device in form of: devicecount-device/resourc-device/resource as
many as matching devicecount
;when busy it will play busy
;when channelunavail, it will play congestion


exten => s,1,Noop()
exten => s,2,Noop()
exten => s,3,Noop()
exten => s,4,GotoIf($[${LEN(${CALLERID(num)})} > 7]?100);if we got cid
longer than 7 then it's an outside number so we leave it
exten => s,5,Set(CALLERID(num)=${ARG2})
exten => s,6,Goto(10)

exten => s,10,Noop()
exten => s,11,Noop(Weare starting to cut)
exten => s,12,Set(DCNT=${CUT(ARG3,,1)})
exten => s,13,Set(CNT=2)
exten => s,14,Goto(50);thats where we assign the DVC var

exten => s,50,Noop(We start assigning devices)
exten => s,51,Noop()
exten => s,52,Set(TCNT=$[${CNT} - 2])
exten => s,53,GotoIf($[${TCNT} = ${DCNT}]?800);congestion
exten => s,54,Set(DVC=${CUT(ARG3,-,${CNT})})
exten => s,55,Set(TCNT=${CNT})
exten => s,56,Set(CNT=$[${TCNT} + 1]);here we increment it
exten => s,57,Goto(callme,1)

exten => s,100,Noop(not setting CID, since we got one)
exten => s,101,Noop()
exten => s,102,Goto(10)

exten => s,800,Noop()
exten => s,801,Congestion()
exten => s,802,Hangup()

exten => callme,1,Noop()
exten => callme,2,Dial(${DVC}${ARG1},,Ww)
exten => callme,3,Goto(${DIALSTATUS},1)
exten => callme,103,Goto(3)

exten => CHANUNAVAIL,1,Noop()
exten => CHANUNAVAIL,2,Noop()
exten => CHANUNAVAIL,3,Goto(s,50)

exten => CONGESTION,1,Goto(CHANUNAVAIL,1)

exten => NOANSWER,1,Goto(s,800)

exten => BUSY,1,Noop()
exten => BUSY,2,Noop()
exten => BUSY,3,Playtones(busy)
exten => BUSY,4,Busy()

Hope this helps.


On 7/24/07, Vieri <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm trying to set a rule to dial out through multiple
> Zap groups so that, say, g0 is the cheaper POTS lines
> group
> and must be used first. However, if g0 is busy or
> disconnected then try dialing out g1.
>
> My g0 group is made up of 4 analog lines connected to
> a 4-FXO card. I disconnected the RJ-11 wires from the
> FXO card
> to simulate a line disconnection. So theoretically all
> calls should immediately go out through g1 but they
> don't.
> They get "stuck" on g0 as I can see in the asterisk
> CLI:
>
> -- Executing Dial("SIP/4053-082393a8",
> "ZAP/g0/5|120|TWm") in new stack
> -- Called g0/5
> -- Started music on hold, class 'default', on
> SIP/4053-082393a8
> -- Zap/32-1 answered SIP/4053-082393a8
> -- Stopped music on hold on SIP/4053-082393a8
> (endless)
>
> Note: Zap channel 32 is part of g0.
>
> I used both FreePBX and a custom made rule.
> With FreePBX, the outgoing dialplan includes something
> like this:
>
> exten =>
> _5,1,Macro(dialout-trunk,1,${EXTEN},,)
> exten =>
> _5,n,Macro(dialout-trunk,2,${EXTEN},,)
> exten => _5,n,Macro(outisbusy,)
> ; trunk 1 is g0, trunk 2 is g1
>
> If I use a custom dialpan that looks something like
> this:
>
> exten => _5,1,Dial(Zap/g0/${EXTEN})
> exten => _5,n,NoOp(${DIALSTATUS})
> exten => _5,n,Dial(Zap/g1/${EXTEN})
> exten => _5,n,HangUp()
>
> and then watch the CLI, I get exactly the same
> behavior as above, ie. I don't get past
> Dial(Zap/g0/${EXTEN}) as
> Zap/32 answers when it shouldn't. And obviously I
> can't get ${DIALSTATUS} to eventually define some
> gotos because it's ANSWERED.
>
> Any ideas as to what I should try?
> Maybe change some setting in zapata.conf?
>
> Thanks
>
> Vieri
>
>
>
>   
> 
> Shape Yahoo! in your own image.  Join our Network Research Panel today!   
> http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
>
>
>
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[asterisk-users] What is the best softphone work with Asterisk

[bilal ghayyad]

Hi List;

I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?

Regards
Bilal


   

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Re: [asterisk-users] TDM04B & FIOS No Hangups Often

[Eric \"ManxPower\" Wieling]

Noah Miller wrote:

> 2) Set callprogress=yes in zapata.conf (if you haven't already done that).

If you set callprogress=yes you will have the opposite problem -- active 
calls will be randomly disconnected.

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Re: [asterisk-users] Dial out through multiple Zap groups

[Eric \"ManxPower\" Wieling]

You cannot detect disconnected analog lines in Asterisk.  You can't even 
determine of the lines have dialtone.  All you can do is determine if 
there is a current active asterisk managed call.



Noah Miller wrote:
> Hi Vieri -
> 
>> I'm trying to set a rule to dial out through multiple
>> Zap groups so that, say, g0 is the cheaper POTS lines
>> group
>> and must be used first. However, if g0 is busy or
>> disconnected then try dialing out g1.
>>
>> My g0 group is made up of 4 analog lines connected to
>> a 4-FXO card. I disconnected the RJ-11 wires from the
>> FXO card
>> to simulate a line disconnection. So theoretically all
>> calls should immediately go out through g1 but they
>> don't.
>> They get "stuck" on g0 as I can see in the asterisk
> 
> You've discovered a big limitation in analog lines.  If this were a
> PRI or BRI, the lines would behave as you want, but analog lines
> won't.
> 
> You can try using ChanIsAvail() to test beforehand if the zap channels
> will accept a call.  I don't know if it will work in this fashion
> (I've never tried).
> 
> 
> - Noah
> 
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> 


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Re: [asterisk-users] Dial out through multiple Zap groups

[Eric \"ManxPower\" Wieling]

Vieri wrote:
> --- Noah Miller <[EMAIL PROTECTED]> wrote:
> 
>> You can try using ChanIsAvail() to test beforehand
>> if the zap channels
>> will accept a call. 
> 
> I'll try that. Thanks Noah.
> My test was with "disconnected" analog lines.
> I will also try to do the same but this time will keep
> the line busy by placing a call and then see if *
> detects it as BUSY and tries to place the new call on
> the next trunk.

group=1
channel = 1-5
group = 1,2
channel 6-10

As you can see a channel can be in more than 1 group.


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Re: [asterisk-users] SIP IP Trunk, between Asterisk and Softswitch

[Jaswinder Singh]

In your case it will send calls without registering to softswitch . Btw what
does your softswitch expects from asterisk ? like is it configured to
authenticate by username alone , user/pass or ip address ?? People here  can
help you better if you post that info .


On 24/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:


Dear List;

I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.

I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username to be sent
for my softswitch as sometimes the softswitch need to
check it.

Also, does asterisk request to register on the
softswitch or it can send directly without
registeration? (Note: the trunk is SIP).

Please check the below configuration and advise me if
it is correct:

[aloonet]
type=peer
qualify=yes
host=193.111.196.240 ; IP Address of the softswitch
canreinvite=yes
context=outbound
disallow=all
allow=g723
nat=no

Is it OK? Will it register on my softswitch or will
send call directly without registeration on it?

Regards
Bilal





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Re: [asterisk-users] rxFAX core dumps

[Sylvain Boily]

Hi,

Install spanDSP 0.0.2-pre26 not 0.0.3.

Le mardi 24 juillet 2007 à 17:02 -0300, [EMAIL PROTECTED] a
écrit :
> Hi Everyone...
> 
> I am running Asterisk 1.2.22 on Debian "Etch".  I installed it from  
> sources.  I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded
> http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c
> http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_txfax.c
> and a Digium TDM card with 4 FXO ports
> 
> When my dialplan send an incoming call from ZAP channel to sendme a fax using
> RxFAX, this application core dump and crashes asterisk. But When I test
> RxFax from a sip phone works fine. For this I tuned zapata.conf disabling
> both:
> ;faxdetect=incoming
> ;faxdetect=outgoing
>  
> And I changed the 'fax' extension by 'elsefaxext'. Again no works and RxFAX
> application core dumps. I switch many versions of tiff and spandsp libraries
> and does not work.
> 
> I have a hungup detection problem too. This issue can affect fax handling?. To
> solve this problem I set this on zapata for each channel:
> busydetect=yes
> busycount=10
> busypattern=500,500
> hanguponpolarityswitch=yes
> callprogress=yes
> 
> But does not work and while a caller hangs asterisk continue playing greetings
> or whatever.
> 
> Can someone give me a hint as to how to solve this or else point me at
> some docs?
> 
> Thanks very much.
> 
> 
> 
> 
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Re: [asterisk-users] SIP IP Trunk, between Asterisk and Softswitch

[Jared Smith]

On Tue, 2007-07-24 at 13:27 -0700, bilal ghayyad wrote:
> Is it OK? Will it register on my softswitch or will
> send call directly without registeration on it?

