date:20051024

Ronald Wiplinger wrote:
 I was looking for the text in the /etc/asterisk directory, but it must
 be somewhere else. Can anybody tell me where? And can it include Chinese
 as well?
Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
/configs directory of your source code tree.

I have never tried with Chinese, but it can handle Swedish :-)

/Olle
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Re: [Asterisk-Users] Where is the text of the voicemail email ??

On Mon, 2005-10-24 at 16:03 +0800, Ronald Wiplinger wrote:
 I was looking for the text in the /etc/asterisk directory, but it must 
 be somewhere else. Can anybody tell me where? And can it include Chinese 
 as well?
voicemail.conf?

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] Where is the text of the voicemail email ??

2005-10-24 Thread Guido Hecken

 I was looking for the text in the /etc/asterisk directory, but it must
 be somewhere else. Can anybody tell me where? And can it include Chinese
 as well?

Isn't it in /etc/asterisk/voicemail.conf ?
In our installations we change the voicemail text in this file.
Maybe you could include another file in this file, so different charsets
could be possible.

Hope it helps a bit...

Regards 
Guido Hecken
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-24 Thread stoffell

On 10/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Yes I did notice it immediately.I intend to tweak more, but for themoment it seems like echo is minimized to zero.
I also encountered some echo problems and used (uncommented :)) following parameters in zconfig.h:
#define ECHO_CAN_MARK3 (instead of MARK2)
#define CONFIG_CALC_XLAW
#define CONFIG_ZAPTEL_MMX

Up untill now it seems to be much better.. It also 'sounds' much better during normal conversation.

Oh, and in the Makefile, changed some flags:
KFLAGS+=-DSTANDALONE_ZAPATA -march=pentium4 -O3 -fomit-frame-pointer
CFLAGS+=-DSTANDALONE_ZAPATA -march=pentium4 -O3 -fomit-frame-pointer

cheers.


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Re: [Asterisk-Users] Where is the text of the voicemail email ??

2005-10-24 Thread Lenz



Hello,
you should look at voicemail.conf, see emailsubject and emailbody. I  
believe that it can handle Chinese as any other language as well, as you  
can specify the charset encoding.

Bye
l.


On Mon, 24 Oct 2005 10:03:22 +0200, Ronald Wiplinger [EMAIL PROTECTED]  
wrote:


I was looking for the text in the /etc/asterisk directory, but it must  
be somewhere else. Can anybody tell me where? And can it include Chinese  
as well?



bye

Ronald Wiplinger





--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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RE: [Asterisk-Users] Where is the text of the voicemail email ??

2005-10-24 Thread Guido Hecken



  I was looking for the text in the /etc/asterisk directory, but it must
  be somewhere else. Can anybody tell me where? And can it include Chinese
  as well?
 Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
 /configs directory of your source code tree.
 
 I have never tried with Chinese, but it can handle Swedish :-)

BTW wouldn't it be helpfull if the voicemailtext could depend on the
language, the user has choosen in extensions.conf? 
Example:
User has language set to de, include language file de in voicemail.conf .

Regards

Guido Hecken


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Re: [Asterisk-Users] asterisk -RT

2005-10-24 Thread Ronald Wiplinger

Tzafrir Cohen wrote:

On Mon, Oct 24, 2005 at 12:04:40PM +0800, Ronald Wiplinger wrote:

I use the command asterisk -RT to connect to a running asterisk box.

There must be some changes to the latest CVS upgrade:

1. it does not remember anything anymore what I have done in the
previous connection.

Why do you use 'asterisk -R' and not 'asterisk -r'?

I found that in man asterisk:
-r Instead of running a new Asterisk process, attempt to
connect to a running Asterisk process and provide a console interface

for controlling it.

-R Much like -r. Instead of running a new Asterisk
process, attempt to connect to a running Asterisk process and provide a
console interface for controlling it. Additionally, if
connection to the Asterisk process is lost, attempt to reconnect for

as long as 30 seconds.

-T Add timestamp to all non-command related output going to
the console when running with verbose and/or logging to the conâ

sole.

bye

Ronald Wiplinger

When you exit a CLI shell with ctrl-C it does not save the history. Is
this a bug?

Try quiting with 'quit', though I'm not sure if it has a different
effect on asterisk -R .

I could reconnect to the asterisk box and with
arrow up I could see all my last commands, now no more.

2. I still cannot see any colors,

The original asterisk starts via
31240 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk
31245 ?Sl 0:00 asterisk -vvvgpT -c

Asterisk only checks for a color terminal at startup. Asterisk does not
color messages in the remote CLI, but rather, when issuing the messages.

--
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Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208

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[Asterisk-Users] Asterisk Realtime - MySQL Extension registration problem

2005-10-24 Thread tijmen van den brink

Hi all,

I'm trying to register an extension but when I do so I get the following mesage:

Sep 24 06:22:46 WARNING[18152]: res_config_mysql.c:135 realtime_mysql:
MySQL RealTime: Failed to query database. Check debug for more info.
Sep 24 06:22:46 NOTICE[18152]: chan_sip.c:10774
handle_request_register: Registration from '1001
sip:[EMAIL PROTECTED]' failed for '10.8.5.51' - Username/auth
name mismatch

I checked wether asterisk was connected to the MySQL database and it appears to be connected.

asterisk1*CLI realtime mysql status 
Connected to [EMAIL PROTECTED], port 3306 with username **blahblah** for 2 days, 12 hours, 33 minutes, 42 seconds.

My database looks like this

sip-peers






Field
Type

Null
Default


Links to
Comments
MIME





id

int(11)

No










name

varchar(80)

No










type

varchar(6)

No
friend









username

varchar(80)

No










secret

varchar(80)

No










qualify

char(3)

No










callerid

varchar(80)

No










host

varchar(31)

No










dtmfmode

varchar(4)

No
info









ipaddr

varchar(15)

No

















Indexes:





Keyname
Type
Cardinality
Field




PRIMARY


PRIMARY


2


id


















Space usage:


Type
Usage


Data
112
Bytes



Index
2,048
Bytes



Total
2,160
Bytes









Row Statistics:


Statements
Value



Format

dynamic




Rows

2




Row lengthø

56




 Row size ø

1,080 Bytes




NextAutoindex

3





Creation

Sep 21, 2005 at 05:58 PM





Last update

Sep 21, 2005 at 05:58 PM


I filled my database with 2 extension like this:


SQL result


Host: localhost
Database: asterisk
Generation Time: Sep 24, 2005 at 06:27 AM
Generated by: phpMyAdmin2.6.4/ MySQL4.1.14-log
SQL query: SELECT * FROM `sip-peers`  LIMIT 0, 30 ;

Rows: 2















id



name



type



username



secret



qualify



callerid



host



dtmfmode



ipaddr








1

friend
1006
1006
yes
Extensie 1006
dynamic
info






2

friend
1001
1001
yes
Extensie 1001
dynamic
info


I think I did something wrong making the database or
filling it, might just be some missing field or something. But somehow
I can't figure out what's going wrong.

Any ideas?

Thanks in advance 

Tijmen
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[Asterisk-Users] NAT Problem after first call

2005-10-24 Thread René Enskat [Teamware GmbH]









Hey all,



I have little problem with my NAT clients on
asterisk.

After I called the clients one time where all is fine
I try to call again and then the asterisk only say CALLED clientid I have
to reset the phone and reregister the phone so I can call again the phone.

Somebody can help me?



Regards rene 









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[Asterisk-Users] configuring Cisco 7905G for SIP - how?

After reading the specifications of Cisco 7905G phone (supports SIP, 
easy to manage etc.), we were so foolish and bought it.
Now we learned the hard way that we have to pay additionally for SIP 
firmware.


So two months after purchase, after much struggle with Cisco 
the-so-called support we have a shiny Cisco 7905G phone, support 
contract, and a newly downloaded SIP firmware.


Unfortunately, the instructions attached to the SIP firmware seem to be 
for a different phone, as they state that the 7905G phone should 
download lddefault.cfg config file (which took some time to configure, 
as it's 50 kilo big). In our case, the 7905G phone tries to download 
SEP0014690620AA.cnf.xml, and XMLDefault.cnf.xml.



Does anyone have a good, step-by-step SIP upgrade instruction for this 
phone?



