date:20050402

Can I use SJphone like a H.323 phone in order to dial and receive call
through Asterisk?Can I consider Asterisk just like a sort of H323
Gateway?

Thanks 4 all!

flx
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[Asterisk-Users] Version 0.72 of IPSwitchBoard Released

2005-04-02 Thread Thorben Jensen









Version 0.72 - 2.
April 2005. Download from http://ipswitchboard.thorben.dk



 Danish
language added

 Transfer
to Queues now supported

 You can
return a substring of the ${CALLERID} variable used in the URL for CRM systems.




To enable
transfer to Queues you must type in an extension that IPSwitchBoard should
transfer to on the Queues Page in the extension column. You can only edit the
extension column when you are not connected to the Asterisk server.



You can return a
substring of the CALLERID used in the URL for a CRM system on the config page:



${CALLERID:offset:length}




Returns a
substring of the string CALLERID, beginning at offset offset and returning the
next length characters. 

If offset is
negative, it is taken leftwards from the right hand end of the string. 



If length is
omitted or is negative, then all the rest of the string beginning at offset is
returned. 



Examples: 

${CALLERID} =
123456789



${CALLERID:1}-returnsthestring23456789


${CALLERID:-4}-returnsthestring6789


${CALLERID:0:3}-returnsthestring123


${CALLERID:2:3}-returnsthestring345


${CALLERID:-4:3}-returnsthestring678



___

IPSwitchBoard is
an FREE Operators Switch Board for Asterisk users 



IPSwitchBoard is
a FREE Windows.Net application that will allow you to do: 


Unattended/attended transfers. 

 Park
calls and retrieve/forward them again. 

 Organize
all your SIP and IAX extensions (automatically retreived from Asterisk). 

 Monitor
all extensions. 

 Monitor
all queues. 

 Monitor
Agents. 

 Monitor
Parked Calls. 


Dynamically log extensions in and out of queues. 


Integration with CRM software on the web. 

 Record
conversations. 

 Drop any
active call. 


Import/Export extensions to/from Asterisk Server DB. 

 Set Do
Not Disturb on Extensions and give a reason. 

 Speed
Dialling. 

 Share
Speed Dial files among all users of IPSwitchBoard. 

 User
selectable ring tones for IPSwitchBoard. 

 User
selectable button colors. 








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[Asterisk-Users] VAD In Asterisk with Zaptel

2005-04-02 Thread parijat

Hi,

I want to know if VAD is possible in Zaptel Analog lines

Reg,
parijat
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[Asterisk-Users] (no subject)

2005-04-02 Thread parijat

Hi,

I want to know if VAD is possible in Zaptel Analog lines

Reg,
parijat
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RE: [Asterisk-Users] VAD In Asterisk with Zaptel

2005-04-02 Thread parijat

Hi,

I want to know if VAD is possible in Zaptel Analog lines

Reg,
parijat

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[Asterisk-Users] Delaying answer of incoming calls

2005-04-02 Thread Justin Ridge

Hi all,
I'd like Asterisk to answer an incoming SIP call and bridge with a POTS 
line.  However, I do NOT want Asterisk to answer the SIP call unless the 
called POTS number answers (otherwise SIP caller is charged).

The wiki says this is the default behaviour, and so does this message
http://lists.digium.com/pipermail/asterisk-users/2004-April/041881.html
But it's not working as expected.  The SIP call is answered immediately 
before the POTS line even starts ringing.

The dial plan is simple:
[from-sip]
exten = s,1,Dial(Zap/1/*829x,20)
I don't have Answer in the dial plan.  I don't think I changed any 
settings that would cause this non-default behaviour, but can you help me 
find why this is happening?

Thanks
JR
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[Asterisk-Users] :: Strange way of receiving calls ::

2005-04-02 Thread Reuben Grech




I have not found any 
references to a strange way in which I would like to receive calls. I have 
Asterisk running with 3 x X100P cards and using AMP. Can anyone give me 
some help on receiving calls in the following manner:

=An incoming 
call is received on any of the incoming POTS ZAP channels
=Call is 
immediately picked up by Asterisk and the caller is greeted with a message like, 
"Thank you for calling , please hold the line your call will be attended to 
shortly"MUSIC
= What I would like 
to achieve is that as soon as the call is immediately picked up by Asterisk, a 
group of phones starts ringing, so that effectively any person in the group can 
attend to the caller, and as soon as this is done, the caller stops hearing the 
message and can speak to a representative.

With AMP I managed 
to set up a group which rings with an incoming POTS call. With AMP also, I 
have also managed to create a Digital Receptionist BUT this requires caller 
input, which is not what I need :(

Any help would be 
appreciated! :) Thanks to you all

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Re: [Asterisk-Users] Re: Are there online forums instead of this email

2005-04-02 Thread Bob Goddard

On Friday 01 April 2005 23:03, Tim Bass wrote:
 Congratulations!

 Tom Ivar Helbekkmo  and Francesco Peeters

 Voted Number One Bullies of Asterisk-Users.

You are the bully. So far the majority wish the email list to
continue and yet you still continue to demand that Digium
convert to a forum.
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Re: [Asterisk-Users] :: Strange way of receiving calls ::

2005-04-02 Thread Peter Svensson

On Sat, 2 Apr 2005, Reuben Grech wrote:

 I have not found any references to a strange way in which I would like to
 receive calls.  I have Asterisk running with 3 x X100P cards and using AMP.
 Can anyone give me some help on receiving calls in the following manner:
  
 = An incoming call is received on any of the incoming POTS ZAP channels
 = Call is immediately picked up by Asterisk and the caller is greeted with a
 message like, Thank you for calling , please hold the line your call
 will be attended to shortlyMUSIC
 = What I would like to achieve is that as soon as the call is immediately
 picked up by Asterisk, a group of phones starts ringing, so that effectively
 any person in the group can attend to the caller, and as soon as this is
 done, the caller stops hearing the message and can speak to a
 representative.
  
 With AMP I managed to set up a group which rings with an incoming POTS call.
 With AMP also, I have also managed to create a Digital Receptionist BUT this
 requires caller input, which is not what I need :(

Woulden't a queue with ringall2 policy and music on hold do exactly what 
you want?

Peter


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[Asterisk-Users] H.323 call '.....' cleared, reason 8 (Transport failure)

2005-04-02 Thread Cenk Yabas




I installed the oh323 channel 
driver and registered to the gate keeper succesfully.

I come through the GK, ring the dialed number forabout 0.5 seconds andloose the line.I contacted the GKand they report that they receive the correct 
dialstring to route the call but the call is ended by my side.
The dialstring looks like 
this:
exten = 
_.,1,Dial(OH323/${EXTEN},60,r)
I use the following channel driver:

asterisk-oh323-0.7.1
openh323-Janus_patch4-src
pwlib-Janus_patch4-src

and the message on asterisk console looks like this:
-- Registered with gatekeeper 
'[EMAIL PROTECTED]'.
-- Executing Dial("SIP/2000-c9fc", "OH323/0012029361212|60|r") 
in new stack
-- H.323 call to 0012029361212 with codec(s) g729
-- Called 0012029361212
-- H.323 call 'ip$localhost/2209' cleared, reason 8 (Transport 
failure)
-- OH323/L2209 is circuit-busy
-- Hungup 'OH323/L2209'
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/2000-c9fc' status is 
'CONGESTION'
-- Executing Dial("SIP/2000-c9fc", "OH323/h|60|r") in new 
stack
-- H.323 call to h with codec(s) g729
-- Called h
-- Hungup 'OH323/L2210'
== Spawn extension (local, h, 1) exited non-zero on 
'SIP/2000-c9fc'
-- H.323 call 'ip$localhost/2210' cleared, reason 1 (Cleared by 
local user)
My oh323 
configuration:
Configuration of 
OpenH323 channel 
driver--Version: 
0.7.1Listening on address: 0.0.0.0:1720Gatekeeper used: [EMAIL PROTECTED] 
(Registered)FastStart/H245Tunnelling/H245inSetup: OFF/ON/ONSupported 
formats in pref. order: g7290Jitter buffer limits (min/max): 20-100 
msTCP port range: 5000 - 31000UDP (RAS) port range: 5000 - 31000UDP 
(RTP) port range: 1 - 2IP Type-of-Service value: 0User input 
mode: 2Max number of inbound H.323 calls: 10Max number of outbound H.323 
calls: 10Max number of simultaneous H.323 calls: 20Max call rate 
(ingress direction): 1.00/30
What might be the 
problem?
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Re: [Asterisk-Users] :: Strange way of receiving calls ::

2005-04-02 Thread Erwin de Raad

With AMP I managed to set up a group which rings with an incoming POTS
call.  With AMP also, I have also managed to create a Digital Receptionist
BUT this requires caller input, which is not what I need :(

Any help would be appreciated! :) Thanks to you all

With AMP you are limited in what you can do through the GUI.
Try selecting '1' at possible options and use 't' (timeout), So no user
input is required. The phones will ringer AFTER your message has completed.
Your caller hears the ringtone, no music.
I don't know where the default timeout value is defined.

Anything further than that is to be hardcoded in extensions.conf

Take care,
Erwin.

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RE: [Asterisk-Users] Sangoma VS. Digium

2005-04-02 Thread Chris Modesitt


Eric Wrote:
 
 Digium has a hardware echo can?

Not shipping, according to their online store.

Crap!, I spend all my time reading emails from this list, now I have to
check Digium's online store twice a day so I can get my hands on one of
those cards!!

Chris. 

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Re: [Asterisk-Users] Issues with ringing on FXS ports

2005-04-02 Thread Wilson Pickett

 Is there a list of these anywhere?  This is now the third one I've heard
 of, with no documentation:  lowpower (IIRC), robust and now boostringer.
 Do I have to go diving in the source, or is there a Wiki I can't find?