No, your configuration wouldn't register to the softswitch... but it
will send a call directly to the softswitch.  (Registration only tells
the softswitch what your Asterisk box's IP address is, but doesn't have
anything to do with you sending calls to the softswitch.)

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] SIP IP Trunk, between Asterisk and Softswitch

[bilal ghayyad]

Dear List;

I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.

I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username to be sent
for my softswitch as sometimes the softswitch need to
check it.

Also, does asterisk request to register on the
softswitch or it can send directly without
registeration? (Note: the trunk is SIP).

Please check the below configuration and advise me if
it is correct:

[aloonet]
type=peer
qualify=yes 
host=193.111.196.240 ; IP Address of the softswitch
canreinvite=yes 
context=outbound 
disallow=all
allow=g723
nat=no 

Is it OK? Will it register on my softswitch or will
send call directly without registeration on it?

Regards
Bilal


   

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[asterisk-users] rxFAX core dumps

[ggonzalez]

Hi Everyone...

I am running Asterisk 1.2.22 on Debian "Etch".  I installed it from  
sources.  I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded
http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c
http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_txfax.c
and a Digium TDM card with 4 FXO ports

When my dialplan send an incoming call from ZAP channel to sendme a fax using
RxFAX, this application core dump and crashes asterisk. But When I test
RxFax from a sip phone works fine. For this I tuned zapata.conf disabling
both:
;faxdetect=incoming
;faxdetect=outgoing
 
And I changed the 'fax' extension by 'elsefaxext'. Again no works and RxFAX
application core dumps. I switch many versions of tiff and spandsp libraries
and does not work.

I have a hungup detection problem too. This issue can affect fax handling?. To
solve this problem I set this on zapata for each channel:
busydetect=yes
busycount=10
busypattern=500,500
hanguponpolarityswitch=yes
callprogress=yes

But does not work and while a caller hangs asterisk continue playing greetings
or whatever.

Can someone give me a hint as to how to solve this or else point me at
some docs?

Thanks very much.




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Re: [asterisk-users] phone directory with asterisk

[John Faubion]

> I can only pointing out this issue, trying to prevent bad user experience
with these phones.

> I'm very very unhappy about the situation but it's out ouf my reach to
change this.

Yes, and I certainly understand how difficult it can be to get upper
management to understand that their cost cutting/joint venture baby is also
cutting into customer satisfaction. One can only hope that you can get
through management before it takes too much toll on the company. Good luck!

John
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Re: [asterisk-users] Digium adaper S101I - IAXy Losing connection

[Joseph]

On Tue, 2007-07-24 at 13:32 -0400, Jared Smith wrote:
> On Tue, 2007-07-24 at 10:48 -0600, Joseph wrote:
> > I have one Digium adapter S101I on a local network and I'm losing the
> > connection periodically.I'm using Asterisk-1.2.21.1
> 
> I know there have been some recent fixes to the IAX implementation in
> Asterisk to handle a few scenarios when IAX connections would drop
> unexpectedly.  If I were you, I'd try with a newer version of Asterisk
> (and would definitely try the 1.4.x series as well).
> 
> -Jared

Thanks, do you know what "iaxy" firmware version is asterisk 1.4 using?
My asterisk-1.2.21.1 is using  "iaxy version 23"

-- 
#Joseph

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[asterisk-users] mISDN & Asterisk 1.4: HFC-S card not responsive

[Arik Raffael Funke]

Hi,

I have installed Asterisk 1.4 with mISDN with the 
install-asterisk.tar.gz script from beronet.com. On my system I have two 
cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to 
work well with mISDN on my system from a previous installation.

Now however, the AVM card works well at first glance, i.e. it 
"registers" incoming calls and works through the asterisk dialplan. 
Calls on the hfc card however seem to be completely ignored. There is 
not the slightest indication in asterisk that call come in. The CLI 
stays completely silent even for debug and verbose levels of 100 for 
core and misdn!

The HFC-S card however does seem to be not completely ignored by 
asterisk: if I plug-in or remove connection with a high misdn debug 
level, it shows the "usual" messages - as it also does for the AVM card. 
Only incoming calls are ignored - n.b. outgoing do not work either...

Below are outputs from the CLI (misdn show config, misdn show stacks, 
pluggin in cable, removing cable), dmesg and lspci -v. I hope somebody 
could give me a hint as to what could be the problem. The system is 
freshly installed and both cards are configured identically.

Cheers,
Arik



= CLI: misdn show config = (n.b. port 1=hfcpci; port 2=avmfritz)
*CLI> misdn show config
Misdn General-Config:
  -> misdn_init: /etc/misdn-init.conf -> debug: 0
  -> tracefile: /var/log/asterisk/misdn.log -> bridging: no 

  -> stop_tone_after_first_digit: yes -> append_digits2exten: yes
  -> dynamic_crypt: no-> crypt_prefix: **
  -> crypt_keys: test,muh -> ntdebugflags: 0
  -> ntdebugfile: /var/log/misdn-nt.log

[PORT 1]
  -> name: intern -> allowed_bearers: all
  -> far_alerting: no -> rxgain: 0
  -> txgain: 0-> te_choose_channel: no
  -> pmp_l1_check: no -> reject_cause: 16
  -> block_on_alarm: no   -> hdlc: no
  -> context: Intern  -> language: en
  -> musicclass: default  -> callerid:
  -> method: standard -> dialplan: 0
  -> localdialplan: 0 -> cpndialplan: 0
  -> nationalprefix: 0-> internationalprefix: 00
  -> presentation: -1 -> screen: -1
  -> always_immediate: no -> nodialtone: no
  -> immediate: no-> senddtmf: yes
  -> hold_allowed: no -> early_bconnect: yes
  -> incoming_early_audio: no -> echocancel: 0
  -> need_more_infos: no  -> noautorespond_on_setup: no
  -> nttimeout: no-> bridging: yes
  -> jitterbuffer: 4000   -> jitterbuffer_upper_threshold: 0
  -> callgroup:   -> pickupgroup:
  -> max_incoming: -1 -> max_outgoing: -1
  -> l1watcher_timeout: 0 -> overlapdial: 0
  -> msns: *  -> faxdetect: no
  -> faxdetect_context:   -> faxdetect_timeout: 5
  -> ptp: no

[PORT 2]
  -> name: intern -> allowed_bearers: all
  -> far_alerting: no -> rxgain: 0
  -> txgain: 0-> te_choose_channel: no
  -> pmp_l1_check: no -> reject_cause: 16
  -> block_on_alarm: no   -> hdlc: no
  -> context: Intern  -> language: en
  -> musicclass: default  -> callerid:
  -> method: standard -> dialplan: 0
  -> localdialplan: 0 -> cpndialplan: 0
  -> nationalprefix: 0-> internationalprefix: 00
  -> presentation: -1 -> screen: -1
  -> always_immediate: no -> nodialtone: no
  -> immediate: no-> senddtmf: yes
  -> hold_allowed: no -> early_bconnect: yes
  -> incoming_early_audio: no -> echocancel: 0
  -> need_more_infos: no  -> noautorespond_on_setup: no
  -> nttimeout: no-> bridging: yes
  -> jitterbuffer: 4000   -> jitterbuffer_upper_threshold: 0
  -> callgroup:   -> pickupgroup:
  -> max_incoming: -1 -> max_outgoing: -1
  -> l1watcher_timeout: 0 -> overlapdial: 0
  -> msns: *  -> faxdetect: no
  -> faxdetect_context:   -> faxdetect_timeout: 5
  -> ptp: no
*CLI>



= CLI: misdn show stacks =
*CLI> misdn show stacks
BEGIN STACK_LIST:
   * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:UP Blocked:0  Debug:1
   * Port 2 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0  Debug:1
*CLI>


= CLI: when plugging in hfc card =
*CLI> misdn set debug 100
changing debug level for all ports to 100
*CLI>
*CLI> P[ 0] Got empty Msg..
P[ 0] MGMT: Short status dinfo 101
P[ 0] MGMT: SSTATUS: L1_ACTIVATED
P[ 0] Got empty Msg..
*CLI>



= CLI: removing cable from hfc card =
*CLI> P[ 0] Got empty Msg..
P[ 0] MGMT: Short status dinfo 100
P[ 0] MGMT: SSTATUS: L1_DEACTIVATED
P[ 1] $$$ find_chan: No 

Re: [asterisk-users] Digium adaper S101I - IAXy Losing connection

[Jared Smith]

On Tue, 2007-07-24 at 10:48 -0600, Joseph wrote:
> I have one Digium adapter S101I on a local network and I'm losing the
> connection periodically.I'm using Asterisk-1.2.21.1

I know there have been some recent fixes to the IAX implementation in
Asterisk to handle a few scenarios when IAX connections would drop
unexpectedly.  If I were you, I'd try with a newer version of Asterisk
(and would definitely try the 1.4.x series as well).

-Jared


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Re: [asterisk-users] SIP jitter buffer and asterisk native bridge

[Russell Bryant]

Damon Estep wrote:
> Thanks a bunch. So in theory the media gateway at the far end should be
> able to properly jitter buffer the entire RTP path from the ATA via
> asterisk, correct?
> 
> Would this be the same in 1.2 and it 1.4?