--
Tomek
http://wpkg.org
WPKG - software deployment and upgrades with Samba
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[Asterisk-Users] AGI Dial hangup with *

2005-10-24 Thread Alexey Goloshubin


Hi!

I have an AGI script (ala ASTCC) that is executed via DeadAGI.

The script does exec Dial. The caller is allowed to press '*' to hangup 
the call.


I want to distinguish between caller pressing *, callee hanging up and 
caller hanging up. In the first two situations the script should 
continue telling the caller that the call has ended and so on. In the 
third case the script is terminated.


The return value from exec Dial is -1 on any of the three hangups above. 
DIALSTATUS does not help much either.


Any pointers are very welcome!

Sincerely,
Alexey




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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?

2005-10-24 Thread tijmen van den brink

I set up a Cisco 7960 in about 20 minutes with this document. I hope it works for you.

http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html

On 10/24/05, Tomasz Chmielewski [EMAIL PROTECTED] wrote:
After reading the specifications of Cisco 7905G phone (supports SIP,easy to manage etc.), we were so foolish and bought it.Now we learned the hard way that we have to pay additionally for SIPfirmware.
So two months after purchase, after much struggle with Ciscothe-so-called support we have a shiny Cisco 7905G phone, supportcontract, and a newly downloaded SIP firmware.Unfortunately, the instructions attached to the SIP firmware seem to be
for a different phone, as they state that the 7905G phone shoulddownload lddefault.cfg config file (which took some time to configure,as it's 50 kilo big). In our case, the 7905G phone tries to download
SEP0014690620AA.cnf.xml, and XMLDefault.cnf.xml.Does anyone have a good, step-by-step SIP upgrade instruction for thisphone?--Tomekhttp://wpkg.orgWPKG - software deployment and upgrades with Samba
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: 
[EMAIL PROTECTED]Skype: [EMAIL PROTECTED]SIP:[EMAIL PROTECTED]
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[Asterisk-users] Asterisk Realtime - MySQL Extension registration problem

2005-10-24 Thread tijmen van den brink

Hi all,

I'm trying to register an extension but when I do so I get the following mesage:

Sep 24 06:22:46 WARNING[18152]: res_config_mysql.c:135 realtime_mysql:
MySQL RealTime: Failed to query database. Check debug for more info.
Sep 24 06:22:46 NOTICE[18152]: chan_sip.c:10774
handle_request_register: Registration from '1001
sip:[EMAIL PROTECTED]' failed for '
10.8.5.51' - Username/auth
name mismatch

I checked wether asterisk was connected to the MySQL database and it appears to be connected.

asterisk1*CLI realtime mysql status 
Connected to [EMAIL PROTECTED], port 3306 with username **blahblah** for 2 days, 12 hours, 33 minutes, 42 seconds.

My database looks like this

sip-peers






Field
Type

Null
Default


Links to
Comments
MIME





id

int(11)

No










name

varchar(80)

No










type

varchar(6)

No
friend









username

varchar(80)

No










secret

varchar(80)

No










qualify

char(3)

No










callerid

varchar(80)

No










host

varchar(31)

No










dtmfmode

varchar(4)

No
info









ipaddr

varchar(15)

No

















Indexes:





Keyname
Type
Cardinality
Field




PRIMARY


PRIMARY


2


id


















Space usage:


Type
Usage


Data
112
Bytes



Index
2,048
Bytes



Total
2,160
Bytes









Row Statistics:


Statements
Value



Format

dynamic




Rows

2




Row length�

56




 Row size �

1,080 Bytes




NextAutoindex

3





Creation

Sep 21, 2005 at 05:58 PM





Last update

Sep 21, 2005 at 05:58 PM


I filled my database with 2 extension like this:


SQL result


Host: localhost
Database: asterisk
Generation Time: Sep 24, 2005 at 06:27 AM
Generated by: phpMyAdmin2.6.4/ MySQL4.1.14-log
SQL query: SELECT * FROM `sip-peers`  LIMIT 0, 30 ;

Rows: 2















id



name



type



username



secret



qualify



callerid



host



dtmfmode



ipaddr








1

friend
1006
1006
yes
Extensie 1006
dynamic
info






2

friend
1001
1001
yes
Extensie 1001
dynamic
info


I think I did something wrong making the database or
filling it, might just be some missing field or something. But somehow
I can't figure out what's going wrong.

Any ideas?

Thanks in advance 

Tijmen

-- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: 
[EMAIL PROTECTED]SIP:[EMAIL PROTECTED]
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[Asterisk-Users] 0.2.0-RC8o (* 1.0.9) + No Caller ID

2005-10-24 Thread Giovanni Miano

I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o
and * 1.0.9

I dont recive callerid from TDM400P fxo port but isdn hasnt problems

If i try to use only  TDM400P 1fxs/1fxo without bristuff.. all work ok

is it bug of bristuff ?

--
Giovanni Miano
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RE: [Asterisk-Users] Where is the text of the voicemail email ??

On Mon, 2005-10-24 at 10:32 +0200, Guido Hecken wrote:
 
   I was looking for the text in the /etc/asterisk directory, but it must
   be somewhere else. Can anybody tell me where? And can it include Chinese
   as well?
  Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
  /configs directory of your source code tree.
  
  I have never tried with Chinese, but it can handle Swedish :-)
 
 BTW wouldn't it be helpfull if the voicemailtext could depend on the
 language, the user has choosen in extensions.conf? 
 Example:
 User has language set to de, include language file de in voicemail.conf .

Yes and no.  Yes I think having language independant voicemail emails
would be a good thing.  No I dont think it should be related to
extensions.conf.  It should be on a per person basis, regardless of who
called into the system.  The reason for this is that the caller isnt
reading the email and the person getting the email may request that it
be a specific language.

Something I havent tried is setting emailbody etc on a per context
basis.  The ability to create more customized emails may exist, although
you would have to have a context for the languages itself rather than
per user, so it does have that limitation.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Trying to clarify ideas about spands, libtiff FC4

2005-10-24 Thread Doug Lytle


Carlos Alperin wrote:


Same question that before:


Mandrake/Mandrive 2.6.13.4 (thanks for the info)

What version of Asterisk?
 


CVS head, dated 2005-10-03


What version of Spandsp?
 


.0.0.2pre21 (I now see that there is a pres21a updated October 20th)


What version of Libtiff?
 


libtiff3-3.6.1-4.4.101mdk


What version of Libtiff-devel?
 


libtiff3-devel-3.6.1-4.4.101mdk


And the million dollars question: Is the fax working? (Lets say more than
50% of the cases?)

 

I only have around 3 people using it, but one of them receive around 20 
faxes a day with minimal complaints.  Faxes are coming over a Pri and 
converted to PDF before being sent via email to the end user.


Doug


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[Asterisk-Users] zap show channel 1 says PRI signalling on my zaphfc BRI card

2005-10-24 Thread Lars Dybdahl

I have a zaphfc compatible BRI card, and I cannot receive any phone
calls, because calls don't see to be detected by my system. I'm
running Fedora Core 4, bristuff from Junghanns.net, and this is my
zaptel.conf:

[EMAIL PROTECTED] asterisk]# more /etc/zaptel.conf
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=nl
defaultzone=nl

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

This is my zapata.conf:

[EMAIL PROTECTED] asterisk]# cat zapata.conf
;
; Zapata telephony interface
;
; Configuration file

[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net

;pridialplan = dynamic
;prilocaldialplan = local
pridialplan = unknown
prilocaldialplan = unknown
nationalprefix = 0
internationalprefix = 00

echocancel=yes
echotraining = 100
echocancelwhenbridged=yes

immediate=yes
group = 1
context=isdnincoming
channel = 1-2

Here is the result of ztcfg -vv:

[EMAIL PROTECTED] asterisk]# ztcfg -vv

Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.

This is the list of zap* drivers loaded:

[EMAIL PROTECTED] asterisk]# lsmod |grep zap
zaphfc 17300  3
zaptel231940  11 zaphfc
crc_ccitt   2241  1 zaptel

I think the problem relates to this:

[EMAIL PROTECTED] asterisk]# asterisk -r
Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o currently running on
131 (pid = 17652)
Verbosity is at least 17
131*CLI zap show channel 1
Channel: 1
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: isdnincoming
Caller ID string:
Destroy: 0
InAlarm: 1
Signalling Type: PRI Signalling
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
PRI Flags:
PRI Logical Span: Implicit
Actual Hookstate: Onhook

Any ideas, anyone?