Good point! There is the bugtracker to search but this is one of those
subjects (like the Dial application) that could use a whole chapter
somewhere.

There is a bugfix to change the ring frequency and a recent solution
to a problem I had regarding the cadeces used. I think the onlu
current way would be to search the zaptel stuff on the bugtracker.
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[Asterisk-Users] Registration to multiple GKs

2005-04-02 Thread VoIP Newbie

Hi all,

How can I configure chan_h323 or chan_oh323 to register to multiple GK
and route calls in-between?

Many thanks.
Newbie
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Re: [Asterisk-Users] Call bridging

2005-04-02 Thread Wilson Pickett

 How do I dialout using an extensions.conf and connect to an outside number?
 For example, I would like for a person in the IVR to be able to press a
 number and it dial out using another FXO card and another POTS line and 
 then bridge the two calls together.

So you've given up on trying to use three-way calling? In that case
the original proposed solution in the first thread of using
Dial(ZAP/2) seems like it would work.
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[Asterisk-Users] SIP register more then 1 with same username

2005-04-02 Thread Bashir Ullah - www.Lamsre.Com

Hi all * user

I did connected with * from 2  sip-softphone and i registered with asterisk
under same username and password and working both fine. but * shows only
one.

is there any way to find them both by using any tips.

Bashir

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Re: [Asterisk-Users] :: Strange way of receiving calls ::

2005-04-02 Thread Wilson Pickett

 = An incoming call is received on any of the incoming POTS ZAP channels 
 = Call is immediately picked up by Asterisk and the caller is greeted with a
 message like, Thank you for calling , please hold the line your call
 will be attended to shortlyMUSIC 
 = What I would like to achieve is that as soon as the call is immediately
 picked up by Asterisk, a group of phones starts ringing, so that effectively
 any person in the group can attend to the caller, and as soon as this is
 done, the caller stops hearing the message and can speak to a
 representative. 

If I understand this question (and I'm not 100% sure I do) :

Playback(message)
Dial(${phones_to_ring},45,m) 

If message is 3 seconds long, that would be the only delay.
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[Asterisk-Users] {extensions.conf} Dialing plans with queues....





Hi all,

This is the situation:
 I have a call coming in from the POTS line and I pass this through
to the [incoming] section by including the [incoming] inside [sip].
 Then s,... starts. It picks up the call and then places the call in
a "Reception" queue.

This is the problem:
 When the call is in the "Reception" queue, I would like to play a
voice menu while the call is in the queue.
 If the user responds to the "Voice Menu" by dialling a number, then
I would like to pass this call to that extension.
 I know you can put a time out on the "Reception" queue - but I need
to give the caller the option to by-pass the 
 Receptionist.


/extensions.conf/

[incoming]
exten = s,1,Wait,1 ;
Wait a second, just for fun
exten = s,2,Answer ;
Answer the line
;___TODO__ ;Play a "Thank you for
calling ..."
exten = s,3,Queue(QUEUE-Reception) ;Place call in
reception queue
;___TODO__ ;Play "Voice Menu -
Sales;Accounts;Support
exten = 1,1,QUEUE(QUEUE-Support) ;Pressed "1", place
call in queue.conf::[QUEUE-Support] 
need some help ova here plz

[sip]
include = incoming
;include the incoming calls context

exten = 101,1,Dial(SIP/Reception,20,tr)
exten = 200,1,Queue(QUEUE-Support)


/Asterisk Out-put/

Asterisk Ready.
 -- Starting simple switch on 'Zap/4-1'
 -- Executing Wait("Zap/4-1", "1") in new stack
 -- Executing Answer("Zap/4-1", "") in new stack
 -- Executing Queue("Zap/4-1", "QUEUE-Reception") in new stack
 -- Started music on hold, class 'default', on Zap/4-1
 -- Called SIP/Reception
 -- Stopped music on hold on Zap/4-1
Apr 2 13:51:16 WARNING[1994]: chan_sip.c:860 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
Apr 2 13:51:16 NOTICE[2004]: app_queue.c:1103 wait_for_answer: No one
is answering queue 'QUEUE-Reception' (1/0/0)



Thank you all.
-- 
Kind Regards
Etienne




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Re: [Asterisk-Users] Are there online forums instead of this emailforum??

2005-04-02 Thread Bob Goddard

On Thursday 31 March 2005 23:11, Tim Bass wrote:
 The UNIX Forums have over 28 thousand registered users. I have many
 years of experience in both email lists and on line forums and I can tell
 you without a doubt that on-line forums are far superior to email lists.
 There is no comparison.

28 thousand registered users? That is a meaningless statistic. How many
of them have accessed the system in the last month? I'll hazard an
educated guess and say it is insignificant compared to comp.unix.*
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[Asterisk-Users] astcc problems

2005-04-02 Thread wassim darwish

i have downloaded astcc and confiugured it on web but
the problem is when a call comes by the right callerid
it gives me on CLI like this:

-- Executing DeadAGI(Zap/1-1,
astcc.agi|01475969|s) in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/astcc.agi
Detected dry run!
AGI Environment Dump:
 -- accountcode =
 -- callerid = 01475969
 -- calleridname = unknown
 -- channel = Zap/1-1
 -- context = incoming
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = s
 -- language = en
 -- priority = 3
 -- rdnis = unknown
 -- request = astcc.agi
 -- type = Zap
 -- uniqueid = 1112430048.4
Apr  2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-tone does not exist in
any format
Res is
Silent Level is
Card no is 12345

Card has face value 3 and has used 0


3 dollars and 0 cents remain
Apr  2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-youhave does not exist
in any format
-- Playing 'digits/3' (language 'en')
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-dollars does not exist
in any format
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-remaining does not
exist in any format
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-badphone does not
exist in any format
-- AGI Script astcc.agi completed, returning 0

I dont know what the problem and what this warnings
mean and how can i fix them please help.
and thanks
 



__ 
Do you Yahoo!? 
Make Yahoo! your home page 
http://www.yahoo.com/r/hs
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[Asterisk-Users] Little question

Can I use SJphone like a H.323 phone in order to dial and receive call
through Asterisk?Can I consider Asterisk just like a sort of H323
Gateway?

Thanks 4 all!

flx
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Re: [Asterisk-Users] {extensions.conf} Dialing plans with queues....

Oh well, found it with some searching
Add the menu as a context in queue.conf and then define that menu in 
extensions.conf.
Mind only single digit numbers are valid.

Kind Regards
Etienne
Etienne Pretorius wrote:
Hi all,
This is the situation:
I have a call coming in from the POTS line and I pass this through 
to the [incoming] section by including the [incoming] inside [sip].
Then s,... starts. It picks up the call and then places the call 
in a Reception queue.

This is the problem:
When the call is in the Reception queue, I would like to play a 
voice menu while the call is in the queue.
If the user responds to the Voice Menu by dialling a number, 
then I would like to pass this call to that extension.
I know you can put a time out on the Reception queue - but I 
need to give the caller the option to by-pass the
Receptionist.

*/extensions.conf/*
[incoming]
exten = s,1,Wait,1 ; Wait 
a second, just for fun
exten = s,2,Answer ; 
Answer the line
;___TODO__  ;Play a Thank you for 
calling ...
exten = s,3,Queue(QUEUE-Reception)  ;Place call in 
reception queue
;___TODO__  ;Play Voice Menu - 
Sales;Accounts;Support
exten = 1,1,QUEUE(QUEUE-Support)  ;Pressed 1, place 
call in queue.conf::[QUEUE-Support]need some help ova 
here plz

[sip]
include = incoming 
;include the incoming calls context

exten = 101,1,Dial(SIP/Reception,20,tr)
exten = 200,1,Queue(QUEUE-Support)
*/Asterisk Out-put/*
Asterisk Ready.
-- Starting simple switch on 'Zap/4-1'
-- Executing Wait(Zap/4-1, 1) in new stack
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing Queue(Zap/4-1, QUEUE-Reception) in new stack
-- Started music on hold, class 'default', on Zap/4-1
-- Called SIP/Reception
-- Stopped music on hold on Zap/4-1
Apr  2 13:51:16 WARNING[1994]: chan_sip.c:860 retrans_pkt: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Critical Request)
Apr  2 13:51:16 NOTICE[2004]: app_queue.c:1103 wait_for_answer: No one 
is answering queue 'QUEUE-Reception' (1/0/0)


Thank you all.
--
Kind Regards
Etienne
 


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Re: [Asterisk-Users] VoIP Provider problems

2005-04-02 Thread Bob Goddard

On Friday 01 April 2005 04:28, Joseph Gutowski wrote:
 Ok, since I guess no one else wanted to bite -- I will.

 I installed PingPlotter, switched to UDP just to be the same as you,
 and ran it against sip.broadvoice.com. Absolutley no problems, no
 packet loss at all.

 Ran it with all of the published proxy addresses, again no problems.

 I then used the 63.251.209.126 that you posted, and it was awful (at
 least it appears awful). I have reliable 20% packet loss at each of
 two Verio hops (nothing lost at the far end).

Don't take this the wrong way, but you are showing a bit of
ignorance about how TCP/IP works.

The apparent packet loss you are seeing may be just fine tuning
of the routers in question.

The routers may be set up not to send ICMP host/network
unreachables back to the originating system if they are
required to send more than one in a configured time period.

Routers have better things to do than continually tell you
that a host is unreachable.


B
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Re: [Asterisk-Users] Queues

2005-04-02 Thread David Choo

Hi Henry,

Thanks for the advise. I'll check that out.

Best Regards,

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Engineered for Changing Businesses
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Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
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   Subject 
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 After calls come in, it works fine, however, I notice that even when
 SIP/602 is on the phone, Asterisk will still ring her. I believe its due
 to
 the fact that the phone support call-waiting. Is there anyway that I can
 disable this support only on queues and ring the next extension in this
 case, which is SIP/603?

