date:20051017


Patrick de Kok schrieb:

What large number of answers?
 
If I scroll through the lists no answers are present..and previous posts 
do not seem to help as well..


That is the point. No one seems to use ISDN together with chan_modem. Do 
yourself a favour and use chan_capi or chan_capi-cm

--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA

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Re: [Asterisk-Users] Newbi stating question


Lorenzo wrote

I'm working in an italian company who wan't start using asterisk.
 
My problem:
1. What kind of hardware I need for make a PBX who speek with 3/4 ISDN 
BRI line and internal use a SIP VoIP telephone?


4 ISDN channels or lines. If 4 ISDN channels, get two HFC-S based cards, 
if 4 lines (i.e. 8 channels) get either an AVM C4 or an junghanns.net 
quadbri card.



2. I can install ASTERISK and HYLAFAX on the same machine ?


Yes. Running (mostly) fine here, using an AVM C4 connected to two lines. 
Hylafax is receiving all faxes, using a ATA connected to Asterisk for

faxing out.

Take a look at voip-info.org and search for zaphfc and/or chan_capi.
--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA

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[Asterisk-Users] Polycom MWI

2005-10-17 Thread Wilson Pickett

Hi,

I have lookedaround and don't see this anywhere. Is there a way to
tell the ip500 to not make the aural MWI blips?
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Re: [Asterisk-Users] huge problem compiling * on gcc4.x (SUSE 10.0)

2005-10-17 Thread Dave Cotton

On Sun, 2005-10-16 at 21:21 +0200, [Ludwig IT-Services - GMAIL ] -
Michael Ludwig wrote:
 Hello to all of you!
 
 I'm very new to this list and to asterisk and stuff at all.
 To build my asterisk server I installed a new machine running the new
 SUSE Linux 10.0 (retail version on DVD).
 I need asterisk (tried 1.0.9), bristuff (off junghanns.net,
 -0.2.0-RC8o) and the florz-patch because I have two HFC-S-ISDN cards
 in that machine.
 Now when it comes to compiling I get a huge bunch of warnings and
 stuff, zaptel 1.0.9.2 fails to compile and asterisk 1.0.9 also fails
 to compile.
 
 SUSE 10.0 uses gcc 4.0.2 and as I asked in some other mailing list and
 forums, that is the reason why * stuff fails to compile.
 
 Is there any stable asterisk version available which does compile fine
 on a gcc4.x ?
 
 If not, will the * source be changed to finely compile on gcc 4.x?
 If yes, when will that be? (I need the * stuff now).
 If not, why not?
 
 What's on with the 1.2.0-beta stuff out there on the asterisk.org webpages?
 Does that one compile on gcc4.x ?
 
 Please help! I really need my * box now...

On my systems gcc -v gives
gcc version 4.0.1 (4.0.1-5mdk for Mandriva Linux release 2006.0)

I'm compiling and running 1.0.9 and CVS-HEAD.
Warnings are _not_ errors!
What is and stuff? 

-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] Delayed ringing on some SIP phones

Hello all,

One of the buildings I have an asterisk box deployed in is used by two small
companies on two floors. They have an agreement between them whereby they'll
answer each other's incoming calls and take messages if the office is empty
/ everyone is on the phone.

Each of them has an  ISDN BRI delivered to asterisk via zaphfc, then dropped
into a context as follows:
exten = s,1,SetCIDName(Company 1)
exten = s,2,Dial(SIP/200SIP/201etc.,30)
exten = s,3,Voicemail(su200)

Each company is able to see on the LCD on their SIP phones whether the call
is for them or the folks up/downstairs.

What I'd like to do is implement a delayed ringing strategy - i.e. if the
call comes in for Company 1, only their SIP phones will ring for the first
15 seconds, then if there's not been an answer, company 2's SIP phones will
also start ringing.

Is there any way to do this without stopping Company 1's phones ringing
(i.e. timing out the dial statement after 15 seconds)?