Lars Dybdahl.
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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


tijmen van den brink schrieb:

 I set up a Cisco 7960 in about 20 minutes with this document. I hope 
it works for you.


 
http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html



I too managed to set up a 7960 phone.

But I have (had?) a problem with 7905 phone (still minor problems with 
that, like a wrong timezone).


BTW, I managed to solve it - the contents of the SEP0014690620AA.cnf.xml 
file have to be like this (with the right asterisk box IP address), and 
then it downloads the other files:



Default
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName192.168.11.15/processNodeName
/callManager
/member
/members
/callManagerGroup


Too bad Cisco 7905 documentation doesn't even mention *.cnf.xml files, 
their contents, etc.
Too bad Cisco binaries attached to 7905 firmware complain option not 
recognized when parsing even default config files (you need to 
convert the text files to some other mysterious format)...



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Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

2005-10-24 Thread Lars Dybdahl

We switched the QuadBRI machine to CentOS, and that worked. Thank you
all very much :-)

Lars.

On 10/15/05, Harald Holzer [EMAIL PROTECTED] wrote:
 You can find RPMS with bristuff included at:
 http://www.laimbock.com/asterisk/

 the are compiled for centos.
 rebuilding the SRPMS under FC3 work without a problem.

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Re: [Asterisk-Users] Where is the text of the voicemail email ??

Guido Hecken wrote:
 
I was looking for the text in the /etc/asterisk directory, but it must
be somewhere else. Can anybody tell me where? And can it include Chinese
as well?

Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
/configs directory of your source code tree.

I have never tried with Chinese, but it can handle Swedish :-)
 
 
 BTW wouldn't it be helpfull if the voicemailtext could depend on the
 language, the user has choosen in extensions.conf? 
 Example:
 User has language set to de, include language file de in voicemail.conf .
Yes, that is a good idea. Any coders?

/O
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[Asterisk-Users] Re: zap show channel 1 says PRI signalling on my zaphfc BRI card

2005-10-24 Thread Lars Dybdahl

The problem is solved.

Even though /etc/asterisk/zapata.conf is located in the asterisk
directory, it seems that it is necessary to removed and reload the
kernel modules in order to make changes take effect - that wasn't
obvious to me.

Lars Dybdahl.

On 10/24/05, Lars Dybdahl [EMAIL PROTECTED] wrote:
 I have a zaphfc compatible BRI card, and I cannot receive any phone
 calls, because calls don't see to be detected by my system. I'm
 running Fedora Core 4, bristuff from Junghanns.net, and this is my
 zaptel.conf:
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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


Tomasz Chmielewski ha scritto:

But I have (had?) a problem with 7905 phone (still minor problems with 
that, like a wrong timezone).


You can easy change it with the phone web page.

Sergio
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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?

Add this tag to the SEP file listed below. The phone will upgrade the 
firmware with the one mentioned in the loadInformation tag


loadInformationfirmware_filename_without_extension/loadInformation

Sergio

Tomasz Chmielewski ha scritto:

BTW, I managed to solve it - the contents of the 
SEP0014690620AA.cnf.xml file have to be like this (with the right 
asterisk box IP address), and then it downloads the other files:



Default
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName192.168.11.15/processNodeName
/callManager
/member
/members
/callManagerGroup


Too bad Cisco 7905 documentation doesn't even mention *.cnf.xml files, 
their contents, etc.
Too bad Cisco binaries attached to 7905 firmware complain option not 
recognized when parsing even default config files (you need to 
convert the text files to some other mysterious format)...





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[Asterisk-Users] Asterisk vs Sipura SIP problem?

2005-10-24 Thread Frank Tarczynski

I am trying to use a SIP provider for outgoing and incoming calls under 
Asterisk.  I am running a recent CVS-head 1.09 build and the SIP 
provider is using a SPA-3000.  I can register with the SIP provider's 
server and outgoing calls seem to work just fine.


But I cannot get incoming calls to work at all.

I see absolutely no indication in the Asterisk SIP debug output that 
incoming SIP calls are coming from this provider!  But in output from 
ethereal I find that my Asterisk box responds to the initial INVITE with 
a 484 Address Incomplete.  There is no response from the SIP provider 
and a few seconds later my Asterisk sends an ACKnowledge.


Absolutely none of this shows-up in the Asterisk output!

The INVITE is addressed to sip:[EMAIL PROTECTED]:5060 and all my Asterisk 
extensions are 4 digits starting with 1s.  Shouldn't the SPA-3000 
respond back to the 484 again?  Or since it is using just 1 is no 
additional response sent?


I tried creating an extension context of [1] but this has no effect.  I 
just keep getting the 484 responses.


Do I need to ask the SIP provider to configure the SPA-3000 
differently?  Have the INVITE request changed?  How/where would I create 
a context that Asterisk can use/understand?


No. TimeSourceDestination   Protocol 
Info
  1016 20:48:20.196168 XXX-IPA.155.115.200.in-addr.arpa lyla.domian.com   
SIP/SDP  Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description

Frame 1016 (1084 bytes on wire, 1084 bytes captured)
   Arrival Time: Oct 23, 2005 20:48:20.196168000
   Time delta from previous packet: 0.00161 seconds
   Time since reference or first frame: 32.762675000 seconds
   Frame Number: 1016
   Packet Length: 1084 bytes
   Capture Length: 1084 bytes
Ethernet II, Src: 00:04:e2:bc:76:80, Dst: 00:0e:0c:62:cb:08
   Destination: 00:0e:0c:62:cb:08 (lyla.domain.com)
   Source: 00:04:e2:bc:76:80 (ipcop.domain.com)
   Type: IP (0x0800)
Internet Protocol, Src Addr: XXX-IPA.155.115.200.in-addr.arpa 
(200.115.155.XXX), Dst Addr: lyla.domain.com (192.168.0.4)
   Version: 4
   Header length: 20 bytes
   Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
    00.. = Differentiated Services Codepoint: Default (0x00)
    ..0. = ECN-Capable Transport (ECT): 0
    ...0 = ECN-CE: 0
   Total Length: 1070
   Identification: 0x (0)
   Flags: 0x04 (Don't Fragment)
   0... = Reserved bit: Not set
   .1.. = Don't fragment: Set
   ..0. = More fragments: Not set
   Fragment offset: 0
   Time to live: 42
   Protocol: UDP (0x11)
   Header checksum: 0x280c (correct)
   Source: XXX-IPA.155.115.200.in-addr.arpa (200.115.155.XXX)
   Destination: lyla.domain.com (192.168.0.4)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
   Source port: 5060 (5060)
   Destination port: 5060 (5060)
   Length: 1050
   Checksum: 0xafc6 (correct)
Session Initiation Protocol
   Request-Line: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
   Method: INVITE
   Resent Packet: False
   Message Header
   Via: SIP/2.0/UDP 200.115.155.XXX:5060
   Via: SIP/2.0/UDP 200.115.155.YYY:5061;branch=z9hG4bK-5d2fda22
   From: office1 sip:[EMAIL PROTECTED];tag=c7f8491e8db6d4ao1
   SIP Display info: office1 
   SIP from address: sip:[EMAIL PROTECTED]

   SIP tag: c7f8491e8db6d4ao1
   To: sip:[EMAIL PROTECTED]
   SIP to address: sip:[EMAIL PROTECTED]
   Call-ID: [EMAIL PROTECTED]
   CSeq: 101 INVITE
   Max-Forwards: 69
   Contact: office1 sip:[EMAIL PROTECTED]:5060
   Expires: 240
   User-agent: Sipura/SPA3000-2.0.10(GWf)
   Content-Length: 431
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Supported: x-sipura
   Content-Type: application/sdp
   Record-Route: sip:200.115.155.XXX:5060;lr
   Message body
   Session Description Protocol
   Session Description Protocol Version (v): 0
   Owner/Creator, Session Id (o): - 20897320 20897320 IN IP4 
200.115.155.XXX
   Owner Username: -
   Session ID: 20897320
   Session Version: 20897320
   Owner Network Type: IN
   Owner Address Type: IP4
   Owner Address: 200.115.155.XXX
   Session Name (s): -
   Connection Information (c): IN IP4 200.115.155.XXX
   Connection Network Type: IN
   Connection Address Type: IP4
   Connection Address: 200.115.155.XXX
   Time Description, active time (t): 0 0
   Session Start Time: 0
   Session Stop Time: 0
   Media Description, name and address (m): audio 5004 RTP/AVP 4 0 2 8 
18 96 97 98 100 101
   Media Type: audio
   Media Port: 5004
   Media Proto: RTP/AVP
   Media Format: ITU-T G.723
   Media Format: ITU-T G.711 PCMU
   Media Format: ITU-T G.721

Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


Sergio Chersovani schrieb:

Tomasz Chmielewski ha scritto:

But I have (had?) a problem with 7905 phone (still minor problems with 
that, like a wrong timezone).