Yes, that is correct, but only for 1.4.  In the case of Asterisk 1.2, if the 
media is flowing through Asterisk, then I believe the timestamp information is 
lost.

> The best practice in the example given would be to rely on adaptive
> jitter buffers at the ATA and the media gateway, and not force jitter
> buffers in the SIP<>SIP asterisk bridge (1.4)

Correct.  That would be unless the endpoint has no jitterbuffer.  If that was 
the case, then forcing it to happen in the middle at Asterisk may help, but 
obviously can't fix anything that happens between Asterisk and the endpoint.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] SIP jitter buffer and asterisk native bridge

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Russell Bryant
> Sent: Tuesday, July 24, 2007 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP jitter buffer and asterisk native
bridge
> 
> Damon Estep wrote:
> > Anyone know the answer? Has it been validated with packet captures,
or
> > code review?
> 
> All of the timing information should be passed across the bridge in
all of
> the
> frames that come in over RTP.  I can't say I verified this with packet
> captures,
> but I did look for this in the code review for the jitterbuffer code
in
> 1.4.  I
> know there is explicit code to ensure this is the case.

Russell,

Thanks a bunch. So in theory the media gateway at the far end should be
able to properly jitter buffer the entire RTP path from the ATA via
asterisk, correct?

Would this be the same in 1.2 and it 1.4?

The best practice in the example given would be to rely on adaptive
jitter buffers at the ATA and the media gateway, and not force jitter
buffers in the SIP<>SIP asterisk bridge (1.4)

Damon

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Re: [asterisk-users] Poor sound quality on incoming calls

[Matthew J. Roth]

Ryan Parlee wrote:
> I am experiencing extreme jitter/slowdown on Playback() or Background().
> I've looked thoroughly on voip-info.org and elsewhere for help regarding
> this issue but cannot figure this out.  I can make outgoing calls with no
> problems.
>
> When I run zttest I get the following:
>
> Opened pseudo zap interface, measuring accuracy...
> -34.204102% -20.263672% -24.218750% -26.586914% -23.852539% -13.256836%
> -25.207520%
> -13.366699% -15.441895% -14.550781% -22.351074% -21.765137%
> --- Results after 12 passes ---
> Best: 0.00 -- Worst: -34.204102 -- Average: -21.255493
>   
Ryan,

For some tips on fixing the playback issues, see this post: 
http://lists.digium.com/pipermail/asterisk-users/2007-June/189978.html

First, you may want to investigate why you are getting such poor 
accuracy from zttest.  It seems that you have an underlying problem.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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Re: [asterisk-users] [beginner] Problem of detecting call

[Tzafrir Cohen]

On Tue, Jul 24, 2007 at 03:44:29PM +, karim H wrote:
> Hello,
> I have some problem to start asterisk.
> First I have followed a lot of tutorials to complete correctly the install 
> process. Now it works when I type zttool I can see when I am or not 
> connected to the PSTN.
> But, I run asterisk with  verbose and I can't see the call detection.
> There is no detection of the call.
> 
> I have a X100P card FXO with only one line. So only one channel
> I configured my zaptel.conf like this :
> ___
> loadzone=fr
> defaultzone=fr
> fxsks=1
> ___
> 
> and My zapata.conf like this :
> ___
> [trunkgroups]
> 
> [channels]
> 
> language=fr
> context=from-pstn
> signalling=fxs_ks
> rxwink=300 ; Atlas seems to use long (250ms) winks
> ;
> ; Whether or not to do distinctive ring detection on FXO lines
> ;
> ;usedistinctiveringdetection=yes
> 
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> rxgain=0.0
> txgain=0.0
> group=0
> callgroup=1
> pickupgroup=1
> immediate=no
> 
> ;faxdetect=both
> faxdetect=incoming
> ;faxdetect=outgoing
> ;faxdetect=no

channel => 1

; next time just use zaptel/xpp/utils/genzaptelconf

-- 
   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] Dialplan

[Anselm Martin Hoffmeister]

Am Montag, den 23.07.2007, 14:33 -0400 schrieb Matt:
> Hi,
> What dialplan option do I need to send a call out like this:
> 
> NPA-NXX- local calls
> 1-NPA-NXX- - long distance
> 
> Won't 'national' send it out NPA-NXX- no matter if it's long
> distance or not?

I do not understand your point here. If the user dials 1-212-5551212,
you could send out exactly that string, as in
exten => _1NX,1,Dial(SIP/[EMAIL PROTECTED])

and if she dials 617-1234567, similarly.

Or do you wish Asterisk to magically remove the leading "1", but only
for two or three area codes, because in that case the calls will be
charged as "local calls"? In that case, you might require your users to
_always_ dial the leading "1" and get away with something like

exten => _1617XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _1857XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _1NX,1,Dial(SIP/[EMAIL PROTECTED])

(assuming 617 and 857 are local area codes)

ymmv, and the documentation about pattern in dialplans
http://www.voip-info.org/wiki/index.php?page=Asterisk+config
+extensions.conf
should be the next text you read, probably.

If this is not what you want, please describe your idea.

BR
Anselm (who never owned a landline in the NANP...)


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Re: [asterisk-users] SIP jitter buffer and asterisk native bridge

[Russell Bryant]

Damon Estep wrote:
> Anyone know the answer? Has it been validated with packet captures, or 
> code review?

All of the timing information should be passed across the bridge in all of the 
frames that come in over RTP.  I can't say I verified this with packet 
captures, 
but I did look for this in the code review for the jitterbuffer code in 1.4.  I 
know there is explicit code to ensure this is the case.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] Upgrade Procedure

[Anselm Martin Hoffmeister]

Am Montag, den 23.07.2007, 16:21 -0400 schrieb Michael J. Liberatore:
> I noticed in 1.4.x I can no longer use n+101 ?  I use this all over my
> dial plan and wouldn't even know how to replace it.  Like when trying to
> call out and a channel is busy, would I need to do all if then's???  How
> can I upgrade and keep n+101? 

Please read the documentation, for example at
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial

(other commands can be found linked from
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of
+application+commands
)

There is an additional option you will have to set to the Dial() to
restore the "jump to n+101" behaviour, named "j". So you would for
example change

exten=>123,4,Dial(SIP/sip123,30,w)

to

exten=>123,4,Dial(SIP/sip123,30,jw)

Other commands may also feature such an option, if appropriate - should
be found easily in voip-info.

I _think_ there is also a kind of global option to restore the n+101
behaviour for the entire dialplan (instead of defaulting to setting
variables), actually
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+General
might be your best friend there.

HTH
Anselm


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Re: [asterisk-users] phone directory with asterisk

[Tim Koehler]

Sury, but the phone is no snom phone, it's China trash full of bugs!

snom India is just distributing this trash it.

I can only pointing out this issue, trying to prevent bad user experience
with these phones.

I'm very very unhappy about the situation but it's out ouf my reach to
change this.


Cheers


Tim


On 7/24/07, John Faubion <[EMAIL PROTECTED]> wrote:


 > To prevent further missunderstanding please do not refer the SI-120 as
a snom
> phone. If you need support please contact snom India.

Tim,

If it is sold by snom India, and one is to contact snom India, I can
certainly see how one could infer that it is indeed a snom phone.

John

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--
---
snom technology AG

Tim Koehler
Partner Manager
[EMAIL PROTECTED]
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[asterisk-users] Problem Hangup Help

[FERNANDO VILLARROEL]

Hello list, i need help.

My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP

Asterisk >> SIP ), I get a ring tone. When I
now decide to hang up (e.g. if 

nobody answers), the called telephone continues to
ring almost forever.

the sip.conf:

[2563105] 
accountcode = 2563105
username = 2563105
secret = 135
callerid = 412563105
context = test
canreinvite = no
dtmfmode = rfc2833
host = dynamic
insecure = very
language = es
nat = yes
qualify = yes
type = friend
disallow=all
allow=g729

[nyphone]
accountcode=nyphone
canreinvite=no
reinvite=yes
dtmfmode=rfc2833
host=72.55.143.XXX
insecure=very
language=es
nat=no
qualify=no
type=peer
disallow=all
allow=g729


My extensions.conf

exten => _00X.,1,dial(sip/${EXTEN:[EMAIL PROTECTED],45)
exten => _00X.,2,hangup


Nyphone is my provider for everyone calls
international.

I attach sip debug one call.

I use Asterisk 1.2.13

I hope you understand me and help.

Best regards

Fernando Villarroel Noriel.
Chillan
Chile

Sorry my English.