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

2005-10-17 Thread Craig Guy

Do you have a permit line in manager.conf for connections from 127.0.0.1 
such as:


permit = 127.0.0.0/255.0.0.0

And also a bind entry:

bindaddr = 0.0.0.0

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

Sent: Monday, October 17, 2005 1:21 PM
Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk



Hi,

Yes it is enabled I have even checked various logs and nothing... I
checked '/var/log/messages', '/var/log/secure',
'/var/log/asterisk/full', and even '/var/log/mysqld.log' nothing, nada,
nein - its odd that a failed connection attempt is not logged somewhere,
perhaps I must somehow turn logging on for the asterisk management
portal. Any ideas?

Thanks

[EMAIL PROTECTED] wrote:


On 10/17/2005, Michael Furdyk [EMAIL PROTECTED] wrote:



He is just using telnet to check for the port being open/working... (not
telneting to the telnet port)

-- Mike

-Original Message-
[EMAIL PROTECTED]
Sent: Monday, October 17, 2005 12:28 AM
Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

On 10/17/2005, Chuck Bunn [EMAIL PROTECTED] wrote:



Hi,

I cannot do the following:

telnet 127.0.0.1 5038



Is telnet enabled?

Brett




Here it is Sunday - And I been wrong already this week...

Is manager.conf 'enabled=yes'?

Brett
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[Asterisk-Users] Solved? = Playback audio before answered by a queue member

2005-10-17 Thread KRTorio

Regarding my previous post:

Playback audio before answered by a queue member

I added a ResetCDR() command at the middle:


exten = XX,1,Background(audiofile) ;answers the channel immediately
exten = XX,2,ResetCDR();clean slate
exten = XX,3,Queue(Qname|tdn|||);new answer time written

Looking at the CDR, the billsec is no longer the same as duration. This should be the actual talktime that I'm looking for.

Are there any side effects to look out when I use ResetCDR()?
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Re: [Asterisk-Users] Need language variable to user account

2005-10-17 Thread brett

On 10/17/2005, Ronald Wiplinger [EMAIL PROTECTED] wrote:
 [EMAIL PROTECTED] wrote:

 On 10/17/2005, Ronald Wiplinger [EMAIL PROTECTED] wrote:


 My users do have different language requests. I would like to give them
 their wish language.

 I could setup an extra database for that.
 I wonder if it would be much work to add this field in sip.conf (and
 realtime)?



 Ronald...

 IF I had customers who needed different languages via sip.conf...
 I would use the - er... language= setting in there.

 But I don't know if realtime uses it - and I don't know what version of
 asterisk (or [EMAIL PROTECTED] - or whatever) you are using.

 Brett

Brett,

 how would you do that? Giving each language group a different context?
 At least that was I came up with.
 Actually the question is going even further,    Think about that:

 I will create 20 features, 20 different pay plans (tariffs), 
 One of the 20 features is the language.
 If I would use context, I have soon 400 x 10 possible languages
 (slightly exaggerated, hehehehe)

 I guess, if I know how to add languge, than I can add the other features
 as well, ...

 My next try is to setup a feature mysql  database for each user. This
 database will be queried at the beginning of the context and than give
 you all the variables you may need, ...
 Maybe something like that exists?

Apparently I misunderstood the use of the word 'language'...

You mean English, Spanish, Italian, and German - or the actual wording of
prompt itself?

For a language - set the 'language=en' or 'language=es' in the
sip.conf for
that user.  It is 'supposed' to be carried through.  Should be
something on
the wiki about it.

If you mean the wording or the prompts/IVR etc - well - that's why you
get
the 'big bucks'.  8-)

Brett
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Re: [Asterisk-Users] Delayed ringing on some SIP phones

2005-10-17 Thread Hauke Zuehl


Hi :)

Chris Bagnall wrote:

Hello all,

What I'd like to do is implement a delayed ringing strategy - i.e. if the
call comes in for Company 1, only their SIP phones will ring for the first
15 seconds, then if there's not been an answer, company 2's SIP phones will
also start ringing.

Is there any way to do this without stopping Company 1's phones ringing
(i.e. timing out the dial statement after 15 seconds)?



Well, I asked this, too and the solution was:
exten = s,1,Dial(SIP/company1,15)
exten = s,2,Dial(SIP/company1SIP/company2,30)


Thanks in advance.