You can easy change it with the phone web page.


yup, I just figured that out :)

one more issue though.

any idea why a custom logo isn't displayed on a 7905G phone?

I see in tftp server logs that the logo file is downloaded, but it isn't 
there on a telephone display.


This same logo is displayed fine on a 7960 Cisco phone.


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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?

2005-10-24 Thread Erik

Logo file for a 7905 isn't a BMP file, it has to be converted with a cisco util


Tomasz Chmielewski wrote:
 Sergio Chersovani schrieb:
 
 Tomasz Chmielewski ha scritto:

 But I have (had?) a problem with 7905 phone (still minor problems
 with that, like a wrong timezone).



 You can easy change it with the phone web page.
 
 
 yup, I just figured that out :)
 
 one more issue though.
 
 any idea why a custom logo isn't displayed on a 7905G phone?
 
 I see in tftp server logs that the logo file is downloaded, but it isn't
 there on a telephone display.
 
 This same logo is displayed fine on a 7960 Cisco phone.
 
 


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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


Tomasz Chmielewski ha scritto:


any idea why a custom logo isn't displayed on a 7905G phone?


The logo image file need to be encoded. You will find the tools at the 
cisco website


Sergio

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[Asterisk-Users] new toy

[InfoWorld: Top News] Aruba unveils portable access point for VoIP
http://www.infoworld.com/cgi-bin/redirect?source=rssurl=http://www.infoworld.com/article/05/10/24/HNaruba_1.html

basically it creates a VPN connection to let remote users connect with
some level of security.  It also has an access point built in.  3x3
inches or about 7.6x7.6cm


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UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


Erik schrieb:

Logo file for a 7905 isn't a BMP file, it has to be converted with a cisco util


aah, now I see.
and what tool is that and where can I get this?
in my firmware package I have only two tools:

- cfgfmt.linux (a tool for converting text configuration into cisco 
format, which doesn't recognize 80% options)

- prserv.linux

I searched the whole Cisco IP Phone 7905 Series Administration Guide, 
but besides the copyright notes, logo is not mentioned.



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[Asterisk-Users] cvs head + spandsp

2005-10-24 Thread Cirelle Enterprises


is the cvs head version considered 1.0 or 1.1 with
regard to spandsp

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Cirelle Enterprises Inc. 
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603-425-2221

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Re: [Asterisk-Users] Where is the text of the voicemail email ??

2005-10-24 Thread Tom Rymes

On Oct 24, 2005, at 6:25 AM, Olle E. Johansson wrote:Guido Hecken wrote:I was looking for the text in the /etc/asterisk directory, but it mustbe somewhere else. Can anybody tell me where? And can it include Chineseas well?Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the/configs directory of your source code tree.I have never tried with Chinese, but it can handle Swedish :-)BTW wouldn't it be helpfull if the voicemailtext could depend on thelanguage, the user has choosen in extensions.conf? Example:User has language set to de, include language file de in voicemail.conf . Yes, that is a good idea. Any coders?I would like to be able to edit the pager notification e-mail. Right now it seems that the voicemail app's code has to be edited and recompiled to edit the pager message. We use numeric pagers, so I needed to add a numeric first line of the message body. The rest of the message can follow for alpha pagers, but the first line has to be numeric for numeric only pagers.Maybe someone could change that, because patching the code for every recompile is a bit of a painTom___
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Re: [Asterisk-Users] cvs head + spandsp

Cirelle Enterprises wrote:
 is the cvs head version considered 1.0 or 1.1 with
 regard to spandsp
 
CVS head would be considered 1.1 at this time.

/O

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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


Tomasz Chmielewski ha scritto:

I searched the whole Cisco IP Phone 7905 Series Administration 
Guide, but besides the copyright notes, logo is not mentioned.


lol
http://www.google.com/search?q=site%3Acisco.com+7905+bmp2logo

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Re: [Asterisk-Users] Where is the text of the voicemail email ??

2005-10-24 Thread Carlton O'Riley

Tom Rymes wrote:
 On Oct 24, 2005, at 6:25 AM, Olle E. Johansson wrote:
 
 Guido Hecken wrote:

 I was looking for the text in the /etc/asterisk directory, but it must
 be somewhere else. Can anybody tell me where? And can it include
 Chinese
 as well?


 Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
 /configs directory of your source code tree.

 I have never tried with Chinese, but it can handle Swedish :-)


 BTW wouldn't it be helpfull if the voicemailtext could depend on the
 language, the user has choosen in extensions.conf? 
 Example:
 User has language set to de, include language file de in voicemail.conf .


 Yes, that is a good idea. Any coders?
 
 
 I would like to be able to edit the pager notification e-mail. Right now
 it seems that the voicemail app's code has to be edited and recompiled
 to edit the pager message. We use numeric pagers, so I needed to add a
 numeric first line of the message body. The rest of the message can
 follow for alpha pagers, but the first line has to be numeric for
 numeric only pagers.
 
 Maybe someone could change that, because patching the code for every
 recompile is a bit of a pain
 
 Tom
 
 
 

This is a feature that has been asked for by one of my clients.  I was
wondering if this was going to be added to the voicemail.conf file,
seems like the only thing that isn't user configurable.  I doubt it
would take much code to do this.  I haven't looked into it as of yet,
but if I did fix it I'd like to make it part of the 1.2 and CVS-HEAD
release and not a patch that has to be loaded every single time.


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Re: [Asterisk-Users] cvs head + spandsp

2005-10-24 Thread Cirelle Enterprises





Olle E. Johansson wrote:

  Cirelle Enterprises wrote:
  
  
is the cvs head version considered 1.0 or 1.1 with
regard to spandsp


  
  CVS head would be considered 1.1 at this time.

/O

___

  

Thanks


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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?


Sergio Chersovani schrieb:

Tomasz Chmielewski ha scritto:

I searched the whole Cisco IP Phone 7905 Series Administration 
Guide, but besides the copyright notes, logo is not mentioned.



lol
http://www.google.com/search?q=site%3Acisco.com+7905+bmp2logo


I see, it's called bmp2logo.exe and it's for Windows only :(
anything like that that works with Linux?

BTW, I searched through 7905 Admin Guide for h323 (as it's the first 
link in google for cisco 7905 admin guide), assuming it's the same, 
and neither logo nor bmp2logo are mentioned there :)



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[Asterisk-Users] could set up only messagenet.it

2005-10-24 Thread Andres Baravalle

Hi,
I'm trying to configure a few service providers in asterisk. I could
configure correctly one only, messagenet.it.

I'm trying to register to sipgate and sipphone, with no success
(cannot register). I could configure in iax.conf voipbuster and
sipdiscount, it registers but it does not allow me to make calls.

I'm inside a (departmental) firewall, not sure of the ports that are
closed, but I can configure the voip providers in eyebeam without
problems.

Any suggestions?