You can use setgroup/checkgroup combo I think.  What kind of phone is
SIP/602.  Most phone have a config to disable call waiting.

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Re: [Asterisk-Users] Preserve g729 registration over reinstall??

2005-04-02 Thread Martijn van Oosterhout

On Thu, Mar 31, 2005 at 01:43:41PM -0800, Mike Matthews wrote:
 I purchased the g729 codec from Digium.  Every time I reinstall
 Asterisk (or Linux) I naturally lose the registration. Digium only
 allows one reinstallation without calling them which is a nusance for
 both them and me. Is there any way to preserve the registration
 across a reinstall?  Perhaps by backing up a directory or a file? Any
 help appreciated.

Check the archives to be sure, but I beleive the relevent files are
stored in /var/lib/asterisk/licences or something like that.

Good luck.
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd
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[Asterisk-Users] TDM04B - TDM PCI MASTER ABORT

2005-04-02 Thread JOAO CARLOS MOURA

Hi all
I have four cards TDM04B with modules FXO.
I am using the stabled version of the Asterisk in a Pentium IV with 1GB of 
RAM and RedHat 9
When I load the modules zaptel and wctdm, I receive the message: TDM PCI 
Master Abort.
How I decide this problem?

Thank's
Joao Carlos Moura 

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Re: [Asterisk-Users] VoIP Provider problems

2005-04-02 Thread Johnathan Corgan

Bob Goddard wrote:
The apparent packet loss you are seeing may be just fine tuning
of the routers in question.
This is the conclusion I came to as well; however, with the way 
PingPlotter works the router is not sending ICMP unreachables but rather 
ICMP TTL expired responses.  In any case, the routers in question may 
either be:

1) ...intentionally discarding the received UDP ping packets (these 
are not ICMP pings, but rather UDP packets with TTL down to zero when 
they get to the router), because the router has better things to do.

2) ...throttling the ICMP TTL expired responses to a certain rate per 
period of time, as you suggest.  This would appear as packet loss.

3) ...actually congested, with the received UDP pings (and other types 
of packets) getting discarded on the input side at the rate shown in the 
data.

I wish there was a way to measure 3) without being affected by 1) and 2).
I agree then, that PingPlotter is not a highly accurate way to measure 
path quality.  Still, though, looking over the data for a couple days 
now it is easy to see cyclical patterns that go from 1% to 30% 
(PingPlotter measured) loss, and an easily seen correlation with the 
voice quality of my outbound Broadvoice calls.

Interestingly enough, switching from a Firefly soft phone on my 
workstation, using IAX2/ulaw, to an analog phone-TDM400 FXS port right 
at the Asterisk server has made a big difference.  So some of the 
perceived crappiness was in the soft phone-Asterisk path and was 
probably being exacerbated by the network loss on the net or at 
Broadvoice's router.

-Johnathan
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RE: [Asterisk-Users] H.323 call '.....' cleared,reason 8 (Transport failure)

2005-04-02 Thread Alex Vishnev









Cenk,



Are you sure that remote will handle H245
tunneling? If the remote does not know how to do that, you will get transport
failure. I would suggest doing FastStart instead and
see if you are getting the same results. Of course, you can verify that the
remote can handle faststart as well.



Alex











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas
Sent: Saturday, April 02, 2005
6:20 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] H.323
call '.' cleared,reason 8 (Transport failure)





I
installed the oh323 channel driver and registered to the gate keeper
succesfully.

I come
through the GK, ring the dialed number forabout 0.5 seconds
andloose the line.I contacted the GKand they report that they
receive the correct dialstring to route the call but the call is ended by my
side.

The
dialstring looks like this:

exten
= _.,1,Dial(OH323/${EXTEN},60,r)

I use the
following channel driver:

asterisk-oh323-0.7.1

openh323-Janus_patch4-src

pwlib-Janus_patch4-src

and the
message on asterisk console looks like this:

--
Registered with gatekeeper '[EMAIL PROTECTED]'.

--
Executing Dial(SIP/2000-c9fc, OH323/0012029361212|60|r)
in new stack

-- H.323
call to 0012029361212 with codec(s) g729

-- Called
0012029361212

-- H.323
call 'ip$localhost/2209' cleared, reason 8 (Transport failure)

--
OH323/L2209 is circuit-busy

-- Hungup
'OH323/L2209'

==
Everyone is busy/congested at this time (1:0/1/0)

== Auto
fallthrough, channel 'SIP/2000-c9fc' status is 'CONGESTION'

--
Executing Dial(SIP/2000-c9fc, OH323/h|60|r) in new
stack

-- H.323
call to h with codec(s) g729

-- Called
h

-- Hungup
'OH323/L2210'

== Spawn
extension (local, h, 1) exited non-zero on 'SIP/2000-c9fc'

-- H.323
call 'ip$localhost/2210' cleared, reason 1 (Cleared by local user)

My oh323
configuration:

Configuration
of OpenH323 channel driver
--
Version: 0.7.1
Listening on address: 0.0.0.0:1720
Gatekeeper used: [EMAIL PROTECTED]
(Registered)
FastStart/H245Tunnelling/H245inSetup: OFF/ON/ON
Supported formats in pref. order: g7290
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 5000 - 31000
UDP (RAS) port range: 5000 - 31000
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: 2
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 20
Max call rate (ingress direction): 1.00/30



What might be the problem?








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Re: [Asterisk-Users] Shorewall firewall rules

2005-04-02 Thread Rich Adamson

 I'm trying to get firewalling working but I am clueless as to which ports 
 I need to open, I keep opening more ports and it's not working :(
 
 Basically I want SIP and IAX2 to work. IAX2 works fine, but SIP is giving 
 me a headache. It seems that the stateless firewall is not able to handle 
 SIP. I'm using shorewall as my firewall with these rules:
 
 ACCEPT  netfwudp 4569
 ACCEPT  fw net   udp 4569,5060,1:2
 
 My rtp.conf says this:
 rtpstart=1
 rtpend=2
 
Others have already commented on the above. Here's a couple more items
to think about.

The udp ports required for rtp varies by sip phone vendor. In other
words, the exact ports required are not necessarily those shown above.
It also makes a difference as to which device initiates the first rtp
transmission. As noted, the rtpstart and rtpend are for asterisk only,
and are used as its source port when communicating with an exernal
sip device (phone or another asterisk).

If you look at the Xten documentation, you'll find that soft phone
uses rtp udp ports in the low 8000 range.

If you look at the Cisco 7960's, you'll find they use 16384 to 32768,
and those particular values can be seen/changed in SIPDefault.cnf file.

The exact rtp port to be used by each sip device never became a
standard in the rfc, so each vendor is allowed to chose whatever
udp port range they felt like using as their default.

Opening udp ports from 1024 to 64000 will likely help, but you might
as well dump the firewall if you're going to open everything like
that.

Also note that each line/conversation will use another udp port.
So, in the case of the xten product, the first line/conversation
may use port 8000. If you put that line on hold and start another
(second) rtp session, that line/conversation will use something
like 8002 (or whatever).

Use something like ethereal to sniff the packets on the outside of
your firewall, and you'll see the exact udp ports used for whatever
device you're trying to communicate with.

If you don't feel like implementing ethereal, then open all the ports
as someone suggested, then do a netstat -an during a real sip call,
and you'll see the exact udp ports in use. Once you're comfortable
with your understanding of what ports are actually used, then
adjust your firewall to support those ports.


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RE: [Asterisk-Users] astcc problems

2005-04-02 Thread Karl H. Putz

There is an error in the ASTCC makefile.  It places the sound files in the
wrong directory.

You can either modify the Makefile and change the line:

SOUNDSDIR=/usr/share/asterisk/sounds

to

SOUNDSDIR=/var/lib/asterisk/sounds

and then re-install.  Or simply move the files from the improper directory
to the correct one.


Karl Putz

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of wassim
darwish
Sent: Saturday, April 02, 2005 7:27 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] astcc problems


i have downloaded astcc and confiugured it on web but
the problem is when a call comes by the right callerid
it gives me on CLI like this:

-- Executing DeadAGI(Zap/1-1,
astcc.agi|01475969|s) in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/astcc.agi
Detected dry run!
AGI Environment Dump:
 -- accountcode =
 -- callerid = 01475969
 -- calleridname = unknown
 -- channel = Zap/1-1
 -- context = incoming
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = s
 -- language = en
 -- priority = 3
 -- rdnis = unknown
 -- request = astcc.agi
 -- type = Zap
 -- uniqueid = 1112430048.4
Apr  2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-tone does not exist in
any format
Res is
Silent Level is
Card no is 12345

Card has face value 3 and has used 0


3 dollars and 0 cents remain
Apr  2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-youhave does not exist
in any format
-- Playing 'digits/3' (language 'en')
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-dollars does not exist
in any format
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-remaining does not
exist in any format
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-badphone does not
exist in any format
-- AGI Script astcc.agi completed, returning 0

I dont know what the problem and what this warnings
mean and how can i fix them please help.
and thanks




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Re: [Asterisk-Users] VoIP Provider problems

2005-04-02 Thread Rich Adamson

  The apparent packet loss you are seeing may be just fine tuning
  of the routers in question.
 
 This is the conclusion I came to as well; however, with the way 
 PingPlotter works the router is not sending ICMP unreachables but rather 
 ICMP TTL expired responses.  In any case, the routers in question may 
 either be:
 
 1) ...intentionally discarding the received UDP ping packets (these 
 are not ICMP pings, but rather UDP packets with TTL down to zero when 
 they get to the router), because the router has better things to do.
 
 2) ...throttling the ICMP TTL expired responses to a certain rate per 
 period of time, as you suggest.  This would appear as packet loss.
 
 3) ...actually congested, with the received UDP pings (and other types 
 of packets) getting discarded on the input side at the rate shown in the 
 data.
 