Regards,

Chris


HTH and regards,
Hauke
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Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

2005-10-17 Thread brett

On 10/17/2005, Chuck Bunn [EMAIL PROTECTED] wrote:
 Hi,

 Yes it is enabled I have even checked various logs and nothing... I
 checked '/var/log/messages', '/var/log/secure',
 '/var/log/asterisk/full', and even '/var/log/mysqld.log' nothing, nada,
 nein - its odd that a failed connection attempt is not logged somewhere,
 perhaps I must somehow turn logging on for the asterisk management
 portal. Any ideas?

Are you 'sure' Asterisk is running?

ps ax

asterisk -r (which maybe shouldn't work if you can't telnet...)

asterisk -c ? ends with a CLI ?  And stays that way?

Brett
P.S. I just connected and it didn't even show that THAT occurred...
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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)

2005-10-17 Thread tim panton



On 17 Oct 2005, at 01:57, Kevin P. Fleming wrote:


Ronald Wiplinger wrote:



Ok, ok, 



Thanks :-)


Combining our findings now:  It seems that firefly wants to  
register every 1200 seconds, but iax.conf only allows 60. How can  
I stop this warning message?




Asterisk has never defaulted to allowing IAX2 registrations longer  
than 60 seconds, but previously it did not say anything when it was  
limiting the expiration period.




By the way, there is a reason for this. It ensures that there is  
traffic (initiated by the client) often
enough to keep the 'connection' in a NATing firewall's map of ports.  
This means that a
'new' call (ie incoming) message from asterisk to the client will be  
seen by the firewall as part of that

'recent' conversation and allowed through (and correctly forwarded).

You have two choices: reconfigure your softphone to only request a  
60 second expiration interval, or reconfigure Asterisk to allow  
longer registrations. There is no direct way to make the message go  
away without reconfiguring one end or the other.


So unless you _know_ the timeouts on all the firewalls involved, I'd  
play safe and change the

firefly end.

Tim.
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Re: [Asterisk-Users] Delayed ringing on some SIP phones


 One of the buildings I have an asterisk box deployed in is used by two small
 companies on two floors. They have an agreement between them whereby they'll
 answer each other's incoming calls and take messages if the office is empty
 / everyone is on the phone.
 
 Each of them has an  ISDN BRI delivered to asterisk via zaphfc, then dropped
 into a context as follows:
 exten = s,1,SetCIDName(Company 1)
 exten = s,2,Dial(SIP/200SIP/201etc.,30)
 exten = s,3,Voicemail(su200)
 
 Each company is able to see on the LCD on their SIP phones whether the call
 is for them or the folks up/downstairs.
 
 What I'd like to do is implement a delayed ringing strategy - i.e. if the
 call comes in for Company 1, only their SIP phones will ring for the first
 15 seconds, then if there's not been an answer, company 2's SIP phones will
 also start ringing.
 
 Is there any way to do this without stopping Company 1's phones ringing
 (i.e. timing out the dial statement after 15 seconds)?

Either this is a very simple question or I'm missing something...

Wouldn't something like this work for you?

[incoming-bri-one]
exten = s,1,SetCIDName(Company 1)
exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones
exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; company 1's  2's 
phones
exten = s,4,Voicemail(su200)

[incoming-bri-two]
exten = s,1,SetCIDName(Company 2)
exten = s,2,Dial(SIP/300SIP/301etc.,15) ; company 2's phones
exten = s,3,Dial(SIP/300SIP/301SIP/200SIP/201etc.,15) ; comapny 2's  1's 
phones
exten = s,4,Voicemail(su300)


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Re: [Asterisk-Users] Delayed ringing on some SIP phones

2005-10-17 Thread brett

On 10/17/2005, Chris Bagnall [EMAIL PROTECTED] wrote:
 Hello all,

 One of the buildings I have an asterisk box deployed in is used by two
 small companies on two floors. They have an agreement between them
 whereby they'll answer each other's incoming calls and take messages if
 the office is empty/everyone is on the phone.

 Each of them has an ISDN BRI delivered to asterisk via zaphfc, then
 dropped into a context as follows:
 exten = s,1,SetCIDName(Company 1)
 exten = s,2,Dial(SIP/200SIP/201etc.,30)
 exten = s,3,Voicemail(su200)

 Each company is able to see on the LCD on their SIP phones whether the
 call is for them or the folks up/downstairs.