Here are my configurations files:

sip.conf

[general]
context=default 
realm=rosario.dcs.shef.ac.uk
srvlookup=yes   
defaultexpirey=480
allow=all

;passwords are changed
register = 1234567:[EMAIL PROTECTED]:5061/1234567
register = 2234567:[EMAIL PROTECTED]/2234567
register = 3234567:[EMAIL PROTECTED]:5060/3234567

[messagenet]
type=peer
host=sip.messagenet.it
dtmfmode=info
username=1234567
secret=aaa
fromuser=1234567
fromdomain=sip.messagenet.it
nat=yes
authuser=1234567
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
port=5061

[1001]
type=friend
username=1001
fromuser=1001
host=dynamic
dtmfmode=rfc2833
secret=aa
disallow=all
allow=ulaw
allow=alaw
canreinvite=no

iax.conf:

[general]
allow=all   
jitterbuffer=no
tos=lowdelay

register = 1234567:[EMAIL PROTECTED]
register = 2234567:[EMAIL PROTECTED]

[voipbuster]
type=peer
host=iax.voipbuster.com
username= 1234567
secret= aaa
notransfer=yes
qualify=no
context=internal

[sipdiscount]
type=peer
host=sip.sipdiscount.com
username= 2234567
secret= aaa
notransfer=yes
qualify=no
context=internal

extensions.conf:

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest   
TRUNKMSD=1  

[default]
include = incoming
include = messagenet-outgoing
include = voipbuster-outgoing
include = sipdiscount-outgoing

[incoming]
exten = 1234567,1,Dial(SIP/1001,30,rt)
exten = 1234567,2,Dial(SIP/[EMAIL PROTECTED],30,rt)
exten = 1234567,3,Hangup

[messagenet-outgoing]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _9.,2,Congestion
exten = _9.,3,Busy

[voipbuster-outgoing]
exten = _8.,1,Dial(IAX2/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _8.,2,Congestion
exten = _8.,3,Busy

[sipdiscount-outgoing]
exten = _7.,1,Dial(IAX2/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _7.,2,Congestion
exten = _7.,3,Busy
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RE: [Asterisk-Users] Trying to clarify ideas about spands, libtiffFC4

2005-10-24 Thread Carlos Alperin

Doug,

Thank you very much, that is exactly the operation mode that I'm looking
for. Are you using a TDM405P card for the PRI?

Regards,

Carlos

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Monday, October 24, 2005 5:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Trying to clarify ideas about spands,
libtiffFC4

Carlos Alperin wrote:

Same question that before:


Mandrake/Mandrive 2.6.13.4 (thanks for the info)

What version of Asterisk?
  

CVS head, dated 2005-10-03

What version of Spandsp?
  

.0.0.2pre21 (I now see that there is a pres21a updated October 20th)

What version of Libtiff?
  

libtiff3-3.6.1-4.4.101mdk

What version of Libtiff-devel?
  

libtiff3-devel-3.6.1-4.4.101mdk

And the million dollars question: Is the fax working? (Lets say more than
50% of the cases?)

  

I only have around 3 people using it, but one of them receive around 20 
faxes a day with minimal complaints.  Faxes are coming over a Pri and 
converted to PDF before being sent via email to the end user.

Doug


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RE: [Asterisk-Users] Trying to clarify ideas about spands, libtiff FC4

2005-10-24 Thread Carlos Alperin

Thanks I'll try it.

Regards,

Carlos

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent: Monday, October 24, 2005 12:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Trying to clarify ideas about spands,libtiff 
FC4

I've been using spandsp 0.0.2pre18 and 0.0.2pre21 with libtiff 3.7.3,
Asterisk 
1.0.7 (now 1.0.9) and the Linux kernel 2.6.11.5 kernel with results that are

good enough for me (I'm using fax over IP with the G.711 codec).


On Sunday 23 October 2005 13:23, Carlos Alperin wrote:
 I spent more than 3 weeks, with some little help of people that belongs to
 this forum, and after try differents combinations of versions this is my
 conclusion:



 I tried RH9, FC4  FC4 64

 I tried with CVS 1.0.2, and Stable 1.0.9

 I tried with spandsp 0.0.2pre18, 0.0.2pre20  0.0.2pre21

 Libtiff 3.5.7  libtiff devel 3.5.7

 Libtiff 3.7.1  libtiff devel 3.7.3 (I couldn't find 3.7.1)



 My conclusion is:



 If I need to be able to use fax with Spandsp, app_rxfax.c  app_txfax.c
 with libtiff 3.5.7 (and libtiff devel 3.5.7) there is no way to do that on
 FC4 (get conflict with GTK2+)

 So it looks like I have to go back to RH9 and at least upgrade to kernel
 2.4.31, and try again.



 This is under the presumption that Spandsp,  the rest are going to work.
 (Looking at the forum, that is not a 100% fact).



 It should be a way to save us a lot of time, if somebody can unify all the
 requeriments on each OS, so we can decide before to start which direction
 to follow.



 The reason for RH9  FC4 is because they're more familiar. But if someone
 can show me a working configuration, I don't hesitate to move the
platform.



 By the way, the 64 bits platform still looks to be very unstable and not
so
 fast to implement with Asterisk.



 To the digium support: I understand that your recommendation is to go to
 2.6 kernel, but if I need to run spandsp, how to do that without libtiff
 3.5.7.



 The general experience is libtiff 3.7.1 locks the asterisk when the
machine
 boots.



 Please feel free to send every kind of disappointments opinions. That is
 going to feel me much better that no answers.

 (Even if you can show me how stupid I was doing all kind of mistakes)



 Regards,



 Carlos Alperin

-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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Re: [Asterisk-Users] Trying to clarify ideas about spands,

2005-10-24 Thread Doug Lytle


Carlos Alperin wrote:


Doug,

Thank you very much, that is exactly the operation mode that I'm looking
for. Are you using a TDM405P card for the PRI?

Regards,
 



TE110P 


Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[Asterisk-Users] Thounsand of SIP extension

2005-10-24 Thread Juanjo Portela

Dear colleagues,

I read on the web that implementations of thousand SIP extension on
asterisk became worse and people suggest SER + asterisk.
Anybody checks this? what is the problem?

Any comment

Kind regards,
Juanjo
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[Asterisk-Users] Re: Custom handling of SIP 302 redirect?

On 10/21/05, Steve Davies [EMAIL PROTECTED] wrote:

 I have noticed that when a SIP redirect is sent back to Asterisk by a
 SIP peer, that Asterisk will (quite appropriately) do a
 Dial(LOCAL/redirect-number) in the context of the original callee.

 It also forks the CDR, which is excellent. Sadly, under these
 circumstances, I need to alter the caller-ID to be a valid value, set
 the 'src' to be the correct extension no., and set the accountcode to
 something recognisable as an outbound call by that user.


I have managed to get part-way through this problem... It seems that
ACCOUNTCODE persists across the dial, so I can set that to a
meaningful value most of the time, and use the data later for billing,
sadly this is only a small (20 character) field, so I can only
transfer a limited amount of data.

Other fields such as userdata do not persist, and variables that are
set in the dialplan do not stay in-scope either. Can anyone suggest
another mechanism for passing data across?

Perhaps this should be raised as a feature-request such that the
caller-ID field is populated from the SIP client that sends the
redirect? Looking at the source I expected this to happen already, but
it is a fairly complex interaction.

Kind regards,
Steve
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Re: [Asterisk-Users] Re: Custom handling of SIP 302 redirect?

Steve Davies wrote:
 On 10/21/05, Steve Davies [EMAIL PROTECTED] wrote:
 
I have noticed that when a SIP redirect is sent back to Asterisk by a
SIP peer, that Asterisk will (quite appropriately) do a
Dial(LOCAL/redirect-number) in the context of the original callee.

It also forks the CDR, which is excellent. Sadly, under these
circumstances, I need to alter the caller-ID to be a valid value, set
the 'src' to be the correct extension no., and set the accountcode to
something recognisable as an outbound call by that user.

 
 
 I have managed to get part-way through this problem... It seems that
 ACCOUNTCODE persists across the dial, so I can set that to a
 meaningful value most of the time, and use the data later for billing,
 sadly this is only a small (20 character) field, so I can only
 transfer a limited amount of data.
 
 Other fields such as userdata do not persist, and variables that are
 set in the dialplan do not stay in-scope either. Can anyone suggest
 another mechanism for passing data across?
 
 Perhaps this should be raised as a feature-request such that the
 caller-ID field is populated from the SIP client that sends the
 redirect? Looking at the source I expected this to happen already, but
 it is a fairly complex interaction.
 
Just so you don't have to comment on your own comment to your own
mail... :-)

It should go through the dialplan. What we could do is to set a variable
so you could catch it being a call forward in the dial plan so you could
treat it any way you want.