 I wish there was a way to measure 3) without being affected by 1) and 2).

The deceptive part of doing the above is that once you see
congestion (lack of an icmp response), you still have absolutely
no idea what device was at fault.

In other words, as the ttl value is increased and additional icmps
are sent, you might see what you believe is congestion, but you still
don't have any clue as to whether hop #2, #5, or #10 actually was
involved with that congestion.


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RE: [Asterisk-Users] Registration to multiple GKs

2005-04-02 Thread Alex Vishnev

I don't think you can. The rules of h323 is so that you can register with a
single gk at a time.

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie
Sent: Saturday, April 02, 2005 6:37 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Registration to multiple GKs

Hi all,

How can I configure chan_h323 or chan_oh323 to register to multiple GK
and route calls in-between?

Many thanks.
Newbie
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[Asterisk-Users] Two accounts at one provider and a 302 redirect problem

2005-04-02 Thread Christian Peter

Hi,

I've got a problem with my incoming calls (SIP). First I tried to route
different providers to different extensions in which ._ matched the call
and called the internal phones and so on.

Then I got this Nikotel Account. I managed to get it working. Small hint
for the people trying Nikotel and having problems with the internal 99er
numbers. Nikotel redirects with the 302 message to a user called
user at 63.214.186.6. Asterisk can't redirect unless you define a sip.conf
section with [63.214.186.6]. See
http://bugs.digium.com/bug_view_page.php?bug_id=0001974 and read
marksters comment at the bottom.

So now it worked.

Then I tried two Nikotel Accounts on the same asterisk machine. Problem
is that now every internal 99er call goes to the extension defined in
[63.214.186.6]. My idea is now to route every incoming call to one
extension. I tried to recognize the called number int the extension like
this:

9978389389,1,Answer
9978389389,2,...ring phone 1

9948389390,1,Answer
9948389390,2,...ring phone 2

(these are fake numbers)

But I did not manage to route the call based on the called numbers. Only

_.,1,Answer
_.,2,...ring a phone

did work.

Does anyone have a hint? I would appreciate any comments!
Thanks in advance

Christian Peter

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Re: [Asterisk-Users] Are there online forums instead of this email


YE HAA.. We 'be out har' in rural internet space, mamma
Stepping in cow pats .   Way out past the fancy, hi falutin',
city-slicker, collaborative tools of 2005   ..
 

You babble all this nonsense, and *YOU ARE* the guy advocating moderation?
BWAHAHAHAHAHAHAHAHAAA!
Now please go back to your superior women-and-students web forum world 
and leave us in peace.

--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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Re: [Asterisk-Users] Two accounts at one provider and a 302 redirect problem

2005-04-02 Thread Rich Adamson

 I've got a problem with my incoming calls (SIP). First I tried to route
 different providers to different extensions in which ._ matched the call
 and called the internal phones and so on.
 
 Then I got this Nikotel Account. I managed to get it working. Small hint
 for the people trying Nikotel and having problems with the internal 99er
 numbers. Nikotel redirects with the 302 message to a user called
 user at 63.214.186.6. Asterisk can't redirect unless you define a sip.conf
 section with [63.214.186.6]. See
 http://bugs.digium.com/bug_view_page.php?bug_id=0001974 and read
 marksters comment at the bottom.
 
 So now it worked.
 
 Then I tried two Nikotel Accounts on the same asterisk machine. Problem
 is that now every internal 99er call goes to the extension defined in
 [63.214.186.6]. My idea is now to route every incoming call to one
 extension. I tried to recognize the called number int the extension like
 this:
 
 9978389389,1,Answer
 9978389389,2,...ring phone 1
 
 9948389390,1,Answer
 9948389390,2,...ring phone 2
 
 (these are fake numbers)
 
 But I did not manage to route the call based on the called numbers. Only
 
 _.,1,Answer
 _.,2,...ring a phone
 
 did work.
 
 Does anyone have a hint? I would appreciate any comments!

Not enough info to guess with any reasonableness.

Assuming you are using a register statement for each account, that
statement should look something like:
 register=myuserid:[EMAIL PROTECTED]/1234

The 1234 at the end of that statement tells your provider what
digits to dial when contacting your asterisk. So, in this example,
an entry in extensions.conf like:
 exten = 1234,1,Dial(SIP/3000,15,r)
would cause that incoming call to ring sip phone x3000. For 

If you use a different suffix on each provider's register statement,
you should be able to put together an associated extensions.conf
entry to handle each separately.

On the other hand, if you use a register statement like:
 register=myuserid:[EMAIL PROTECTED]
without the suffix, then an extensions.conf entry like:
 exten = s,1,Dial(SIP/3000,15,r)
would work, but it doesn't distinguish between multiple providers.

If you need a better answer, then post your register statements in
sip.conf (change the passwords) for each provider, along with the
appropriate sections of extensions.conf that handle the incoming
calls.


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[Asterisk-Users] Asterisk Voice mail with CCM

2005-04-02 Thread Nathan Reeves

Anyone running Cisco Call Manager and using Asterisk for voice mail
services?  Things working well or is the concept a bit of a hassle to
implement?

TIA
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Re: [Asterisk-Users] VoIP Provider problems

2005-04-02 Thread Johnathan Corgan

Rich Adamson wrote:
In other words, as the ttl value is increased and additional icmps
are sent, you might see what you believe is congestion, but you still
don't have any clue as to whether hop #2, #5, or #10 actually was
involved with that congestion.
Sure.  But there is a way around this.
The traceroute-style statistics gathering technique that PingPlotter 
uses tries all the hops at the same time and plots the return rate for 
each one.  So a 10 hop path has 10 packets go out, with individual 
packet's TTL set to expire at each hop.  Done over and over again and 
averaged over many probes, you get a very clear picture.  Packet loss at 
one node affects all the probes to that node and further ones, resulting 
in an increasing loss rate as you go down the path. For example:

Hop Loss
1   0%
2   1%
3   1%
4   5%
5   5%
6   6%
7   15%
8   15%
9   16%
10  16%
It's easy to see there is a big problem between hops 6 and 7 and a 
smaller problem between hops 3 and 4.

With the broadvoice router I was seeing (at first) a jump from 0% to 9% 
at my local ISP, then small increments over the next 10 hops until it 
was at about 14%, then a big jump to 29% at the last hop.

The data has varied cyclically between as high as the above and as low 
as 1% all the way across.  Right this very moment, it is 2% within my 
ISP, still 2% all the way to PNAP, then a jump to 14% at the broadvoice 
ingress router at PNAP.

Again, temper the above with the fact that the packet loss may be 
intentional, and these statistics not representative of real RTP 
traffic, as per my previous message.  But I can predict with high 
accuracy what the caller on the other end of my broadvoice call will say 
about my voice quality based on the last number I see for the broadvoice 
ingress router.

-Johnathan
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[Asterisk-Users] Starting with Asterisk-SIP

2005-04-02 Thread ruben cuevas rumin

Hi all,

I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.

I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.

I have read the documentation, but It's so much big and I don't know
what I have to do.

Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.

It would be wonderful if someone could help me.

Thanks in advance.

Best Regards,

  Rubén.
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[Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Hi All * users...
Question:
   In extensions.conf - I am awaire that you can use macro's but what I 
am wondering about.. is that can you create a macro to do dynamic Zap 
channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for Out 
break calls, so i was just wandering if some1 could help me a bit over 
here.

[outgoing]  ;Dial 
0 on the phone for external line

exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically 
allocated to Zap/4 needs to be dynamic
exten = _0,2,Goto(102)
exten = _0,102,Congestion
exten = _0,103,Hangup

I'll apreciate any help in this regard.
--
Kind Regards
Etienne

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Re: [Asterisk-Users] Starting with Asterisk-SIP

2005-04-02 Thread Mike

On Sat, 2 Apr 2005, ruben cuevas rumin wrote:
Hi all,
I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.
I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.
I have read the documentation, but It's so much big and I don't know
what I have to do.
Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.
It would be wonderful if someone could help me.
Thanks in advance.
Best Regards,
 Rubén.
Do this:
in
/usr/src/asterisk
run make samples
now you will have the sample configs in /etc/asterisk
then take a look at voip-info.org
Michael


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Re: [Asterisk-Users] Starting with Asterisk-SIP


I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.
 

You should go and read some docs:
http://www.digium.com/handbook-draft.pdf
I have read the documentation, but It's so much big and I don't know
what I have to do.
 

The handbook isn't that big. You should be fine.
Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.
 

Well I can't tell you what to do but I can tell you the following:
- Entries sip.conf, iax.conf have nothing to do with contexts (they 
refer to them though, through context = blah directives
)
- Entries in extensions.conf are all contexts.

This was the only thing I had to realize before I could use the software.
It would be wonderful if someone could help me.
 

Unfortunately, I can't do for you the research work you're supposed to 
be doing. But do take a look at voip-info.org. There's a lot of good 
stuff there - it's a wonderful resource and will help you get going.

Cheers,
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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RE: [Asterisk-Users] Starting with Asterisk-SIP

2005-04-02 Thread Kerry Garrison

If you ar really lost I would suggest getting [EMAIL PROTECTED] and getting your
system up and running fairly easily. Then you can look at the config files
in case you want to do it yourself later.

Kerry Garrison
http://www.geekgazette.com
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ruben cuevas
rumin
Sent: Saturday, April 02, 2005 8:24 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Starting with Asterisk-SIP

Hi all,

I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and the
SIP clients.

I think I can use Asterisk for this issue. I have installed it and I have
run it, but I don't know how I have to configure it.

I have read the documentation, but It's so much big and I don't know what I
have to do.

Someone could tell me what configuration files have I to use, and what have
I to put in this files?. If is it posible, I would like someone send me some
simple examples of this files.

It would be wonderful if someone could help me.