   

Choose the right car based on your needs.  Check out Yahoo! Autos new Car 
Finder tool.
http://autos.yahoo.com/carfinder/SIP Debugging Enabled for IP: 72.55.143.XXX:5060
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/2563105-0819cf80", "sip/[EMAIL 
PROTECTED]|45") in new stack
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105" ;tag=as726ac50a
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2475 2475 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called [EMAIL PROTECTED]
vaca*CLI> 
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105" ;tag=as726ac50a
Call-ID: [EMAIL PROTECTED]
To: 
Contact: 
Proxy-Authenticate: DIGEST realm="VoipSwitch", 
nonce="118490324119231120007472128429"
Content-Length: 0


<->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 72.55.143.XXX:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105" ;tag=as726ac50a
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105" ;tag=as726ac50a
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="test770", realm="VoipSwitch", 
algorithm=MD5, uri="sip:[EMAIL PROTECTED]", 
nonce="118490324119231120007472128429", 
response="413be923621811a639c3b0e83d3a2e74", opaque=""
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2475 2476 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
vaca*CLI> 
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105" ;tag=as726ac50a
Call-ID: [EMAIL PROTECTED]
To: ;tag=1907470723212675853288937
Contact: 
Content-Type: application/sdp
Content-Length: 215

v=0
o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX
s=VoipSIP
i=Audio Session
c=IN IP4 72.55.143.XXX
t=0 0
m=audio 8936 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<->
--- (9 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.55.143.XXX:8936
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), 
combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event)

[asterisk-users] Digium adaper S101I - IAXy Losing connection

[Joseph]

I have one Digium adapter S101I on a local network and I'm losing the
connection periodically.I'm using Asterisk-1.2.21.1

There is no pattern, sometimes it stays connected for a weak sometimes
longer. 
In comparison I have few Sipura adapters and they stay connected for
months (never lost single connection).

-- 
#Joseph

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[asterisk-users] VoiceTronix 12 + Far-End Hangup Detection

[Gleidson Antonio Henriques]

Hi all,

   I´ve 2 VoiceTronix 12 cards in my Machine with asterisk 1.4.8 running, 
everything is working fine except the detection of far-end hangup.
   There is someone in this list with VoiceTronix 12+asterisk and got 
working the detection ?
   Below are some useful information about drivers and configs.
   Someone could please give me a help ?

   Thanks in Advance,


Gleidson Antonio Henriques

VPB DRIVER - 4.1.29
ASTERISK - 1.4.8
vtcore.conf --

[general]
name=vtcore
channels=24
cards=2

[card0]
dtmfms=70
cutthrough=10.0
type=OpenSwitch12
channels=12
channel0=fxs
channel1=fxs
channel2=fxs
channel3=fxs
channel4=fxs
channel5=fxs
channel6=fxs
channel7=fxs
channel8=fxo
channel9=fxo
channel10=fxo
channel11=fxo

[card1]
dtmfms=70
cutthrough=10.0
type=OpenSwitch12
channels=12
channel0=fxo
channel1=fxs
channel2=fxs
channel3=fxs
channel4=fxs
channel5=fxs
channel6=fxs
channel7=fxs
channel8=fxo
channel9=fxo
channel10=fxo
channel11=fxo

vpb.conf --

[general]
type = v12pci
cards = 2
relaxdtmf = 1
break-for-dtmf = no
ast-dtmf-det=1
indication = 1

[interfaces]

board = 0
txgain = 12
rxgain = 12
txhwgain = 12
rxhwgain = 12
echocancel = on
UseLoopDrop = 1
context = menu-principal
mode = fxo
channel = 8
channel = 9
channel = 10
channel = 11

context = menu-principal
mode = dialtone
channel = 0
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7

board = 1
txgain = 12
rxgain = 12
txhwgain = 12
rxhwgain = 12
echocancel = on
UseLoopDrop = 1
context = menu-principal
mode = fxo
channel = 8
channel = 9
channel = 10
channel = 11

context = menu-principal
mode = dialtone
channel = 0
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7 


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[asterisk-users] Poor sound quality on incoming calls

[Ryan Parlee]

Hello,

I am experiencing extreme jitter/slowdown on Playback() or Background().
I've looked thoroughly on voip-info.org and elsewhere for help regarding
this issue but cannot figure this out.  I can make outgoing calls with no
problems.

I am running Linux 2.6.18-4-686 and Asterisk 1.4.8 and I do not have any
zaptel hardware installed.  

When I run without ztdummy I experience poor sound quality and sound
overlaps in Playback() or Background().  For example, the default busy
message says "The person at extension, at extension, at extension, is
unavailable, please press.. the pound key, the pound key, the pound key"

When I run with ztdummy, the default busy message says "The peson
aaat extension iis unnnavailable..."


I found sites talking about conflicts with Asterisk and ACPI on 2.6 kernels
so I rebuilt my kernel with all ACPI and APM options turned off.  

When I run zttest I get the following:

Opened pseudo zap interface, measuring accuracy...
-34.204102% -20.263672% -24.218750% -26.586914% -23.852539% -13.256836%
-25.207520%
-13.366699% -15.441895% -14.550781% -22.351074% -21.765137%
--- Results after 12 passes ---
Best: 0.00 -- Worst: -34.204102 -- Average: -21.255493



Any help would be appreciated!

Thanks, 
Ryan 



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Re: [asterisk-users] phone directory with asterisk

[John Faubion]

> To prevent further missunderstanding please do not refer the SI-120 as a
snom
> phone. If you need support please contact snom India.

Tim,

If it is sold by snom India, and one is to contact snom India, I can
certainly see how one could infer that it is indeed a snom phone.

John
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[asterisk-users] SIP jitter buffer and asterisk native bridge

[Damon Estep]

There is a theory that says that jitter buffers should not be used until
the end of the voice path where jitter might be introduced. With that in
mind, and in this scenario, the jitter buffers should reside at the ATA
and media gateway;

 

ATA (SIP UA)  <> ASTERISK NATIVE BRIDGE <> MEDIA GATEWAY (SIP TO TDM)

 

That raises a question about the Asterisk Native Bridge; Are the UDP RTP
packets bridged in such a way that out of order packet arrivals between
the ATA and asterisk can still be buffered and corrected at the media
gateway, or are the RTP sequence numbers re-written by the Asterisk
native bridge so the media gateway is now unaware that they are not in
the same order as they were initially transmitted?

 

Anyone know the answer? Has it been validated with packet captures, or
code review?

 

Thanks a bunch!

 

Damon

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[asterisk-users] exit ChanSpy with DTMF

[GDrayer]

Any thoughts?  Am I stuck with modifying chanspy itself to allow an exit
DTMF?

>Message: 6
>Date: Thu, 12 Jul 2007 08:43:17 -0400
>From: <[EMAIL PROTECTED]>
>Subject: [asterisk-users] exit ChanSpy with DTMF
>To: 
>Message-ID:
>
<[EMAIL PROTECTED]>1advertisi
ng.com>
>   
>Content-Type: text/plain;  charset="us-ascii"
>
>Part of a supervisor menu I'm writing requires that I allow the
>supervisor to choose to ChanSpy a channel from the main menu then
return
>back to the menu to choose other options when she's done.  Is there a
>way to 'exit' ChanSpy and continue down the dialplan?  Or is a caller
>stuck in ChanSpy until they hangup the phone?
>
>Thanks.
>George

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[asterisk-users] [beginner] Problem of detecting call

[karim H]

Hello,
I have some problem to start asterisk.
First I have followed a lot of tutorials to complete correctly the install 
process. Now it works when I type zttool I can see when I am or not 
connected to the PSTN.
But, I run asterisk with  verbose and I can't see the call detection.
There is no detection of the call.

I have a X100P card FXO with only one line. So only one channel
I configured my zaptel.conf like this :
___
loadzone=fr
defaultzone=fr
fxsks=1
___

and My zapata.conf like this :
___
[trunkgroups]

[channels]

language=fr
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
_

Could someone give me a way to start, an URL, something to read to 
understand why it isn't detected.


PS : My X100P is a digium true one. I work under ubuntu 7.04.And I already 
run apache red5 on my computer.

Thanks a lot for your help

Kheraud

_
Windows Live Spaces : créez votre blog à votre image ! 
http://www.windowslive.fr/spaces


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Re: [asterisk-users] MySQL components in asterisk-addons not being built

[Nasir Iqbal]

Hi,

please see your ./configure output especially few last lines.

and note missing thins.

Regards 

Nasir Iqbal

On Tue, 2007-07-24 at 10:45 -0400, hugolivude wrote:
> Thanks Tahir.  I already got the asterisk-addons though - that's what
> I'm having trouble with!   BTW - asterisk-addons also provides a
> menuselect now.  The problem is that the MySQL components all show up
> XXX even though I have MySQL installed.
> 
> Hugh
> 
> On 7/24/07, Tahir Almas <[EMAIL PROTECTED]> wrote:
> > Hi Hugh,
> >
> > MySQL CDR is not included in default asterisk distribution. so there is
> > no entry for MySQL CDR in "make menuselect". you must install additional
> > Addons after asterisk installation. you can download from
> >
> > http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.2.tar.gz
> >
> > 1. Extract them
> > 2. Make
> > 3. Make Install
> > 4. cdr_addon_mysql.so will be installed including all other modules.
> >
> > Regards
> >
> > Nasir Iqbal
> >
> >
> > On Tue, 2007-07-24 at 08:17 -0400, hugolivude wrote:
> > > I'm trying to add MySQL CDR recording in Asterisk 1.4.6.  I'm
> > > following the instructions posted here:
> > >
> > > http://www.voip-info.org/wiki-Asterisk+cdr+mysql
> > >
> > > I have MySQL installed and it works fine - starts on stratup, I can
> > > create DBs, tables and so on and I can connect through php.  rpm -qa
> > > indicates:
> > >
> > > MySQL-server-5.0.22-0
> > > MySQL-devel-5.0.22-0
> > > MySQL-client-5.0.22-0
> > >
> > > However I still get XXX for all of the MySQL add ons when I do:
> > >
> > > make menuselect
> > >
> > > Any pointers for me on how to troubleshoot and fix this problem?
> > >
> > > Thanks,
> > > Hugh
> > >
> > > ___
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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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Re: [asterisk-users] MySQL components in asterisk-addons not being built

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

hugolivude wrote:
> Thanks Tahir.  I already got the asterisk-addons though - that's what
> I'm having trouble with!   BTW - asterisk-addons also provides a
> menuselect now.  The problem is that the MySQL components all show up
> XXX even though I have MySQL installed.
> 
> Hugh

Using menuselect, highlight the MySQL option, then look down to see what
packages are required.  Don't forget to ./configure after loading the
packages before rerunning menuselect.

Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.6 (GNU/Linux)

iD8DBQFGphdmCFu3bIiwtTARAsw1AKCtOHNyvAgqMzzyBBYOH/VeaOYcdACfZtWu
SGjXzBf+8N+BqldyFithAzI=
=yKkt
-END PGP SIGNATURE-

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Re: [asterisk-users] bristuff for hfc card on Xscale 80219

[Stelios Koroneos]

Is your system compiled as BE or LE ?
bristuff will compile but will not work on BE systems without some
patches/endianess fixes as some of the buffer pointers are little endian
Also with the existing bristuff you will get invalid data due to the fact
that the cache controller of xscale can not snoop into the DMA transfer
cycles and update the cache
To avoid this you need to allocate a non cachable memory region for buffer.
In sort. Although it compiles it does not work :/


Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Thomas Winter
> Sent: Wednesday, July 18, 2007 1:44 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] bristuff for hfc card on Xscale 80219
>
>
> Hi,
>
> compile and load of modules works fine.
>
> After ztcfg I can see
> .
> .
> Changing signalling on channel 1 from Unused to Clear channel
> Changing signalling on channel 2 from Unused to Clear channel
> Changing signalling on channel 3 from Unused to HDLC with FCS check
>
> and then the board is frozen.
>
> Any ideas?
>
> regards
> Thomas
>
>
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Re: [asterisk-users] MySQL components in asterisk-addons not being built

[hugolivude]

Thanks Tahir.  I already got the asterisk-addons though - that's what
I'm having trouble with!   BTW - asterisk-addons also provides a
menuselect now.  The problem is that the MySQL components all show up
XXX even though I have MySQL installed.

Hugh

On 7/24/07, Tahir Almas <[EMAIL PROTECTED]> wrote:
> Hi Hugh,
>
> MySQL CDR is not included in default asterisk distribution. so there is
> no entry for MySQL CDR in "make menuselect". you must install additional
> Addons after asterisk installation. you can download from
>
> http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.2.tar.gz
>
> 1. Extract them
> 2. Make
> 3. Make Install
> 4. cdr_addon_mysql.so will be installed including all other modules.
>
> Regards
>
> Nasir Iqbal
>
>
> On Tue, 2007-07-24 at 08:17 -0400, hugolivude wrote:
> > I'm trying to add MySQL CDR recording in Asterisk 1.4.6.  I'm
> > following the instructions posted here:
> >
> > http://www.voip-info.org/wiki-Asterisk+cdr+mysql
> >
> > I have MySQL installed and it works fine - starts on stratup, I can
> > create DBs, tables and so on and I can connect through php.  rpm -qa
> > indicates:
> >
> > MySQL-server-5.0.22-0
> > MySQL-devel-5.0.22-0
> > MySQL-client-5.0.22-0
> >
> > However I still get XXX for all of the MySQL add ons when I do:
> >
> > make menuselect
> >
> > Any pointers for me on how to troubleshoot and fix this problem?
> >
> > Thanks,
> > Hugh
> >
> > ___
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

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Re: [asterisk-users] Upgrade Procedure

[Andrew Latham]

read this 
http://www.tuxtone.com/index.php/VOIP:Asterisk_Dial_Plan

On 7/23/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote:
> I noticed in 1.4.x I can no longer use n+101 ?  I use this all over my
> dial plan and wouldn't even know how to replace it.  Like when trying to
> call out and a channel is busy, would I need to do all if then's???  How
> can I upgrade and keep n+101?
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Noah
> Miller
> Sent: Monday, July 23, 2007 3:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Upgrade Procedure
>
> > >> You have to first uninstall your Asterisk1.2 like this--
> > >>
> > >> First you have to stop your asterisk...using--
> > >>
> > >> 1. killall -9 asterisk or killall -9 safe_asterisk, whichever you
> are using.
> > >
> > > In my experience, you don't need to do this step.  In fact, you can
> > > keep the old asterisk running, compile and install asterisk 1.4 on
> > > top of it.  Then issue a "restart when convenient" command from the
> > > asterisk 1.2 prompt, and Asterisk 1.4 will come up after the
> restart.
> >
> > The problem with this is that the upgraded Zaptel will not be active.
> > Compile and install Zaptel, LibPRI and Asterisk (in the order), then
> > stop asterisk, unload the zaptel drivers, then load everything.
>
> I've found that you don't really need to do a full stop of asterisk
> either.  Just compile and install both zaptel and asterisk.  Issue the
> "restart when convenient", and after asterisk restarts, then restart
> zaptel (unload old version and load new version).
>
>
> - Noah
>
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>
>
> This E-mail, including any attachments, may be intended solely for
> the personal and confidential use of the sender and recipient(s) named
> above. This message may include advisory, consultative and/or
> deliberative material and, as such, would be privileged and confidential
> and not a public document. Pursuant to 42 CFR, any information in this
> e-mail identifying a former, present, or potential client of Straight & 
> Narrow is confidential. If you have received this e-mail in error, you must 
> not review, transmit, convert to hard copy, copy, use or disseminate this 
> e-mail or any attachments to it and you must delete this message. You are 
> requested to notify the sender by return e-mail.
>
>
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-- 
/*
 Andrew Latham
 LATHAMA (lay-th-ham-eh)
 [EMAIL PROTECTED]
 [EMAIL PROTECTED]
*/

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Re: [asterisk-users] Dial out through multiple Zap groups

[Vieri]

--- Noah Miller <[EMAIL PROTECTED]> wrote:

> You can try using ChanIsAvail() to test beforehand
> if the zap channels
> will accept a call. 

I'll try that. Thanks Noah.
My test was with "disconnected" analog lines.
I will also try to do the same but this time will keep
the line busy by placing a call and then see if *
detects it as BUSY and tries to place the new call on
the next trunk.



   

Building a website is a piece of cake. Yahoo! Small Business gives you all the 
tools to get online.
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Re: [asterisk-users] G729 with SIP and H.323

[Dovid B]

I was able to get H.323 to work with G729 on 1.2.18. It works real well.

- Original Message - 
From: "Cesc Santa" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, July 24, 2007 1:54 AM
Subject: [asterisk-users] G729 with SIP and H.323


> Hi,
>
> I need an Asterisk with G729 support. Preference is with Asterisk
> 1.2(.18), but if not possible, then it can be 1.4.
> Question is, can I enable G729 for both protocols? do the H323
> implementation allow it? I found the codec support for H323 in 1.2.18
> very poor ... only got u/a-law to work ... not even GSM.
> Would the Digium G729 license be good both for SIP and H323?
>
> Cesc
>
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Re: [asterisk-users] Upgrade Procedure

[Dovid B]

try priorityjumping=yes in extensions.conf

- Original Message - 
From: "Michael J. Liberatore" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, July 23, 2007 11:21 PM
Subject: Re: [asterisk-users] Upgrade Procedure


>I noticed in 1.4.x I can no longer use n+101 ?  I use this all over my
> dial plan and wouldn't even know how to replace it.  Like when trying to
> call out and a channel is busy, would I need to do all if then's???  How
> can I upgrade and keep n+101?
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Noah
> Miller
> Sent: Monday, July 23, 2007 3:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Upgrade Procedure
>
>> >> You have to first uninstall your Asterisk1.2 like this--
>> >>
>> >> First you have to stop your asterisk...using--
>> >>
>> >> 1. killall -9 asterisk or killall -9 safe_asterisk, whichever you
> are using.
>> >
>> > In my experience, you don't need to do this step.  In fact, you can
>> > keep the old asterisk running, compile and install asterisk 1.4 on
>> > top of it.  Then issue a "restart when convenient" command from the
>> > asterisk 1.2 prompt, and Asterisk 1.4 will come up after the
> restart.
>>
>> The problem with this is that the upgraded Zaptel will not be active.
>> Compile and install Zaptel, LibPRI and Asterisk (in the order), then
>> stop asterisk, unload the zaptel drivers, then load everything.
>
> I've found that you don't really need to do a full stop of asterisk
> either.  Just compile and install both zaptel and asterisk.  Issue the
> "restart when convenient", and after asterisk restarts, then restart
> zaptel (unload old version and load new version).
>
>
> - Noah
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> This E-mail, including any attachments, may be intended solely for
> the personal and confidential use of the sender and recipient(s) named
> above. This message may include advisory, consultative and/or
> deliberative material and, as such, would be privileged and confidential
> and not a public document. Pursuant to 42 CFR, any information in this
> e-mail identifying a former, present, or potential client of Straight & 
> Narrow is confidential. If you have received this e-mail in error, you 
> must not review, transmit, convert to hard copy, copy, use or disseminate 
> this e-mail or any attachments to it and you must delete this message. You 
> are requested to notify the sender by return e-mail.
>
>
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> 



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Re: [asterisk-users] Wake-Up Call didn't work

[James FitzGibbon]

On 7/24/07, Asterisk guy <[EMAIL PROTECTED]> wrote:



 -- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold()
(Retry 1)

Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel

Jul 24 08:23:17 NOTICE[21177]: channel.c:2409 __ast_request_and_dial:
Unable to request channel Local/[EMAIL PROTECTED]


( but i have a extension 6009 login to * ) ,  what is the problem?