Another question is the context. We have the normal context, the
transfer context, the subscription context - do we need to add another
context or can we reuse one of these for forwards? I would suspect that
using the transfer context with a flag for call forwarding
(CALLFORWARD=yes) would work.

/O
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RE: [Asterisk-Users] Hardware setup question

2005-10-24 Thread Juan Janczuk



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Robert Webb
Enviado el: Domingo, 23 de Octubre de 2005 06:24 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Hardware setup question



I have just a quick setup question about how some of you
have hardware setup.

Basically, for a system that has an average volumes of
calls in an office setting, are you using one or two
network cards. I am just wondering if it owuld be any
advantage to having one NIC for the extensions and one NIC
for your trunks.

Robert
___

Hi, Robert.

I have 2 nic's, but not divided as you mention.
As my server is serving phones on the LAN and WAN sides, I have a nic for
the internal network, and another nic for the internet side.
The internal one, manages only phones (Hardware SIP phones in my case), and
the external is managing IAX trunks (Test trunks, up to now), and external
hardware SIP phones connected via VPN.

Hope this help.
Juan.
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.361 / Virus Database: 267.12.5/147 - Release Date: 24/10/2005

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[Asterisk-Users] How to setup parked/on-hold so sorresponding buttons on VoIP phones light up

2005-10-24 Thread Christian Buchter




We have Snom 190s in an office of about 30. Trying to use the 5 lit
buttons on the right to be used for parked calls/calls on hold. In other
words, want to be able to transfer someone to either an extension that
maps to the buttons or anyone on hold gets put into that queue of lit
buttons so anyone else can pick up.

Anyone doing anything similar with the Snom 190s?

TIA




_
This email has been scanned by MessageLabs on behalf of E-INS
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[Asterisk-Users] Hangup ZAP channel

2005-10-24 Thread Juanjo Portela

Dear Colleagues,

I Have my * with a X100P clon card. When a call in from the PSTN and
nobody answer the call go to the voicemail, then the caller my hangup
or press #. If the caller hangup the ZAP channel never hangup, but if
the caller press # the ZAP channel hangup. Even every time the outside
part of the communication hangup the ZAP channel doesn´t detect
anything and never hangup the channel.
What is going wrong? Mayyou help me?

Thank you in advance,
Juanjo
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Re: [Asterisk-Users] 0.2.0-RC8o (* 1.0.9) + No Caller ID

2005-10-24 Thread Massimo De Nadal


Giovanni Miano wrote:


I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o
and * 1.0.9
I dont recive callerid from TDM400P fxo port but isdn hasnt problems
If i try to use only  TDM400P 1fxs/1fxo without bristuff.. all work ok
is it bug of bristuff ?
 


Maybe, why not try bristuff 0.2.0-RC8p ?
For me works fine (tdm400p cid detection).

maxx



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[Asterisk-Users] polycom, call waiting, queues

2005-10-24 Thread Bartosz Jozwiak


Hi Guys,

I have a small problem.
I would like to disable call waiting function in Polycom phones while all
calls are handled by queues.
So far nothing, could not find an option in Polycom config to disable it.
Any help would be appreciated.

Bartosz

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Re: [Asterisk-Users] Re: Custom handling of SIP 302 redirect?

On 10/24/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
 Steve Davies wrote:
  On 10/21/05, Steve Davies [EMAIL PROTECTED] wrote:
 
 I have noticed that when a SIP redirect is sent back to Asterisk by a
 SIP peer, that Asterisk will (quite appropriately) do a
 Dial(LOCAL/redirect-number) in the context of the original callee.
 
 It also forks the CDR, which is excellent. Sadly, under these
 circumstances, I need to alter the caller-ID to be a valid value, set
 the 'src' to be the correct extension no., and set the accountcode to
 something recognisable as an outbound call by that user.
 
  I have managed to get part-way through this problem... It seems that
  ACCOUNTCODE persists across the dial, so I can set that to a
  meaningful value most of the time, and use the data later for billing,
  sadly this is only a small (20 character) field, so I can only
  transfer a limited amount of data.
 
  Other fields such as userdata do not persist, and variables that are
  set in the dialplan do not stay in-scope either. Can anyone suggest
  another mechanism for passing data across?
 
  Perhaps this should be raised as a feature-request such that the
  caller-ID field is populated from the SIP client that sends the
  redirect? Looking at the source I expected this to happen already, but
  it is a fairly complex interaction.
 
 Just so you don't have to comment on your own comment to your own
 mail... :-)

:-) The thought is much appreciated.

 It should go through the dialplan. What we could do is to set a variable
 so you could catch it being a call forward in the dial plan so you could
 treat it any way you want.

 Another question is the context. We have the normal context, the
 transfer context, the subscription context - do we need to add another
 context or can we reuse one of these for forwards? I would suspect that
 using the transfer context with a flag for call forwarding
 (CALLFORWARD=yes) would work.

I agree that the transfer context, falling back to the subscription
context of the callee would make sense. I like the CALLFORWARD=yes
idea, although it might be useful to extend this so it would add
CALLFORWARDBY=SIP/phone1 and CALLFORWARDEXTEN=nnn where nnn is the
value of $EXTEN when the redirect occurred.

Of course all of this could be done manually if variables stayed
in-scope through the redirect.

Thanks again,
Steve
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[Asterisk-Users] Add SIP extension

2005-10-24 Thread Chrispen Chisvo


hi

Anyone to help me add extensions to my Asterisk PBX. Step by step please, help me.

I have been doing the following: you can correct me if I am missing anything:

on the xlite client running Windows XP, on ip address 192.168.1.35:
-Enable: Yes
- Display name: anytext
- Username:135
- Authorisation User: 135
- Password:
- Domain/Realm: 192.168.1.37
- Sip Proxy: 192.168.1.37
- Outbound Proxy: 192.168.1.37

in the extension.conf file I put the following:
exten = 135,1,Dial(SIP/[EMAIL PROTECTED])

Now I do a similar thing for the other extension, the two can dial each other, but I am not getting voice to go through. I am running Asterisk on SUSE Linux 9.3.

Another thing, the xlite clients cannot register on the PBX, I get the following from Asterisk:

Oct 24 17:20:21 NOTICE[25549]: chan_sip.c:7733 handle_request: Registration from 'Osie sip:[EMAIL PROTECTED]' failed for '192.168.1.35'

Help


-- 
Rgds
Chrispen Chisvo
Ecoweb Zimbabwe
Cell: +263 91 222 443
Tel: +263 4 758 194
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[Asterisk-Users] Largest working config files?

Hi,

I hope this is not a FAQ - I have not been able to find it if it is
covered already...

I have a dial-plan on my asterisk system that is becoming potentially
quite large and complex - Of the order of 12 lines of dialplan per
extension number. Most of this is in order to record suitable CDR
data, access voicemail, and play polite goodbye messages etc. The
operation of each extension can potentially be unique, making a common
[extensions-generic] almost impossible to write.

Does anybody have experience of how big an extensions.conf can get
before problems start occuring? If anyone has experienced problems,
what sort of things happen?

Thanks,
Steve
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-24 Thread Eric \ManxPower\ Wieling

In CVS-HEAD and 1.2Beta the new KB1 echocan is enabled by default and 
has solved most of our echo issues.


stoffell wrote:

On 10/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Yes I did notice it immediately. I intend to tweak more, but for the
moment it seems like echo is minimized to zero.




I also encountered some echo problems and used (uncommented :)) following
parameters in zconfig.h:
#define ECHO_CAN_MARK3 (instead of MARK2)
#define CONFIG_CALC_XLAW
#define CONFIG_ZAPTEL_MMX

Up untill now it seems to be much better.. It also 'sounds' much better
during normal conversation.

Oh, and in the Makefile, changed some flags:
KFLAGS+=-DSTANDALONE_ZAPATA -march=pentium4 -O3 -fomit-frame-pointer
CFLAGS+=-DSTANDALONE_ZAPATA -march=pentium4 -O3 -fomit-frame-pointer

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RE: [Asterisk-Users] Largest working config files?