Thanks in advance.

Best Regards,

  Rubén.
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Never Mind. oops.
I just needed to play around with some syntax.
Zap/1,2,3,4/$EXTEN
Ps: Is there a better santax because 1-4 doesn't work.
Kind Regards
Etienne
Etienne Pretorius wrote:
Hi All * users...
Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for Out 
break calls, so i was just wandering if some1 could help me a bit 
over here.

[outgoing]  ;Dial 
0 on the phone for external line

exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically 
allocated to Zap/4 needs to be dynamic
exten = _0,2,Goto(102)
exten = _0,102,Congestion
exten = _0,103,Hangup

I'll apreciate any help in this regard.
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Nope - I jumped to conclusions.
It just tries channel 1 the whole time.
Any ideas any1
Kind Regards
Etienne

Etienne Pretorius wrote:
Never Mind. oops.
I just needed to play around with some syntax.
Zap/1,2,3,4/$EXTEN
Ps: Is there a better santax because 1-4 doesn't work.
Kind Regards
Etienne
Etienne Pretorius wrote:
Hi All * users...
Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for 
Out break calls, so i was just wandering if some1 could help me a 
bit over here.

[outgoing]  ;Dial 
0 on the phone for external line

exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically 
allocated to Zap/4 needs to be dynamic
exten = _0,2,Goto(102)
exten = _0,102,Congestion
exten = _0,103,Hangup

I'll apreciate any help in this regard.
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

2005-04-02 Thread Steven Critchfield

On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote:
 Never Mind. oops.
 
 I just needed to play around with some syntax.
 
 Zap/1,2,3,4/$EXTEN
 
 Ps: Is there a better santax because 1-4 doesn't work.

Look at groups in the /etc/asterisk/zaptel.conf

Once you define your groups, you can just 
exten = _0,1,Dial(Zap/g1/$EXTEN)
And asterisk will pick some available channel out of the channels
defined in group 1 to use for dialing out.

  Hi All * users...
 
  Question:
 In extensions.conf - I am awaire that you can use macro's but what 
  I am wondering about.. is that can you create a macro to do dynamic 
  Zap channel allocation for a out going call?
 I don't want to reserve a channel/port in the TDM400P card for Out 
  break calls, so i was just wandering if some1 could help me a bit 
  over here.
 
  [outgoing]  ;Dial 
  0 on the phone for external line
 
  exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically 
  allocated to Zap/4 needs to be dynamic
  exten = _0,2,Goto(102)
  exten = _0,102,Congestion
  exten = _0,103,Hangup
 
  I'll apreciate any help in this regard.
 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Starting with Asterisk-SIP

2005-04-02 Thread Jeff Heath

This may not work for you on a student budget unless you're willing to
cut into your beer budget ... :-)

I recently got the book and cd from Signate and got a system up pretty
quickly that can make calls back and forth between a couple of SIP
phones directly attached to the server.  

take a look at www.signate.com

Jeff Heath


On Sat, 2005-04-02 at 11:24, ruben cuevas rumin wrote:
 Hi all,
 
 I'm a Telecomunication Engeenering student. I have to develop a VoIP
 apliccation using SIP protocol. I have to develop the SIP Server, and
 the SIP clients.
 
 I think I can use Asterisk for this issue. I have installed it and I
 have run it, but I don't know how I have to configure it.
 
 I have read the documentation, but It's so much big and I don't know
 what I have to do.
 
 Someone could tell me what configuration files have I to use, and what
 have I to put in this files?. If is it posible, I would like someone
 send me some simple examples of this files.
 
 It would be wonderful if someone could help me.
 
 Thanks in advance.
 
 Best Regards,
 
   Rubén.
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

2005-04-02 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Etienne Pretorius wrote:
 Nope - I jumped to conclusions.
 
 It just tries channel 1 the whole time.
 
 Any ideas any1
 
 Kind Regards
 Etienne
 
 
 
 Etienne Pretorius wrote:
 
 Never Mind. oops.

 I just needed to play around with some syntax.

 Zap/1,2,3,4/$EXTEN

 Ps: Is there a better santax because 1-4 doesn't work.

 Kind Regards
 Etienne


 Etienne Pretorius wrote:

 Hi All * users...

 Question:
In extensions.conf - I am awaire that you can use macro's but what
 I am wondering about.. is that can you create a macro to do dynamic
 Zap channel allocation for a out going call?
I don't want to reserve a channel/port in the TDM400P card for
 Out break calls, so i was just wandering if some1 could help me a
 bit over here.

 [outgoing]  ;Dial
 0 on the phone for external line

 exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically
 allocated to Zap/4 needs to be dynamic
 exten = _0,2,Goto(102)
 exten = _0,102,Congestion
 exten = _0,103,Hangup

 I'll apreciate any help in this regard.


Allocate the channels to a group in zaptel.conf (group=1)

then dial with Zap/g1 which will take the lowets available channel.

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?





Thank you very much, that sorted out the problem.
Kind Regards
Etienne


Steven Critchfield wrote:

  On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote:
  
  
Never Mind. oops.

I just needed to play around with some syntax.

Zap/1,2,3,4/$EXTEN

Ps: Is there a better santax because 1-4 doesn't work.

  
  
Look at groups in the /etc/asterisk/zaptel.conf

Once you define your groups, you can just 
exten = _0,1,Dial(Zap/g1/$EXTEN)
And asterisk will pick some available channel out of the channels
defined in group 1 to use for dialing out.

  
  

  Hi All * users...

Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for "Out 
break" calls, so i was just wandering if some1 could help me a bit 
over here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically 
allocated to Zap/4 needs to be dynamic
exten = _0,2,Goto(102)
exten = _0,102,Congestion
exten = _0,103,Hangup

I'll apreciate any help in this regard.

  

  



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Re: [Asterisk-Users] Preserve g729 registration over reinstall??

2005-04-02 Thread Mike Matthews

For the record here, I am quoting from an email from Digium on the subject:

"You will need to backup /var/lib/asterisk/licenses. You will also needto backup the codec_g729 and format_g729 in your/usr/lib/asterisk/modules/ directory. The ethernet cards in yourmachine cannot be changed. Otherwise you will have to reregister yourcodec.
If required you may reregister your codec. If you run into any problemsreregistering, we will assist you on with that problem.
Please refer to http://www.digium.com/index.php?menu=asterisk_g729 foradditional instructions."

Thanks to Digium Support for the prompt and thorough response.Mike Matthews [EMAIL PROTECTED] wrote:

I purchased the g729 codec from Digium. Every time I reinstall Asterisk (or Linux) I naturally lose the registration. Digium only allows one reinstallation without calling them which is a nusance for both them and me. Is there any way to preserve the registration across a reinstall? Perhaps by backing up a directory or a file? Any help appreciated.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users"Put down that coffee...coffee is for Closers!"Phone: 918-770-4503Fax: 206-666-1720email: [EMAIL PROTECTED]sip: [EMAIL PROTECTED]___
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Re: [Asterisk-Users] Re: Are there online forums instead of this email


You are the bully. So far the majority wish the email list to
continue and yet you still continue to demand that Digium
convert to a forum.
 

Looks like M. Bass likes to troll about mailing lists, see this post:
http://info.ccone.at/INFO/Mail-Archives/procmail/Feb-2003/msg00230.html
As usual, the pro-censorship moralizing zealots don't apply their own 
rules to themselves. Nothing new here...

Best Regards,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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[Asterisk-Users] Re: Starting with Asterisk-SIP

At the URL http://www.voip-info.org may find some examples.

Gettin' started
First of all you must define a possible dialplan that you can
configure in the file extensions.conf. Dialplan may include several
options, just like a simple comunication between two softphone(for
example Sjphone) using SIP through the Asterisk PBX.
After this, you must define setting about the other configuration
files (.conf, like sip.conf.. etc..)related to the dialplan defined..
and so on...

However you must easily find several interesting examples over
Internet if you search them^_^

I am an Electronic Engineer student too ^_^

bye
flx


On Sat, 2 Apr 2005 18:24:17 +0200, ruben cuevas rumin
[EMAIL PROTECTED] wrote:
 Hi all,
 
 I'm a Telecomunication Engeenering student. I have to develop a VoIP
 apliccation using SIP protocol. I have to develop the SIP Server, and
 the SIP clients.
 
 I think I can use Asterisk for this issue. I have installed it and I
 have run it, but I don't know how I have to configure it.
 
 I have read the documentation, but It's so much big and I don't know
 what I have to do.
 
 Someone could tell me what configuration files have I to use, and what
 have I to put in this files?. If is it posible, I would like someone
 send me some simple examples of this files.
 
 It would be wonderful if someone could help me.
 
 Thanks in advance.
 
 Best Regards,
 
   Rubén.
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RE: [Asterisk-Users] Starting with Asterisk-SIP

2005-04-02 Thread Damon Estep

Start here to get it running, plan on burning a couple days playing with it. 
There is no fast way to get comfortable with it other than hands on and 
research on the list and wiki.

http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ruben cuevas 
rumin
Sent: Saturday, April 02, 2005 9:24 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Starting with Asterisk-SIP

Hi all,

I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.

I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.

I have read the documentation, but It's so much big and I don't know
what I have to do.

Someone could tell me what configuration files have I to use, and what
have I to put in this files?. If is it posible, I would like someone
send me some simple examples of this files.

It would be wonderful if someone could help me.

Thanks in advance.

Best Regards,

  Rubén.
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[Asterisk-Users] SjPhoneH323

 Can I use SJphone like a H.323 phone in order to dial and receive call
through Asterisk?Can I consider Asterisk just like a sort of H323
Gateway?

Thanks 4 all!

flx
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Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?





But what happens when the channel is busy etc?

It does not seem to drop the "Attempt a naitive bridge". I saw this in
the commented out section and thought that it'll work but well it still
hangs at "Attempting a naitive bridge".