 What I'd like to do is implement a delayed ringing strategy - i.e. if the
 call comes in for Company 1, only their SIP phones will ring for the first
 15 seconds, then if there's not been an answer, company 2's SIP phones
 will also start ringing.

 Is there any way to do this without stopping Company 1's phones ringing
 (i.e. timing out the dial statement after 15 seconds)?

Bingo!  You got it!  Timeout the dial after X seconds - and then do a Dial
to both companies for another another X seconds.

Remember - busy does a jump to n+101 (some one is there...) and
unavailable
just goes to the next step.

Brett
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Re: [Asterisk-Users] Need language variable to user account

2005-10-17 Thread trixter aka Bret McDanel

On Mon, 2005-10-17 at 02:42 -0500, [EMAIL PROTECTED] wrote:
 Apparently I misunderstood the use of the word 'language'...
 
 You mean English, Spanish, Italian, and German - or the actual wording of
 prompt itself?
 
 For a language - set the 'language=en' or 'language=es' in the
 sip.conf for
 that user.  It is 'supposed' to be carried through.  Should be
 something on
 the wiki about it.
 
 If you mean the wording or the prompts/IVR etc - well - that's why you
 get
 the 'big bucks'.  8-)
 
 Brett

You could still use language.  Its not limited to two letters, at least
not with SetLanguage() (I havent seen the realtime parts for this so I
dont know there, but it shouldnt be).  

You could use language just to mean 'use a different set of prompts' and
if the prompt doesnt exist in the subdir it will use the 'default
language' prompt.  

So if you have greetings (greeting.gsm):
default: hello there
lang1: how now brown cow
lang2: the quick brown fox jumps

and create in your /usr/share/asterisk/sounds (or whatever) directory a
directory called 'lang1' and a directory called 'lang2' and placed the
appropriate greeting.gsm in each directory the prompts would be
different if you did
language=lang1 *or* SetLanguage(lang1) depending on how you wanted to do
it.

If you cant do this from the realtime stuff you could do it via an AGI
or dbget() to get the appropriate language as needed.


-- 
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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[Asterisk-Users] AGI Problem

2005-10-17 Thread René Enskat [Teamware GmbH]


Hmm still have problems with the get variable with PHP i have this error
now separated with a script:

Sending string GET VARIABLE CALLERIDNUM\n to Asterisk...
Wroten bytes to STDOUT: 25

Reading 80 bytes response from Asterisk...
Received response: 510 Invalid or unknown command



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Re: [Asterisk-Users] Delayed ringing on some SIP phones

2005-10-17 Thread trixter aka Bret McDanel

On Mon, 2005-10-17 at 02:50 -0600, Rich Adamson wrote:
  Is there any way to do this without stopping Company 1's phones ringing
  (i.e. timing out the dial statement after 15 seconds)?
 
 Either this is a very simple question or I'm missing something...
 
 Wouldn't something like this work for you?
 
 [incoming-bri-one]
 exten = s,1,SetCIDName(Company 1)
 exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones
 exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; company 1's  
 2's phones
 exten = s,4,Voicemail(su200)
 
 [incoming-bri-two]
 exten = s,1,SetCIDName(Company 2)
 exten = s,2,Dial(SIP/300SIP/301etc.,15) ; company 2's phones
 exten = s,3,Dial(SIP/300SIP/301SIP/200SIP/201etc.,15) ; comapny 2's  
 1's phones
 exten = s,4,Voicemail(su300)
 

There is a potential race condition however, and that is also what he
wanted to know if there was a different way than that :)

The race condition occurs in between priorities 2  3.  If someone picks
up just as the phone stops ringing on priority 2 but before it starts on
3 (granted a small window but a window none the less) then it wont be
answered and instead it will seem to most users to be a defective pbx
system (they picked up the phone and it didnt answer but kept
ringing).  

Short of an AGI that will create call files as needed and patch the call
through when its answered I dont see another way however.  Basically the
AGI would create a call file that would ring the extensions and when
someone answers it would transfer the inbound call to the person that
answered.  A bit messier and potentially more problems (or at least it
seems like it) but it wouldnt stop the dial command halfway through.