2005-10-24 Thread Kevin Walsh

Steve Davies [EMAIL PROTECTED] wrote:
 I hope this is not a FAQ - I have not been able to find it if it is
 covered already... 
 
 I have a dial-plan on my asterisk system that is becoming potentially
 quite large and complex - Of the order of 12 lines of dialplan per
 extension number. Most of this is in order to record suitable CDR
 data, access voicemail, and play polite goodbye messages etc. The
 operation of each extension can potentially be unique, making a common
 [extensions-generic] almost impossible to write.

Have you looked into creating a couple of macros to reuse your code?

[local-extensions]
exten = 2100,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
exten = 2101,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
exten = 2102,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
exten = 2103,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
...

As you can see, I only need one line per extension;  All of the call
logic is in the [macro-call-local] macro.  The maintenance is a lot
simpler too, of course.

 
 Does anybody have experience of how big an extensions.conf can get
 before problems start occuring? If anyone has experienced problems, what
 sort of things happen? 
 
I have no idea.  Here's our dialplan line count (quite small because of
the macros):

 218 extensions/incoming.conf
 354 extensions/internal.conf
 225 extensions/macros.conf
 471 extensions/outgoing.conf
 151 extensions/routes.conf
1419 total

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] polycom, call waiting, queues

2005-10-24 Thread Chris HARIGA


Bartosz Jozwiak wrote:


Hi Guys,

I have a small problem.
I would like to disable call waiting function in Polycom phones while all
calls are handled by queues.
So far nothing, could not find an option in Polycom config to disable it.
Any help would be appreciated.

Bartosz

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Hi,

U can doit from asterisk, not from the phone.

Best regards,

Chris HARIGA
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Re: [Asterisk-Users] Modem Over IP: solutions ?

2005-10-24 Thread Jorge Mendoza

We use use RS232 to Ethernet converters to solve this kind of 
applications, for instance Moxa.


Jorge Mendoza

Jean-Michel Hiver wrote:

Hi,

I have a potential client who has legacy alarm systems which use 
modems to transmit encoded data to a remote location through the PSTN. 
They wish to replace the 'PSTN' bit with an IP link.


I am aware that it would be best if the data was transmitted directly 
over IP rather than modulated and then sent on the internet, but that 
is not possible because of the legacy equipment.


I was wondering if there was some specialized ATAs of some kind that 
would do TDMoIP and which could be used for this purpose?


Link latency is about 300ms with no more than 10ms jitter. If you have 
a solution please let me know!


Cheers,
Jean-Michel.


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Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

2005-10-24 Thread Dave Grey



On Oct 23, 2005, at 11:57 PM, Leif Madsen wrote:


On 10/20/05, Darrick Hartman [EMAIL PROTECTED] wrote:


Leif Madsen wrote:


 For those of you who are able to obtain the
full copy, please consider helping us out by creating mirrors and
torrents and posting them to the list by replying to this thread.
Thanks!

Is there any reason why the book wasn't released as a single pdf  
rather
than the individual chapter pdf's?  Using pdftk, I merged the pdfs  
back

into a single document (11mb), then zipped it back up.


Hrmmm... that is a good question, because I guess technically you're
not changing it. However, for now, lets just leave it as be.


The first thing I did was merge it into a single document as well  
(3.2MB total using Acrobat 7, all content included), then I cropped  
the pages down to the print size rather than the 8.5x11 with  
registration marks that they were (1.3 MB total, all content  
included) then I loaded it on my palm.  ;-)


I don't know what was used to create those pdf's, but they definitely  
don't need to be so huge.


Good book so far, btw, I have been enjoying it!

lyd



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Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

2005-10-24 Thread Chrispen Chisvo

Where is the book?

Link please?

On Monday 24 October 2005 10:30, Dave Grey wrote:
 On Oct 23, 2005, at 11:57 PM, Leif Madsen wrote:
  On 10/20/05, Darrick Hartman [EMAIL PROTECTED] wrote:
  Leif Madsen wrote:
   For those of you who are able to obtain the
  full copy, please consider helping us out by creating mirrors and
  torrents and posting them to the list by replying to this thread.
  Thanks!
 
  Is there any reason why the book wasn't released as a single pdf
  rather
  than the individual chapter pdf's?  Using pdftk, I merged the pdfs
  back
  into a single document (11mb), then zipped it back up.
 
  Hrmmm... that is a good question, because I guess technically you're
  not changing it. However, for now, lets just leave it as be.

 The first thing I did was merge it into a single document as well
 (3.2MB total using Acrobat 7, all content included), then I cropped
 the pages down to the print size rather than the 8.5x11 with
 registration marks that they were (1.3 MB total, all content
 included) then I loaded it on my palm.  ;-)

 I don't know what was used to create those pdf's, but they definitely
 don't need to be so huge.

 Good book so far, btw, I have been enjoying it!

 lyd



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Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

2005-10-24 Thread Leif Madsen

On 10/24/05, Chrispen Chisvo [EMAIL PROTECTED] wrote:
 Where is the book?

 Link please?

From my post 11 hours ago in this thread...

Available here: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

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Re: [Asterisk-Users] new toy

2005-10-24 Thread Luki

 [InfoWorld: Top News] Aruba unveils portable access point for VoIP

Funny you mention this... I'm currently testing a similar setup,
Asterisk on OpenWRT:

WAN - WRT54G - [SSL] - UDPTunnel - [IAX] - Asterisk - [SIP] - WiFi

And SIP clients via WiFi, or via wired LAN. At this point this is NOT
using encryption, but I am planing to add stunnel (or similar). A VPN
(like PPTP) is not an option for me because the environment I'm
testing in blocks all but outgoing TCP connections. Hence a SSH-like
TCP connection may work best.

Total cost: $60 for router, plus some hours of fun to get it all
going. But you also do other useful things like traffic shaping on the
router...
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Re: [Asterisk-Users] polycom, call waiting, queues

2005-10-24 Thread Jerry Jones


set calls per button to one - in 1.5 and later code

On Oct 24, 2005, at 11:21 AM, Chris HARIGA wrote:


Bartosz Jozwiak wrote:



Hi Guys,

I have a small problem.
I would like to disable call waiting function in Polycom phones  
while all

calls are handled by queues.
So far nothing, could not find an option in Polycom config to  
disable it.

Any help would be appreciated.

Bartosz

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Hi,

U can doit from asterisk, not from the phone.

Best regards,

Chris HARIGA
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Re: [Asterisk-Users] polycom, call waiting, queues

2005-10-24 Thread Bartosz Jozwiak






Bartosz Jozwiak wrote:


Hi Guys,

I have a small problem.
I would like to disable call waiting function in Polycom phones while all
calls are handled by queues.
So far nothing, could not find an option in Polycom config to disable it.
Any help would be appreciated.

Bartosz



Hi,

U can doit from asterisk, not from the phone.

Best regards,

Chris HARIGA


If i set limits in sip.conf then I cannot make transfers. So this is not a 
solution for me.

With set groups it will not work because this is a queue.

Bartosz 


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Re: [Asterisk-Users] Largest working config files?

On 10/24/05, Kevin Walsh [EMAIL PROTECTED] wrote:
 Steve Davies [EMAIL PROTECTED] wrote:
  I hope this is not a FAQ - I have not been able to find it if it is
  covered already...
 
  I have a dial-plan on my asterisk system that is becoming potentially
  quite large and complex - Of the order of 12 lines of dialplan per
  extension number. Most of this is in order to record suitable CDR
  data, access voicemail, and play polite goodbye messages etc. The
  operation of each extension can potentially be unique, making a common
  [extensions-generic] almost impossible to write.
 
[snip]
 
  Does anybody have experience of how big an extensions.conf can get
  before problems start occuring? If anyone has experienced problems, what
  sort of things happen?
 
 I have no idea.  Here's our dialplan line count (quite small because of
 the macros):

  218 extensions/incoming.conf
  354 extensions/internal.conf
  225 extensions/macros.conf
  471 extensions/outgoing.conf
  151 extensions/routes.conf
 1419 total


Thanks for the figures.

I will almost certainly use Macros to simplify how things work at
present, but will probably only save myself 20-30% due to the number
of varieties of behaviour possible.

Still, 30% is worth having :)

Regards,
Steve
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[Asterisk-Users] Asterisk vs Sipura SIP problem?