Regardless of what endpoints you may have registering to your *, your
dialplan does not allow that endpoint to be reached via extension 6009 in
the 'default' context.

Look at the file that gets put in outgoing (comment out the "rename" in the
AGI script so it stays in /tmp".  Then go read up on call files on the Wiki:

http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

Calls generated by call files need to have a starting point and a
destination.  The starting point for stuff like this is typically a "Local"
channel, and the destination is either a context/extension/priority or an
application with arguments.  Either your starting point or your destination
is invalid.

IMO, skip the AGI for now.  Get the file that your AGI is writting to /tmp
and make a copy of it.  Modify the copy, then move it to the outgoing dir
and see what happens.  If it doesn't work, make more changes.  Without
seeing your dialplan or the callfile, we can't diagnose your problem, but
the error messages are pretty informative as to what asterisk was trying to
do.  Once you've successfully generated a call manually, then go back to
having your AGI try to generate them automatically.

--
j.
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Re: [asterisk-users] Wake-Up Call didn't work

[dave cantera]

on the CLI>   type this command:

dialplan show [EMAIL PROTECTED]  
-and-  
dialplan show [EMAIL PROTECTED]

you should see a dialplan returned to you.   if not, which is what I 
expect, you have to include the section [where6009is] in [local] or 
[default]... i.e.

[local]
include => where6009is
...

[default]
include => where6009is
...

[where6009is]
exten 6009,1,wait(2)
exten 6009,n,NoOp("getting to 6009")
...


Asterisk guy wrote:
> 1  there is a correct file in  /var/spool/asterisk/outgoing
>  
> 2  i run  asterisk -r to monitor it  , it gives out the following 
> error
>  
>  -- Attempting call on Local/[EMAIL PROTECTED]  PROTECTED]> 
> for application MusicOnHold() (Retry 1)
>
> Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such 
> extension/context [EMAIL PROTECTED]  creating local 
> channel
>
> Jul 24 08:23:17 NOTICE[21177]: channel.c:2409 __ast_request_and_dial: 
> Unable to request channel Local/[EMAIL PROTECTED]  PROTECTED]>
>  
>  
> ( but i have a extension 6009 login to * ) ,  what is the problem?
>
>
>  
> On 7/23/07, *James FitzGibbon* <[EMAIL PROTECTED] 
> > wrote:
>
> On 7/23/07, *Dovid B* < [EMAIL PROTECTED]
> > wrote:
>  
>
> Can it be that asterisk does not have permission to copy the
> file over ?  Also check your date settings on the server.
>
>
>
> Yes, it's interesting that the page intro includes the sentence
> "Lots of error checking to make sure its done correctly", but the
> final step that makes the process work (ensuring that the callfile
> ends up in the directory that pbx_spool is watching) doesn't have
> any error checking:
>
> touch( $wakefile, $time_wakeup, $time_wakeup );
>
> rename( $wakefile, $callfile );
>
> The fact that you see files in /tmp when all is said and done
> means that at least some of the script is working.  A few things
> to check:
>
> Do the files in /tmp have the correct timestamp (file matches the
> requested wakeup time)?  If so, then everything preceeding the
> rename seems to have worked, so check if the user running the AGI
> can move files from /tmp to /var/spool/asterisk/outgoing.  Though
> given that it's an AGI being run by *, you'd have to have a pretty
> strange setup for that to fail.  Perhaps the outgoing directory
> just doesn't exist (was never created for some reason?)
>
> If the files don't have the correct timestamp, start following the
> logic backwards.  Do they look complete?  Look through the AGI for
> places where the wakeup file is written to (i.e.
> fputs( $wuc, "maxretries: $parm_maxretries\n"); ) and check that
> everything that should be written is being written
>
> Working backwards you should be able to figure out where the
> script is failing, then you can check everything that comes
> afterwards as the user running the AGI to make sure that
> permissions and directories are set up properly.
>
> -- 
> j.
> ___
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>
> 
>
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> 
>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.10.12/910 - Release Date: 07/21/2007 
> 03:52 PM
>   

-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894




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Re: [asterisk-users] Dialplan

[Matt]

I understand how to make a dialplan.. that wasn't my question.   I
also understand how to make the system add or remove digits from what
the user dialed.

My question is.. if I set it to 'national' doesn't that force the call
to go out with NPA-NXX-?  My understanding is if I set it to
national, and only pass NXX- as say 555-1212 it might end up as
555-1212- so my concern is about the 1.   If I set national
and do 1-555-555-1212 will that go, or will it rip the 1 off?

On 7/23/07, Dave Bour <[EMAIL PROTECTED]> wrote:
> Goto http://www.localcallingguide.com/lca_prefix.php and find the local
> calling area to your number you're based out of.  Then you can build the
> respective lookup for the local and leave the balance to long distance.
> Further to that, you can then control regardless of whether users dial 1
> or not for long distance.
> D.
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: Monday, July 23, 2007 2:33 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Dialplan
> >
> > Hi,
> > What dialplan option do I need to send a call out like this:
> >
> > NPA-NXX- local calls
> > 1-NPA-NXX- - long distance
> >
> > Won't 'national' send it out NPA-NXX- no matter if it's
> > long distance or not?
> >
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> >
>
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Re: [asterisk-users] Astribank-8BRI

[Jon Schøpzinsky]

Hello

We use the 2BRI version of Astribank in production, and it has been working non 
stop for about amonth now, without any problems.
It was a bit difficult to setup, but other than that, it was great.
Great concept with using the USB2 port for channel banks.

Regards
Jon

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lars Bensmann
Sent: 24. juli 2007 04:22
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Astribank-8BRI

Hello,

I'm in the process of building an Asterisk machine and need 5 or 6
BRI-Channels. I was looking for the beroNet and Junghans cards and
stumbled upon the Xorcom Astribank xBRI products.

Has anybody tried out the Astribank xBRI-Channel Banks? Are they
production ready or should I go with a beroNet BN8S0 or JUNGHANNS.NET
octoBRI ISDN?

Thanks in advance,
Lars

-- 
We don't like their sound.  Groups of guitars are on the way out.
  -- Decca Recording Company, turning down the Beatles, 1962

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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.10.16/914 - Release Date: 23-07-2007 
19:45
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.10.16/914 - Release Date: 23-07-2007 
19:45
 

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Re: [asterisk-users] Wake-Up Call didn't work

[Asterisk guy]

1  there is a correct file in  /var/spool/asterisk/outgoing

2  i run  asterisk -r to monitor it  , it gives out the following error

-- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold()
(Retry 1)

Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel

Jul 24 08:23:17 NOTICE[21177]: channel.c:2409 __ast_request_and_dial: Unable
to request channel Local/[EMAIL PROTECTED]


( but i have a extension 6009 login to * ) ,  what is the problem?



On 7/23/07, James FitzGibbon <[EMAIL PROTECTED]> wrote:


On 7/23/07, Dovid B <[EMAIL PROTECTED]> wrote:

 Can it be that asterisk does not have permission to copy the file over ?
> Also check your date settings on the server.
>


Yes, it's interesting that the page intro includes the sentence "Lots of
error checking to make sure its done correctly", but the final step that
makes the process work (ensuring that the callfile ends up in the directory
that pbx_spool is watching) doesn't have any error checking:

touch( $wakefile, $time_wakeup, $time_wakeup );

rename( $wakefile, $callfile );

The fact that you see files in /tmp when all is said and done means that
at least some of the script is working.  A few things to check:

Do the files in /tmp have the correct timestamp (file matches the
requested wakeup time)?  If so, then everything preceeding the rename seems
to have worked, so check if the user running the AGI can move files from
/tmp to /var/spool/asterisk/outgoing.  Though given that it's an AGI being
run by *, you'd have to have a pretty strange setup for that to fail.
Perhaps the outgoing directory just doesn't exist (was never created for
some reason?)

If the files don't have the correct timestamp, start following the logic
backwards.  Do they look complete?  Look through the AGI for places where
the wakeup file is written to (i.e. fputs( $wuc, "maxretries: 
$parm_maxretries\n");
) and check that everything that should be written is being written

Working backwards you should be able to figure out where the script is
failing, then you can check everything that comes afterwards as the user
running the AGI to make sure that permissions and directories are set up
properly.