[outgoing] ;Dial
"0" on the phone for external line

exten = _0,1,Dial(Zap/g2/$EXTEN)
exten = _0,2,Goto(_0-${DIALSTATUS},1)
;Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten = _0-ANSWER,1,Goto(_0,102)
exten = _0-.,1,Goto(_0,1) ;Try
another line

exten = _0,102,Congestion
exten = _0,103,Hangup


Hints?
Kind Regards
Etienne

Technical Support
Kingsley Technologies



Etienne Pretorius wrote:

  
Thank you very much, that sorted out the problem.
  Kind Regards
Etienne
  
  
Steven Critchfield wrote:
  
On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote:
  

  Never Mind. oops.

I just needed to play around with some syntax.

Zap/1,2,3,4/$EXTEN

Ps: Is there a better santax because 1-4 doesn't work.



Look at groups in the /etc/asterisk/zaptel.conf

Once you define your groups, you can just 
exten = _0,1,Dial(Zap/g1/$EXTEN)
And asterisk will pick some available channel out of the channels
defined in group 1 to use for dialing out.

  

  
Hi All * users...

Question:
   In extensions.conf - I am awaire that you can use macro's but what 
I am wondering about.. is that can you create a macro to do dynamic 
Zap channel allocation for a out going call?
   I don't want to reserve a channel/port in the TDM400P card for "Out 
break" calls, so i was just wandering if some1 could help me a bit 
over here.

[outgoing]  ;Dial 
"0" on the phone for external line

exten = _0,1,Dial(Zap/4/$EXTEN)   ;=== statically 
allocated to Zap/4 needs to be dynamic
exten = _0,2,Goto(102)
exten = _0,102,Congestion
exten = _0,103,Hangup

I'll apreciate any help in this regard.

  
  

  
  

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Re: [Asterisk-Users] Snom and Multiple calls

2005-04-02 Thread Karl Brose

Did you enable call waiting?
call_waiting: on

Josh Dady wrote:
I've got an issue on the snoms, and I'm wondering if anyone has some 
recent experience with it; I've contacted the one specific reference I 
found to it in the list archives, and the person in question didn't 
seem to find an answer (and snom doesn't appear to be finished moving 
their offices yet).

On the snom (I've tested this on the 220 and 360), the phone will 
immediately reject any new INVITE that arrives with 486 BUSY HERE if 
there's already a call on the phone opening (i.e., either the phone is 
already ringing or you've dialed a call that hasn't been answered 
yet).  If we were to give one of these phones to our receptionist, 
obviously, that wouldn't be acceptable.  Is there a way to disable 
this behavior?

--
Joshua P. Dady

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[Asterisk-Users] SjPhoneH323

Can I use SJphone like a H.323 phone in order to dial and receive call
through Asterisk?Can I consider Asterisk just like a sort of H323
Gateway?

Thanks 4 all!

flx
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[Asterisk-Users] Passing varibles out of macros

2005-04-02 Thread Joe Presto

Hi, I have a dialplan that has a called party indicate whether they want to
accept a call by pressing 1. I'm using a feature found in CVS head.

It works great, except that if the call is connected, and the called party
hangs up first, the caller goes into voicemail.

I tried to work around this by passing a variable out of the macro, but that
doesn't appear to work.  Here's my dialplan - what's the best way to
accomplish this?

[dg-extensions]
exten = 1,1,Playback(dg-connect-to-sales)
exten = 1,2,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) 
exten = 1,3,Monitor(wav,${CALLFILENAME},m) 
exten = 1,4,SetMusicOnHold,sales
exten = 1,5,Dial(SIP/sales|30|gmM(screen))
exten = 1,6,GotoIf($[${screenresult} = accept ] ?8:7)
exten = 1,7,VoiceMail,su1
exten = 1,8,Wait(0)
exten = 1,107,VoiceMail,su1

[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Read(ACCEPT|all-your-base|1)
exten = s,3,GotoIf($[${ACCEPT} = 1 ] ?7:4)  ;5:4
exten = s,4,SetVar(MACRO_RESULT=CONTINUE) ;do not connect call
exten = s,5,SetVar(screenresult=deny)
exten = s,6,Goto(s,8) 
exten = s,7,SetVar(screenresult=accept)  ;connect call
exten = s,8,Wait(0)

Thanks in advance - Joe

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Re: [Asterisk-Users] Asterisk Voice mail with CCM

2005-04-02 Thread Nathan Alberti

I'm currently in the process of getting it to work for a CCME install, I 
have it all working except for one thing.. I think it was calling a 
phone from the asterisk server the call transfer back to asterisk would 
fail with an authentication issue and die. I'm pretty sure this issue 
can be resolved I just have not had the time recently wo work on it, I 
can provide more info when I'm back in the office next week.

Nathan Reeves wrote:
Anyone running Cisco Call Manager and using Asterisk for voice mail
services?  Things working well or is the concept a bit of a hassle to
implement?
TIA
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Re: [Asterisk-Users] Re: Are there online forums instead of this email

2005-04-02 Thread Joseph

Que? You told someone to grow up?
YE HAA.. We 'be out har' in rural internet space, mamma
Stepping in cow pats .   Way out past the fancy, hi falutin',
city-slicker, collaborative tools of 2005   ..
YE... HAA.   SMTP MAIL!W.
G  I'm really impresses out har'.  Goooll moses.
woo
Um, yeah, check that. Sorry, this horse-and-buggy system gives a lot 
more flexibility and lets tools be built off of it. If you don't like it 
we got that, but please, to repeat you, Grow up and stop posting to 
this tread.

Thanks,
--Joseph
Tim Bass wrote:
Tom Ivar Helbekkmo.   Grow up and stop posting to this tread.
Nobody cares about your bullying insults.
 

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[Asterisk-Users] Outbound calls with xlite and Xpro PocketPC

2005-04-02 Thread Robert Keller

My issue is dialing out to a local prefix does not always connect. The telco
were sorry, your call does not come thru... message is received. If I dial
my cell phone (a 201 prefix vs. a 758 prefix) then my cell phone rings every
time. My clients are Xlite on a Mac and Xpro on a pocketpc.

I have an Asterisk (AMP) server running with an X100P clone card connected
to an analog line. 

What could I be missing?




Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.

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[Asterisk-Users] problem detecting answer on pri card

2005-04-02 Thread Richard

Hi,

I have a digium PRI T1 card connecting to my carrier. However it has
problems to detect the answer signal on some numbers. For example,
1-800-225-2525 is KLM airline's reservation line. It should answer right
away. But * can't detect it is answered and keeps ringing the ip phone. I
put a monitor on the channel, and get the answer messages in the channels.
So somehow the line is answered but * doesn't know. I don't have a problem
to most numbers. The problem only got my attention after one customer
reported it.

A debug on the pri shows,
Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]

So maybe the inband information is not detected by *?

Anyone has the same setup, i.e. PRI to your carrier? Can you please dial the
number 1-800-225-2525 and have 'pri debug'? I'd like to compare the results.
I am not sure if it is * or just my * configuration.

Your help is highly appreciated. I am really stuck here.

Thanks,
Richard


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[Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

2005-04-02 Thread Josh Alberts

I'm having trouble getting asterisk to run at startup using Ubuntu. 
I've checked, and the asterisk dameon is set to run at init 5.  However,
I'm not seeing anything that says that asterisk has been started during
the boot process.  Oddly, when I shut the machine down/run init6, it
says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
came with asterisk (I installed using synaptic and the debian universe
repositories).  I've edited /etc/default/asterisk, uncommented the first
line and changed start asterisk to yes.  Anybody know what might be
wrong?
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[Asterisk-Users] how to tell what ${DIALSTATUS} is being set

2005-04-02 Thread Randy Bush

i often have nufone problems, e.g.

-- Executing Dial(SIP/konaa0p-4b88, IAX2/[EMAIL PROTECTED]/14086661234) 
in new stack
-- Called [EMAIL PROTECTED]/14086661234
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/NuFone/5'

sound of surf (on a boogie board kind of day) for a fairly long while

  == No one is available to answer at this time
-- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
  == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on 
'SIP/konaa0p-4b88'
-- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
  == Spawn extension (dial-gateways, h, 1) exited non-zero on 'SIP/konaa0p-4b88'

i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them.  the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing.  with BUSY vs CONGESTION this is even more of an
issue.

is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?

randy

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Re: [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

2005-04-02 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Josh Alberts wrote:
 I'm having trouble getting asterisk to run at startup using Ubuntu. 
 I've checked, and the asterisk dameon is set to run at init 5.  However,
 I'm not seeing anything that says that asterisk has been started during
 the boot process.  Oddly, when I shut the machine down/run init6, it
 says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
 came with asterisk (I installed using synaptic and the debian universe
 repositories).  I've edited /etc/default/asterisk, uncommented the first
 line and changed start asterisk to yes.  Anybody know what might be
 wrong?

Try starting atserisk at run level 2 or 3 (debian/ubuntu does this a
little differently)


- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iQEVAwUBQk7oqEtP/KMNOfRbAQK3sggAgOe9uzdFlJZ6wzeNGtZA8L6td2sPTdGy
dxbPd0pbjYBLOj0EfF6BvXTJUYYg4FcGvxfTn0i3ByeFPQHDsMNk0HPRot216qPh
BwXEjSmrwy9BznIqoqS9XToHpyhCV2B+W9sX7QY0g6xZM6tf94gocQ8oH9U2CO+f
L2kVB8yHYYNgx1XCwqihZ1GHWvRnELXALWPOYpYnIzRJt+uWp+66V7VcZWaMjPH5
bMOB0gT33GxHk4itOycMxoV1ECueEV9Ghmlk8pABpCA4RveIkDKHGVMFPnqb9hlp
VWv9ESNmIo/czDcxMd+u+/v4+N0aoPznPqrmrRVW3q47jOvFy5yh5w==
=ARpy
-END PGP SIGNATURE-
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Re: [Asterisk-Users] echo paid support

2005-04-02 Thread James Taylor

Thanks guys,
The echo is there when T1 is connected directly to Asterisk box.
If connected to TNT first then to Asterisk via SIP, then Asterisk echo  
training kicks in (you can hear it 1st second or so) then only a little.