2005-10-24 Thread Frank Tarczynski

I am trying to use a SIP provider for outgoing and incoming calls under
Asterisk.  I am running a recent CVS-head 1.09 build and the SIP
provider is using a SPA-3000.  I can register with the SIP provider's
server and outgoing calls seem to work just fine.

But I cannot get incoming calls to work at all.

I see absolutely no indication in the Asterisk SIP debug output that
incoming SIP calls are coming from this provider!  But in output from
ethereal I find that my Asterisk box responds to the initial INVITE with
a 484 Address Incomplete.  There is no response from the SIP provider
and a few seconds later my Asterisk sends an ACKnowledge.

Absolutely none of this shows-up in the Asterisk output!

The INVITE is addressed to sip:[EMAIL PROTECTED]:5060 and all my Asterisk
extensions are 4 digits starting with 1s.  Shouldn't the SPA-3000
respond back to the 484 again?  Or since it is using just 1 is no
additional response sent?

I tried creating an extension context of [1] but this has no effect.  I
just keep getting the 484 responses.

Do I need to ask the SIP provider to configure the SPA-3000
differently?  Have the INVITE request changed?  How/where would I create
a context that Asterisk can use/understand?

No. TimeSourceDestination  
Protocol Info
   1016 20:48:20.196168 XXX-IPA.155.115.200.in-addr.arpa lyla.domian.com  
SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description

Frame 1016 (1084 bytes on wire, 1084 bytes captured)
Arrival Time: Oct 23, 2005 20:48:20.196168000
Time delta from previous packet: 0.00161 seconds
Time since reference or first frame: 32.762675000 seconds
Frame Number: 1016
Packet Length: 1084 bytes
Capture Length: 1084 bytes
Ethernet II, Src: 00:04:e2:bc:76:80, Dst: 00:0e:0c:62:cb:08
Destination: 00:0e:0c:62:cb:08 (lyla.domain.com)
Source: 00:04:e2:bc:76:80 (ipcop.domain.com)
Type: IP (0x0800)
Internet Protocol, Src Addr: XXX-IPA.155.115.200.in-addr.arpa
(200.115.155.XXX), Dst
Addr: lyla.domain.com (192.168.0.4)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
 00.. = Differentiated Services Codepoint: Default (0x00)
 ..0. = ECN-Capable Transport (ECT): 0
 ...0 = ECN-CE: 0
Total Length: 1070
Identification: 0x (0)
Flags: 0x04 (Don't Fragment)
0... = Reserved bit: Not set
.1.. = Don't fragment: Set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 42
Protocol: UDP (0x11)
Header checksum: 0x280c (correct)
Source: XXX-IPA.155.115.200.in-addr.arpa (200.115.155.XXX)
Destination: lyla.domain.com (192.168.0.4)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Source port: 5060 (5060)
Destination port: 5060 (5060)
Length: 1050
Checksum: 0xafc6 (correct)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 200.115.155.XXX:5060
Via: SIP/2.0/UDP 200.115.155.YYY:5061;branch=z9hG4bK-5d2fda22
From: office1 sip:[EMAIL PROTECTED];tag=c7f8491e8db6d4ao1
SIP Display info: office1
SIP from address: sip:[EMAIL PROTECTED]
SIP tag: c7f8491e8db6d4ao1
To: sip:[EMAIL PROTECTED]
SIP to address: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 69
Contact: office1 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 431
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Record-Route: sip:200.115.155.XXX:5060;lr
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 20897320 20897320 IN IP4
200.115.155.XXX
Owner Username: -
Session ID: 20897320
Session Version: 20897320
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 200.115.155.XXX
Session Name (s): -
Connection Information (c): IN IP4 200.115.155.XXX
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 200.115.155.XXX
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 5004 RTP/AVP 4
0 2 8 18
96 97 98 100 101
Media Type: audio
Media Port: 5004
Media Proto: RTP/AVP
Media Format: ITU-T G.723
Media Format: ITU-T G.711 PCMU

Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

2005-10-24 Thread Gavin Spurgeon

Hi List..

How do I offer my help (and bandwidth) to become a mirror for this Book ?

Best Regards


Gavin Spurgeon
Assistant Systems Administrator
[EMAIL PROTECTED]
http://www.leighctc.kent.sch.uk 
Tel: 01322 620501
Fax: 01322 620599
IS HelpDesk : Ext 541


-- 
This message has been scanned for viruses and
dangerous content by the Systems @ the LeighCTC,
and is believed to be clean.

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[Asterisk-Users] RE: Meetme admin option

2005-10-24 Thread Anish Basu

After thinking about it for a few days, I realized that one way to prevent
non-admin users from entering the conference room is to use an AGI script
that actually performs the authentication.  But, I would rather have the
functionality built into the Meetme application.  Are there is any plans in
the near future for implementing this kind of control?  Or should I consider
posting a bounty for this?

Anish Basu
Field Systems Engineer
Softel, Inc.
Phone: (732) 705-9202
Cell: (732) 312-6634 

-Original Message-
From: Anish Basu [mailto:[EMAIL PROTECTED] 
Sent: Friday, October 21, 2005 4:16 PM
To: asterisk-users@lists.digium.com
Subject: Meetme admin option

There is an Meetme command option 'a' for admin.  I tried using this option
and noticed that it allows users to login with the user pin as well as the
admin pin.  In my dialpan I have:

exten = 700, 1, Meetme(500,Mas)

And in meetme.conf, I have:

conf = 500,1234,

After dialing extension 700, I was able to login to the conference using the
user pin '1234'.  When I pressed the star key, I was presented with the
voicemenu Press 1 to mute/unmute yourself, 2 to lock/unlock this
conference, or press 3 to eject the last user, which should only be for
admin. Is there any way to restrict users from logging in unless they have
the correct admin pin?

Anish Basu
Field Systems Engineer
Softel, Inc.
Phone: (732) 705-9202
Cell: (732) 312-6634 

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Re: [Asterisk-Users] Passing parametrs to php agi scripts.

2005-10-24 Thread Kevin Bockman


Adam Rybak wrote:

s,1,DaeadAGI,test.php,parameter1

How get value of parameter1 in php script?
This is actually a PHP question.  You can find it in the PHP manual 
online at http://www.php.net


$_SERVER['argv'][1]


Kevin
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[Asterisk-Users] Urgent - Need Help - Audio Issues

2005-10-24 Thread Dave

Hello, I am running asterisk v 1.2.0Beta on a HP AMD
64 box. Redhat FC4 is installed on the box. I was not
able to compile mpg123 and am therefore using the
moh_native for moh. The snd_atiixp driver is loaded
for the sound card and I can play the demo sound using
desktop utility just fine.

Asterisk is registered to my SIP provider for outbound
and inbound calls and I have X.lite, Sipura phones in
my internal LAN.  

Now each time I playback any sound file, It is played
at double the speed (chipmunk) making it useless. The
happens whether I call from an internal Xlite or
Sipura or from the PSTN (in which case Asterisk get a
SIP INVITE). I also recorded a voice sample using
Record() and that also gives the same result. 

Please help!
Dave




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[Asterisk-Users] SIP to CAPI - Soundcard required?

2005-10-24 Thread Sascha Andres

Hi,

I've a strange problem here. I can dial out via an AVM B1 card.
I have a sip client running. I can hear my conversational partner
but he can't here me. I'm using * 1.0.

Has anyone got this behavior?

Sascha

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Re: [Asterisk-Users] polycom, call waiting, queues

2005-10-24 Thread Mojo with Horan Company, LLC

This is probably the best option for you.  Upgrade to 1.5.2 if that 
works for you and set this option as Jerry mentions.  It's accessible on 
the phone menu.


Jerry Jones wrote:

set calls per button to one - in 1.5 and later code

On Oct 24, 2005, at 11:21 AM, Chris HARIGA wrote:



Bartosz Jozwiak wrote:




Hi Guys,

I have a small problem.
I would like to disable call waiting function in Polycom phones  
while all

calls are handled by queues.
So far nothing, could not find an option in Polycom config to  
disable it.

Any help would be appreciated.

Bartosz

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Hi,

U can doit from asterisk, not from the phone.

Best regards,

Chris HARIGA
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Re: [Asterisk-Users] new toy