--
j.
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Re: [asterisk-users] Dial out through multiple Zap groups

[Noah Miller]

Hi Vieri -

> I'm trying to set a rule to dial out through multiple
> Zap groups so that, say, g0 is the cheaper POTS lines
> group
> and must be used first. However, if g0 is busy or
> disconnected then try dialing out g1.
>
> My g0 group is made up of 4 analog lines connected to
> a 4-FXO card. I disconnected the RJ-11 wires from the
> FXO card
> to simulate a line disconnection. So theoretically all
> calls should immediately go out through g1 but they
> don't.
> They get "stuck" on g0 as I can see in the asterisk

You've discovered a big limitation in analog lines.  If this were a
PRI or BRI, the lines would behave as you want, but analog lines
won't.

You can try using ChanIsAvail() to test beforehand if the zap channels
will accept a call.  I don't know if it will work in this fashion
(I've never tried).


- Noah

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[asterisk-users] MySQL components in asterisk-addons not being built

[hugolivude]

I'm trying to add MySQL CDR recording in Asterisk 1.4.6.  I'm
following the instructions posted here:

http://www.voip-info.org/wiki-Asterisk+cdr+mysql

I have MySQL installed and it works fine - starts on stratup, I can
create DBs, tables and so on and I can connect through php.  rpm -qa
indicates:

MySQL-server-5.0.22-0
MySQL-devel-5.0.22-0
MySQL-client-5.0.22-0

However I still get XXX for all of the MySQL add ons when I do:

make menuselect

Any pointers for me on how to troubleshoot and fix this problem?

Thanks,
Hugh

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Re: [asterisk-users] TDM04B & FIOS No Hangups Often

[Noah Miller]

Hi Mike -

> It seems like since we got FIOS
> installed (including switching to fios phone lines which are supposed to be
> the same on our end) i am having massive problems with asterisk not hanging
> up dead calls for days, even weeks if i dont catch it.  It slowly builds up
> randomly not ending a call and then next thing i know all our lines are busy
> and they all say a call is active in show channels.  i have to shutdown
> asterisk and then restart and then it goes back to normal.

Some things to try:
1) Set the FIOS/zaptel lines to kewlstart signalling (if you haven't
already done that).
2) Set callprogress=yes in zapata.conf (if you haven't already done that).
3) Do you have explicitly declared hangup() statements in your
dialplan?  If not, you could try adding a few in key places as a test.
4) Verify with Verizon that you have disconnect supervision on your
lines.  I would think that they should be able to provide disconnect
supervision on these high tech lines, but maybe the demuxer you
mentioned screws this up.


- Noah

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Re: [asterisk-users] POE injector

[Noah Miller]

> IEEE802.3af uses same 4 wire as data.
> thats what Polycom uses.
> the way i'm seeing it we are better off with poe switch(looking at the
> price).

802.3af has two different modes:

Mode A: uses the same 4 wires as 10/100 ethernet, typically done by
PoE endpoints like switches
Mode B: uses the unused pairs on 10/100 ethernet, typically used by
midspan injectors


- Noah

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Re: [asterisk-users] bristuff for hfc card on Xscale 80219

[Tzafrir Cohen]

On Tue, Jul 24, 2007 at 12:05:47PM +0200, Thomas Winter wrote:
> On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote:
> > On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote:
> > > Hi,
> 
> > Frozen or crashed? Do you see the console of the system?
> 
> serial console is dead.
> 
> kernel is 2.6.18-4 debian Etch.
> bristuff is latest zaptel-1.2.19 and asterisk-1.2.22
> I tried older bristuff before but same result.
> 
> 
> > Can you load zaptel and zaphfc those modules with debug=1
> 
> yes, I assume the board is frozen because of to much load on the pci bus. if 
> making tail of the syslog I can see tons of lines before board is frozen.
> I have seen similar messages before if using two HFC cards on old PC, so I 
> assume not enough horse power for cheap HFC card and bristuff.

It might also be a spinning spinlock.

Any luck with alt-sysrq commands?

-- 
   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] POE injector

[Paul Hayes]

Noah Miller wrote:
>> I'm looking for 24 or 48 port IEEE802.3af POE injector.
>> Any recommendation?
> 
> Yes.  For the price of one of those multi-port injectors, you can come
> close to the price of a new Netgear or 3Com PoE switch.  The injectors
> typically add power to the unused pairs (mode B PoE).  This means you
> can't use them on anything better than fastethernet.  When switches do
> PoE natively, they put the power on the data carrying pairs (mode A
> PoE), so they can do gigabit ethernet.  I think PowerDsine makes a PoE
> injector that uses mode A, and so it can do gigabit ethernet.
> 
> 
> - Noah
> 
The midspans that Phihong make are very good.  They support gigabit pass 
through and have very good overload protection.

cheers,
Paul.

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Re: [asterisk-users] bristuff for hfc card on Xscale 80219

[Thomas Winter]

On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote:
> On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote:
> > Hi,

> Frozen or crashed? Do you see the console of the system?

serial console is dead.

kernel is 2.6.18-4 debian Etch.
bristuff is latest zaptel-1.2.19 and asterisk-1.2.22
I tried older bristuff before but same result.


> Can you load zaptel and zaphfc those modules with debug=1

yes, I assume the board is frozen because of to much load on the pci bus. if 
making tail of the syslog I can see tons of lines before board is frozen.
I have seen similar messages before if using two HFC cards on old PC, so I 
assume not enough horse power for cheap HFC card and bristuff.


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Re: [asterisk-users] phone directory with asterisk

[Tim Koehler]

Hello,


again, as discussed in the other Thread, these are SI-120. The SI-120 is NOT
a snom phone, it is a cheap phone sold by our subsidary in India.

Neither the Software nor the Hardware was German engineered by snom
technology AG. Thus there is no Minibrowser in this phone.

To prevent further missunderstanding please do not refer the SI-120 as a
snom phone. If you need support please contact snom India.

Regards


Tim Koehler

On 7/24/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:


Am Montag, den 23.07.2007, 06:44 -0700 schrieb satish patel:
> Dear all
>
>I have configure asterisk with 100 SIP PHONE ( SNOM )
> but now thing is that my boss need phonebook feature find extention
> number by Pbook so i have read about it there is a feature in asterisk
> but it is with voicemail now i have IP SIP phone of SNOM so how to
> fine phone number by SIP phone ?? how to asterisk directory work ?

As far as I know the popular asterisk phonebook solution ("Directory")
works by calling an extension and punching in the first letters of the
name (calling me, punching 463... for Hoffmeister, for example) and
makes use of the information in voicemail.conf.

Some SNOM phones have a micro browser, it seems you can use it
for phonebook display. Read (way down)
http://www.voip-info.org/wiki/view/Asterisk+phone+snom
and perhaps the manufacturer homepage for details.

Not been there, not done that ;-)

BR
Anselm



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--
---
snom technology AG

Tim Koehler
Partner Manager
[EMAIL PROTECTED]
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[asterisk-users] Diva Server BRI hangs up after about 25 seconds

[kjcsb]

I have a Diva Server BRI installed in a Debian system
running Asterisk 1.2.22. When a call comes in the
Background application plays a couple of times.
Halfway through the second time, the call is
disconnected (total time connected about 25 seconds).
I've posted various configs below. Any advice on how
to debug would be appreciated. We are located in New
Zealand.

CLI showing capi debug (just before hangup)
-- EICONISDN#02: DATA_B3_IND (len=160)
fr.datalen=160 fr.subclass=8
DATA_B3_REQ ID=001 #0x04e1 LEN=0030
  Controller/PLCI/NCCI= 0x10201
  Data32  = 0x816d494
  DataLength  = 0xa0
  DataHandle  = 0x4d9
  Flags   = 0x0
  Data64  = 0x0

DATA_B3_CONF ID=001 #0x04e1 LEN=0016
  Controller/PLCI/NCCI= 0x10201
  DataHandle  = 0x4d9
  Info= 0x0

DATA_B3_IND ID=001 #0x04ed LEN=0022
  Controller/PLCI/NCCI= 0x10201
  Data32  = 0x405b1362
  DataLength  = 0xa0
  DataHandle  = 0x163
  Flags   = 0x0
  Data64  = 0x8b8b8b8b8b8b8b8b

DATA_B3_RESP ID=001 #0x04ed LEN=0014
  Controller/PLCI/NCCI= 0x10201
  DataHandle  = 0x163

-- EICONISDN#02: DATA_B3_IND (len=160)
fr.datalen=160 fr.subclass=8
DATA_B3_REQ ID=001 #0x04e2 LEN=0030
  Controller/PLCI/NCCI= 0x10201
  Data32  = 0x816d574
  DataLength  = 0xa0
  DataHandle  = 0x4da
  Flags   = 0x0
  Data64  = 0x0

DATA_B3_CONF ID=001 #0x04e2 LEN=0016
  Controller/PLCI/NCCI= 0x10201
  DataHandle  = 0x4da
  Info= 0x0

INFO_IND ID=001 #0x04ee LEN=0017
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x8
  InfoElement = <82 90>

INFO_RESP ID=001 #0x04ee LEN=0012
  Controller/PLCI/NCCI= 0x201

-- EICONISDN#02: info element CAUSE 82 90
INFO_IND ID=001 #0x04ef LEN=0015
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x8045
  InfoElement = default