I haven't tried the steps below, but will when I build another box this  
next week.

Thanks,
James
On Fri, 1 Apr 2005 17:09:46 -0700, Damon Estep  
[EMAIL PROTECTED] wrote:

Brian,
By the description of James config there is no zaptel in hus box, looks
like the TDM to SIP conversion is happening on the TNT, is it still your
opinion that the steps below will help?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian M. Arlinghaus
Sent: Wednesday, March 23, 2005 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; James
Taylor
Subject: Re: [Asterisk-Users] echo paid support
James,
After three months of echo, I finally got mine to go away.
I am using a Dell Optiplex GX280 with a P4 3.2GHz processor and 512MB
RAM
and Asterisk version stable 1.07.
Here's what I came up with after reading many of the posts and much
trial
and error.
1. Make sure that ther is no GUI loaded on the asterisk box.
2. Recompile Zaptel with MMX enabled (I think this applies to Intel
Processors only, but someone may correct me if I'm wrong.)
To enable MMX in zaptel, before you compile zaptel, uncomment the line
in
/usr/src/zaptel/zconfig.h file that says:
/* #define CONFIG_ZAPTEL_MMX */
and change it to:
#define CONFIG_ZAPTEL_MMX
3. Recompile Zaptel with the Aggressive Suppressor enabled.  I have
never
read anything about this, but saw it while I was enabling the MMX
support.
From reading zconfig.h, there are different versions of the echo
canceller,
but the comments say that the aggressive suppressor works with
MARK2
which is what was enabled by default in stable 1.07.
To enable the MARK2 AGGRESSIVE SUPPRESSOR in zaptel, before you
compile
zaptel, uncomment the line in /usr/src/zaptel/zconfig.h file that
says:
/* #define AGGRESSIVE_SUPPRESSOR */
and change it to:
#define AGGRESSIVE_SUPPRESSOR
4. Recompile Zaptel with the instructions reordered.  I don't know
what
this does, but it was recommended in these posts for fixing echo.
To reorder the instructions in zaptel, before you compile zaptel,
add
the
following in /usr/src/zaptel/Makefile underneath the comment in the
Makefile
talking about all the config settings being in zconfig.h.  From the
looks
of
it, it might only effect Pentium 4s???
CFLAGS+=-march=pentium4
5. Make sure the Zaptel card is not sharing an IRQ with othe hardware.
In
my case, this involved moving my T100P to another slot and disabling
all
USB
ports.  (I login remotely to administer the asterisk box since the
GX280
doesn't have a PS/2 keyboard port.)
6. Make sure that you have a sufficient processor and sufficient RAM.
I
didn't make any additions to my configuration, but I did remove one
256MB
RAM chip that seemed to be bad leaving 256MB.
Hope this helps.  Again, most of this is not from me, but others here
with
much more knowledge.
Brian Arlinghaus
- Original Message -
From: James Taylor [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, March 17, 2005 10:40 AM
Subject: [Asterisk-Users] echo paid support
 I've got echo problems.
 *** I'm looking for paid support. ***
 I'll accept free support, but don't mind paying if someone really
knows
 what they are doing.


 I've read the wiki, etc.
 Played with the settings in zapata.conf
 Using V400P

 PSTN-_T1-_ASTERISK-_BROADVOICE-_PSTNECHO ON CALLED PHONE
 PSTN-_T1-_ASTERISK-_T1-_PSTNNO ECHO
 VOIP-_ASTERISK-_T1-_PSTN ECHO ON VOIP PHONE G711

 I have another trunk group and different T1's that go to a MAX TNT
first:
 PSTN-_T1-_MAX_TNT-_VOIP-_ASTERISK-_VOIP_PHONE  ECHO ON VOIP
PHONE
 g711


PSTN-_T1-_MAX_TNT-_VOIP_G711-_ASTERISK_IAX_GSM-_ASTERISK_IAX_GSM-
_VOIP_PHONE_g711
 NO ECHO
 --
 James Taylor
 MetroTel
 3505 Summerihll Road
 Suite 11
 Texarkana, Texas  75503
 903-793-1956
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

2005-04-02 Thread James Gardiner


Asterisk 2.0 on Windows..  This is all very much a bit of a Joke, but one
does beg to ask.

When will version 2.0 be released???


James



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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

2005-04-02 Thread Brian Capouch

James Gardiner wrote:
Asterisk 2.0 on Windows..  This is all very much a bit of a Joke, but one
does beg to ask.
When will version 2.0 be released???
2.0 is just now really being talked about in earnest.
I think a better question would be when 1.2 is going to be out.
That one has more narrow bounds.
B.
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RE: [Asterisk-Users] problem detecting answer on pri card

2005-04-02 Thread Alex Vishnev

I have seen that before when you mismatch the type of pri flavor. For
example, if you carrier gives you 4ess and you put 5ess in your config.
There are subtle differences in packets. I would check the configuration on
your carrier side and * side. 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Sent: Saturday, April 02, 2005 1:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] problem detecting answer on pri card

Hi,

I have a digium PRI T1 card connecting to my carrier. However it has
problems to detect the answer signal on some numbers. For example,
1-800-225-2525 is KLM airline's reservation line. It should answer right
away. But * can't detect it is answered and keeps ringing the ip phone. I
put a monitor on the channel, and get the answer messages in the channels.
So somehow the line is answered but * doesn't know. I don't have a problem
to most numbers. The problem only got my attention after one customer
reported it.

A debug on the pri shows,
Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]

So maybe the inband information is not detected by *?

Anyone has the same setup, i.e. PRI to your carrier? Can you please dial the
number 1-800-225-2525 and have 'pri debug'? I'd like to compare the results.
I am not sure if it is * or just my * configuration.

Your help is highly appreciated. I am really stuck here.

Thanks,
Richard


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RE: [Asterisk-Users] problem detecting answer on pri card

2005-04-02 Thread Richard

Both use national as switchtype. I put a traditional pbx to the circuit.
Everything is working. Any suggestion?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Alex Vishnev
 Sent: Saturday, April 02, 2005 9:26 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] problem detecting answer on pri card

 I have seen that before when you mismatch the type of pri flavor. For
 example, if you carrier gives you 4ess and you put 5ess in your config.
 There are subtle differences in packets. I would check the configuration
 on
 your carrier side and * side.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Richard
 Sent: Saturday, April 02, 2005 1:20 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] problem detecting answer on pri card

 Hi,

 I have a digium PRI T1 card connecting to my carrier. However it has
 problems to detect the answer signal on some numbers. For example,
 1-800-225-2525 is KLM airline's reservation line. It should answer right
 away. But * can't detect it is answered and keeps ringing the ip phone. I
 put a monitor on the channel, and get the answer messages in the channels.
 So somehow the line is answered but * doesn't know. I don't have a problem
 to most numbers. The problem only got my attention after one customer
 reported it.

 A debug on the pri shows,
 Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
 progress information may be available inband. (1) ]

 So maybe the inband information is not detected by *?

 Anyone has the same setup, i.e. PRI to your carrier? Can you please dial
 the
 number 1-800-225-2525 and have 'pri debug'? I'd like to compare the
 results.
 I am not sure if it is * or just my * configuration.

 Your help is highly appreciated. I am really stuck here.

 Thanks,
 Richard

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[Asterisk-Users] Book Review: VoIP Telephony with Asterisk

2005-04-02 Thread Kerry Garrison




The book bills itself as a beginner's guide to Asterisk and Voice over IP 
(VoIP). Even with over 270 pages, it isn't possible to go through every single 
feature that Asterisk has to offer but the book does give enough information to 
get you started and even apply a few advanced features to your phone system. For 
those of you not familiar with Asterisk, VoIP, or PBX's we will need a little 
bit of background for you to know if this book is for you. 
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=23Itemid=26

-Kerry
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[Asterisk-Users] Zaptel Anti-MMX Optimizations

2005-04-02 Thread Trevor Peirce

Hello,
Running Fedora Core 2 with a Celeron processor I'm seeing a significant 
problem when enabling MMX optimizations.

I'll gladly submit a bug report, but I don't know what information is 
useful.

First, with -no- MMX optimizations enabled, here is what I see
CLI show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 11118 -24
 ulaw - 5 - 1 3 2 11118 -24
 alaw - 5 1 - 3 2 11118 -24
 g726 - 6 3 3 - 3 21219 -25
adpcm - 5 2 2 3 - 11118 -24
 slin - 4 1 1 2 1 -1017 -23
lpc10 - 7 4 4 5 4 3 -20 -26
 g729 - 7 4 4 5 4 313 - -26
speex - - - - - - - - - - -
 ilbc - 8 5 5 6 5 41421 - -
If I enable MMX_OPTIMIZATIONS (and change nothing else), it gets quite 
worse

CLI show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 14820 - 446270
 ulaw - 5 - 1 3 2 14820 - 446270
 alaw - 5 1 - 3 2 14820 - 446270
 g726 - 6 3 3 - 3 24921 - 446271
adpcm - 5 2 2 3 - 14820 - 446270
 slin - 4 1 1 2 1 -4719 - 446269
lpc10 -   368   365   365   366   365   364 -   383 - 446633
 g729 -23202021201966 - - 446288
speex - - - - - - - - - - -
 ilbc -   266   263   263   264   263   262   309   281 - -
And just incase you think I've got an AMD, here you go...
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 1
model name  : Intel(R) Celeron(R) CPU 1.70GHz
stepping: 3
cpu MHz : 1716.114
cache size  : 128 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca 
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
bogomips: 3399.68

If this isn't supposed to work with Celerons I'd like to update the 
documentation. On the other hand if it's a problem with my system I'd 
like to resolve it :)

TIA!
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Re: [Asterisk-Users] Integrating door intercom?