INFO_RESP ID=001 #0x04ef LEN=0012
  Controller/PLCI/NCCI= 0x201

-- EICONISDN#02: info element DISCONNECT
-- EICONISDN#02: Disconnect case 3
-- CAPI queue frame:[ TYPE: Control (4) SUBCLASS:
Hangup (1) ] [EICONISDN#02]
  == Spawn extension (incoming, 09375, 3) exited
non-zero on 'CAPI/EICONISDN/09375-0'
  == EICONISDN#02: CAPI Hangingup for PLCI=0x201 in
state 2
-- EICONISDN#02: activehangingup (cause=16) for
PLCI=0x201
DISCONNECT_B3_REQ ID=001 #0x04e3 LEN=0013
  Controller/PLCI/NCCI= 0x10201
  NCPI= default

DISCONNECT_B3_CONF ID=001 #0x04e3 LEN=0014
  Controller/PLCI/NCCI= 0x10201
  Info= 0x0

DISCONNECT_B3_IND ID=001 #0x04f1 LEN=0015
  Controller/PLCI/NCCI= 0x10201
  Reason_B3   = 0x0
  NCPI= default

DISCONNECT_B3_RESP ID=001 #0x04f1 LEN=0012
  Controller/PLCI/NCCI= 0x10201

DISCONNECT_REQ ID=001 #0x04e4 LEN=0018
  Controller/PLCI/NCCI= 0x201
  AdditionalInfo 
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

   > CAPI devicestate requested for
EICONISDN/09375
   > CAPI devicestate requested for
EICONISDN/09375
DISCONNECT_CONF ID=001 #0x04e4 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Info= 0x0

INFO_IND ID=001 #0x04f2 LEN=0015
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x805a
  InfoElement = default

INFO_RESP ID=001 #0x04f2 LEN=0012
  Controller/PLCI/NCCI= 0x201

-- EICONISDN#02: info element RELEASE COMPLETE
DISCONNECT_IND ID=001 #0x04f4 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x3490

DISCONNECT_RESP ID=001 #0x04f4 LEN=0012
  Controller/PLCI/NCCI= 0x201

   > EICONISDN#02: CAPI INFO 0x3490: Normal call
clearing
  == EICONISDN#02: Interface cleanup PLCI=0x201

/usr/lib/divas/Config 
Interface mode: TE (verified)
D Channel: ETSI-DSS1 (verified)
NT-2 mode: No
D-Channel Layer activation: Deactivation by other side
Voice companding: National default
Hunt group operation: Standard
Trunk Operation mode: point to point (fixed TEI)
(verified)
TEI value: 0 (verified)
Source of tones: Provided by ISDN
CAPI call distribution: Off
Max fax speed: No limit
Min fax speed: No limit
Fax session limit: 0
T.30 protocol options: None selected
Part 68 voice signal 

[asterisk-users] Dial out through multiple Zap groups

[Vieri]

Hi,

I'm trying to set a rule to dial out through multiple
Zap groups so that, say, g0 is the cheaper POTS lines
group 
and must be used first. However, if g0 is busy or
disconnected then try dialing out g1.

My g0 group is made up of 4 analog lines connected to
a 4-FXO card. I disconnected the RJ-11 wires from the
FXO card 
to simulate a line disconnection. So theoretically all
calls should immediately go out through g1 but they
don't. 
They get "stuck" on g0 as I can see in the asterisk
CLI:

-- Executing Dial("SIP/4053-082393a8",
"ZAP/g0/5|120|TWm") in new stack
-- Called g0/5
-- Started music on hold, class 'default', on
SIP/4053-082393a8
-- Zap/32-1 answered SIP/4053-082393a8
-- Stopped music on hold on SIP/4053-082393a8
(endless)

Note: Zap channel 32 is part of g0.

I used both FreePBX and a custom made rule.
With FreePBX, the outgoing dialplan includes something
like this:

exten =>
_5,1,Macro(dialout-trunk,1,${EXTEN},,)
exten =>
_5,n,Macro(dialout-trunk,2,${EXTEN},,)
exten => _5,n,Macro(outisbusy,)
; trunk 1 is g0, trunk 2 is g1

If I use a custom dialpan that looks something like
this:

exten => _5,1,Dial(Zap/g0/${EXTEN})
exten => _5,n,NoOp(${DIALSTATUS})
exten => _5,n,Dial(Zap/g1/${EXTEN})
exten => _5,n,HangUp()

and then watch the CLI, I get exactly the same
behavior as above, ie. I don't get past
Dial(Zap/g0/${EXTEN}) as
Zap/32 answers when it shouldn't. And obviously I
can't get ${DIALSTATUS} to eventually define some
gotos because it's ANSWERED.

Any ideas as to what I should try?
Maybe change some setting in zapata.conf?

Thanks

Vieri



  

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Re: [asterisk-users] phone directory with asterisk

[Anselm Martin Hoffmeister]

Am Montag, den 23.07.2007, 06:44 -0700 schrieb satish patel:
> Dear all
> 
>I have configure asterisk with 100 SIP PHONE ( SNOM )
> but now thing is that my boss need phonebook feature find extention
> number by Pbook so i have read about it there is a feature in asterisk
> but it is with voicemail now i have IP SIP phone of SNOM so how to
> fine phone number by SIP phone ?? how to asterisk directory work ?

As far as I know the popular asterisk phonebook solution ("Directory")
works by calling an extension and punching in the first letters of the
name (calling me, punching 463... for Hoffmeister, for example) and
makes use of the information in voicemail.conf.

Some SNOM phones have a micro browser, it seems you can use it
for phonebook display. Read (way down)
http://www.voip-info.org/wiki/view/Asterisk+phone+snom
and perhaps the manufacturer homepage for details.

Not been there, not done that ;-)

BR
Anselm



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Re: [asterisk-users] Problem Hangup

[Knud Müller]

FERNANDO VILLARROEL schrieb:
> --- Knud Müller <[EMAIL PROTECTED]> wrote:
>
>   
>> FERNANDO VILLARROEL schrieb:
>> 
>>> Hello list, i need help.
>>>
>>> My problem is that when I want to call (using ISDN
>>> phone or internal SIP client) via the Sip provider
>>>   
>> a
>> 
>>> real phone number (ISDN phone or internal SIP
>>>
>>> Asterisk >> SIP ), I get a ring tone. When
>>>   
>> I
>> 
>>> now decide to hang up (e.g. if 
>>>
>>> nobody answers), the called telephone continues to
>>> ring almost forever.
>>>
>>> the sip.conf:
>>>
>>> [2563105]
>>> accountcode = 2563105
>>> username = 2563105
>>> secret = 135
>>> callerid = 412563105
>>> context = test
>>> canreinvite = no
>>> dtmfmode = rfc2833
>>> host = dynamic
>>> insecure = very
>>> language = es
>>> nat = yes
>>> qualify = yes
>>> type = friend
>>> disallow=all
>>> allow=g729
>>>
>>> [nyphone]
>>> accountcode=nyphone
>>> canreinvite=no
>>> reinvite=yes
>>> username=test770
>>> secret=test770
>>> dtmfmode=rfc2833
>>> host=72.55.143.XXX
>>> insecure=very
>>> language=es
>>> nat=no
>>> qualify=no
>>> type=peer
>>> disallow=all
>>> allow=g729
>>>
>>> I attach sip debug one call.
>>>
>>> I use Asterisk 1.2.13
>>>
>>> I hope you understand me and help.
>>>
>>> Best regards
>>>
>>> Fernando Villarroel Noriel.
>>> Chillan
>>> Chile
>>>
>>> Sorry my English.
>>>
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>>
>>>
>>>
>>>
>>>
>>>   
> 
>   
>>> Looking for a deal? Find great prices on flights
>>>   
>> and hotels with Yahoo! FareChase.
>> 
>>> http://farechase.yahoo.com/
>>>
>>>   
> 
>   
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>>>   
>>>   
> http://lists.digium.com/mailman/listinfo/asterisk-users
>   
>> If I got it right: you register to your SIP Provider
>> which provides a 
>> PSTN Number to you. You dial the PSTN Number which
>> is forwarded to your 
>> asterisk. Your asterisk dials the SIP phone
>> (nyphone)?
>> 
>
> Yes nyphone is my provider for everyone calls
> internationational (prefix 00)
>
> 2563105 is one number provided for my Telco (E1) and
> is one SIP client.
>   
So its an outbound call not an inbound call!
>   
>> Could you attach your dialplan?
>> 
>
> exten => _00X.,1,dial(sip/${EXTEN:[EMAIL PROTECTED],45)
> exten => _00X.,2,hangup
>   
This looks OK. I'd recommend to record the SIP communication with your 
provider. Do on the CLI: "SIP debug". You should see after latest after 
45 seconds that asterisk sends an hangup request to your provider. If * 
sends the request it must be your provider. If it is not sent, something 
is wrong with your *.
>
> the called telephone continues to ring almost forever.
>
>   
>> Knud
>>
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>
>
>
>
> 
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-- 
Knud A. Müller
Geschäftsführer
Tel.: 040/398053-11
Fax: 040/398053-29
e-Mail: [EMAIL PROTECTED]

portrix.net GmbH
Stresemannstr. 375
22761 Hamburg
HRB 79850 (Amtsgericht Hamburg)
Geschäftsführer: Knud Alex Müller, Henning Voss, Niclas Schroeder

http://www.portrix.net 


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