2005-04-02 Thread Jeff Noxon

On Fri, Apr 01, 2005 at 04:07:30PM -0500, C F wrote:
 Well, depends how you set it up. If you leave it as is, it will only
 ring 3 times. You can't just call up the box (without the chip), b/c
 it will just throw you to the other end of the doorbell fon (the co
 port). So no this is one of the more cheaper one and I wouldn't
 recommend it with Asterisk, try Vikingelectronics instead (the c2000
 from them even support callerid). Or you can try Valcom.

In my configuration, the Doorbell Fon has a dedicated FXO port.  I would
not recommend using it any other way.  Asterisk answers immediately
when the user hits the button on the intercom and indicates ringing to
the intercom.  Asterisk sets the Caller ID, and rings my house phones
(and cell phone) with distinctive ring.

Personally I have no need or desire to make calls to the intercom box.
Unless someone pushed the button, I'd be unaware anyone was there in
the first place.

 The distinctive ring doesn't really work with asterisk, since it is
 never (well, almost never 1 out of 5 might repeat, but then again it
 might switch the pattaren with the other box) exactly the same
 pattaren. Caller ID just simply doesn't work with this box, it does'nt
 send callerid, the only thing you acomplish by turning it off, is to
 ring the phones imediatly.

There's no need for the distinctive ring to work with Asterisk unless you
are trying to get by with one FXO port for both the doorbell and a POTS
line.  I have not tested that configuration and would not recommend it.
With a dedicated FXO interface for the doorbell, those issues go away.

Regards,

Jeff
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Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations

2005-04-02 Thread Michael D Schelin

Hay Trevor, what would be the problem if you were using AMD processors?
Trevor Peirce wrote:
Hello,
Running Fedora Core 2 with a Celeron processor I'm seeing a 
significant problem when enabling MMX optimizations.

I'll gladly submit a bug report, but I don't know what information is 
useful.

First, with -no- MMX optimizations enabled, here is what I see
CLI show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 11118 -24
 ulaw - 5 - 1 3 2 11118 -24
 alaw - 5 1 - 3 2 11118 -24
 g726 - 6 3 3 - 3 21219 -25
adpcm - 5 2 2 3 - 11118 -24
 slin - 4 1 1 2 1 -1017 -23
lpc10 - 7 4 4 5 4 3 -20 -26
 g729 - 7 4 4 5 4 313 - -26
speex - - - - - - - - - - -
 ilbc - 8 5 5 6 5 41421 - -
If I enable MMX_OPTIMIZATIONS (and change nothing else), it gets quite 
worse

CLI show translation
   Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
   g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
 g723 - - - - - - - - - - -
  gsm - - 2 2 3 2 14820 - 446270
 ulaw - 5 - 1 3 2 14820 - 446270
 alaw - 5 1 - 3 2 14820 - 446270
 g726 - 6 3 3 - 3 24921 - 446271
adpcm - 5 2 2 3 - 14820 - 446270
 slin - 4 1 1 2 1 -4719 - 446269
lpc10 -   368   365   365   366   365   364 -   383 - 446633
 g729 -23202021201966 - - 446288
speex - - - - - - - - - - -
 ilbc -   266   263   263   264   263   262   309   281 - -
And just incase you think I've got an AMD, here you go...
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 1
model name  : Intel(R) Celeron(R) CPU 1.70GHz
stepping: 3
cpu MHz : 1716.114
cache size  : 128 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca 
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
bogomips: 3399.68

If this isn't supposed to work with Celerons I'd like to update the 
documentation. On the other hand if it's a problem with my system I'd 
like to resolve it :)

TIA!
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Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations

2005-04-02 Thread Trevor Peirce

Michael D Schelin wrote:
Hay Trevor, what would be the problem if you were using AMD processors?
/*
* Define if you want MMX optimizations in zaptel
*
* Note: CONFIG_ZAPTEL_MMX is generally incompatible with AMD
* processors and can cause system instability!
*
*/
/* #define CONFIG_ZAPTEL_MMX */
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RE: [Asterisk-Users] how to tell what ${DIALSTATUS} is being set

2005-04-02 Thread [EMAIL PROTECTED]

Have you tried a simple hangup NoOp to output ${DIALSTATUS} to the CLI?

exten = h,1,NoOp(${DIALSTATUS})

-josiah


Original Message:
-
From: Randy Bush [EMAIL PROTECTED]
Date: Sat, 2 Apr 2005 10:44:46 -0800
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how to tell what ${DIALSTATUS} is being set


i often have nufone problems, e.g.

-- Executing Dial(SIP/konaa0p-4b88, IAX2/[EMAIL PROTECTED]/14086661234)
in new stack
-- Called [EMAIL PROTECTED]/14086661234
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/NuFone/5'

sound of surf (on a boogie board kind of day) for a fairly long while

  == No one is available to answer at this time
-- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
  == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on
'SIP/konaa0p-4b88'
-- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
  == Spawn extension (dial-gateways, h, 1) exited non-zero on
'SIP/konaa0p-4b88'

i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them.  the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing.  with BUSY vs CONGESTION this is even more of an
issue.

is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?

randy

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[Asterisk-Users] Re: how to tell what ${DIALSTATUS} is being set

2005-04-02 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Randy Bush [EMAIL PROTECTED] wrote:
 i often have nufone problems, e.g.
 
 -- Executing Dial(SIP/konaa0p-4b88, IAX2/[EMAIL 
 PROTECTED]/14086661234) in new stack
 -- Called [EMAIL PROTECTED]/14086661234
 -- Call accepted by 66.225.202.72 (format ulaw)
 -- Format for call is ulaw
 -- Hungup 'IAX2/NuFone/5'
 
 sound of surf (on a boogie board kind of day) for a fairly long while
 
   == No one is available to answer at this time
 -- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
   == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on 
 'SIP/konaa0p-4b88'
 -- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
   == Spawn extension (dial-gateways, h, 1) exited non-zero on 
 'SIP/konaa0p-4b88'
 
 i would like to detect this (and many other things) in ${DIALSTATUS}
 conditions so that i can then GotoIf() them.  the problem is that the
 log does not tell me explicitly which ${DIALSTATUS} has been returned,
 leaving me guessing.  with BUSY vs CONGESTION this is even more of an
 issue.
 
 is it reasonable to ask that the log contain the value being set in
 ${DIALSTATUS}?

I find it useful to follow the Dial command with a NoOp command as follows:
 NoOp(DIALSTATUS=${DIALSTATUS})

Then it shows up in the log. It also provides something for my Manager event
parser to see, in order to discover the reason for a failed call.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] wctdm module parameters (Was: Issues with ringing on FXS ports)

2005-04-02 Thread Richard Scobie


[EMAIL PROTECTED] wrote:
Is there a list of these anywhere?  This is now the third one I've heard 
of, with no documentation:  lowpower (IIRC), robust and now boostringer. 
Do I have to go diving in the source, or is there a Wiki I can't find?
I have only ever found the information in the driver source of on the 
CVS list as they have been added.

There is a list of them at the end of wctdm.c.
The non obvious ones I know about:
opermode=COUNTRY
Where COUNTRY is one from the list near the top of wctdm.c This will set 
the A.C. and D.C. line impedance on the FXO modules to suit the telecom 
standard used in that country. Default if not set, is FCC (US/Canada).

fxshonormode=1
If used, it must be in conjuction with the above. This will set the A.C. 
and D.C. line impedance on the FXS modules to match COUNTRY. Default if 
not set, is FCC (US/Canada).

lowpower=1
Reduces ringing volts on FXS to 50V peak.
boostringer=1
Boosts ringing volts on FXS to 89V peak.
Regards,
Richard
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[Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk

2005-04-02 Thread Alexandre Otto Durr

Hi for all!

I saw it on http://signate.com/features.php an Open Source PBX Features with
support Cisco Skinny Call Control Protocol.

Is it possible in Asterisk or I need a license for this?

Has anyone using Asterisk with Cisco Skinny?

TIA

Alexandre 


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Re: [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

2005-04-02 Thread Josh Alberts

By default, synaptic puts init scripts in all runlevel folders.  All are
exactly the same, except for init 6, which is supposed to kill the
process.

On Sat, 02 Apr 2005 12:20:40 -0600,
ron at wellsted.org.uk said:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Josh Alberts wrote:
 I'm having trouble getting asterisk to run at startup using Ubuntu. 
 I've checked, and the asterisk dameon is set to run at init 5.  However,
 I'm not seeing anything that says that asterisk has been started during
 the boot process.  Oddly, when I shut the machine down/run init6, it
 says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
 came with asterisk (I installed using synaptic and the debian universe
 repositories).  I've edited /etc/default/asterisk, uncommented the first
 line and changed start asterisk to yes.  Anybody know what might be
 wrong?

Try starting atserisk at run level 2 or 3 (debian/ubuntu does this a
little differently)


- --
Ron Wellsted
http://www.wellsted.org.uk
ron at wellsted.org.uk
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Re: [Asterisk-Users] problem detecting answer on pri card

2005-04-02 Thread Trevor Peirce

Richard wrote:
A debug on the pri shows,
Ext: 1  Progress Description: Call is not end-to-end ISDN; further call
progress information may be available inband. (1) ]
So maybe the inband information is not detected by *?
 

I can't help you debug, but I see this same progress message and can 
hear their system fine.  No more messages appear until I hangup and a 
DISCONNECT goes out.


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Re: [Asterisk-Users] Book Review: VoIP Telephony with Asterisk