-- 
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] Delayed ringing on some SIP phones

2005-10-17 Thread trixter aka Bret McDanel

On Mon, 2005-10-17 at 03:01 -0500, [EMAIL PROTECTED] wrote:
 Bingo!  You got it!  Timeout the dial after X seconds - and then do a Dial
 to both companies for another another X seconds.
 
 Remember - busy does a jump to n+101 (some one is there...) and
 unavailable
 just goes to the next step.

dont forget that with bristuff (and presumably 1.2-beta1) n+201 is
called if the client isnt connected that you tried to dial.  But if you
are using 1.2 you really should use labels they are so much nicer :)


-- 
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] Modifying Voicemail App

2005-10-17 Thread Neil Skowronek

I hear ya Luki, thanks for looking at the 6600 line
file, at least Xcode gives it pretty colors.

I'll admit it, I'm Clueless, but am determined to
figure this out by finding people who have done any
modifications to the file and learn from them - while
learning C of course, should have 20 years ago :-P

This guy had an interesting C patch for customizing
extention numbers.

http://www.voipuser.org/forum_topic_2952.html

So hear me community, share your app_voicemail.c
experiences!

Neil

--- Luki [EMAIL PROTECTED] wrote:

  Is it an agi, special dailplan, patching the
  app_voicemail.c file? All three?
 
 All voice mail related functionality is in
 app_voicemail.c; sounds are
 the respective vm-*.gsm files in the sounds
 directory.
 
 If you want to modify the functionality and more
 prompts around,
 you'll have to make your hands dirty and confront
 the about 6000 lines
 long app_voicemail.c file... and edit or re-record
 the sounds. Good
 luck.
 
  mark urgent
  add to message
  pause while recording message
 
 I didn't look at the code too closely but neither of
 your three
 requests is quite straight forward, especially if
 you don't have C
 coding experience.
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Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread Lenz



Hello,
are you using Asteriks agents or dialing straight to extensions? because  
if you are using agents for incoming calls and then you dial straight  
out of Asterisk, Asterisk will not know that the agent is busy. One  
possible workaround would be to make a call to the agent using a .call  
file, so that the agent is busy and the queue system recognizes it.

(It's just an idea, I have never tried this)
Thanks
l.


On Sun, 16 Oct 2005 04:04:02 +0200, J Thomas [EMAIL PROTECTED] wrote:


One of my friends is facing this problems and I could not find any
solution to that. Hence this post.

In her Asterisk PBX, she has programmed about 10 agents, and strategy is
rrmemory. Everything works fine. When an agent has received an ACD call,
another call is not presented to him as long as he is on the ACD call.

However when an agent has made an outgoing call, he is still presented
another ACD call when his turn comes. This results in unnecessary delay
in answering that call.

Taking out call waiting is not an option, as an agent can also get a
direct dialed call, and he should be able to pick up that call even when
he is on another call.

Is there a way so that a busy agent (whether busy because of an incoming
call, or outgoing call) is not presented another ACD call?

Thanks,
-- jt


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

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[Asterisk-Users] Re: Modifying Voicemail App

2005-10-17 Thread Justin Newman

You will need to modify /usr/src/asterisk/apps/app_voicemail.c. Fairly easy
task.

On another note, I'm surprised the IVR within apps such as voicemail isn't
drawn out into a app specific app/dialplan. The application flow could then
be easily customized by end users. This wouldn't be too hard to do...

-J

 Date: Sun, 16 Oct 2005 22:57:48 -0700 (PDT)
 From: Neil Skowronek [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Modifying Voicemail App

 I want to add things to the prompts like:

 mark urgent

 add to message

 pause while recording message

 Any examples of how to do this?

 I'd also like to switch around prompts, not simply
 edit the sound files.

 Is it an agi, special dailplan, patching the
 app_voicemail.c file? All three?

 Any input/examples are welcome.

 -thanks


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RE: [Asterisk-Users] Delayed ringing on some SIP phones

 Wouldn't something like this work for you?
 [incoming-bri-one]
 exten = s,1,SetCIDName(Company 1)
 exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones
 exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; 
 company 1's  2's phones
 exten = s,4,Voicemail(su200)
 [incoming-bri-two]
 exten = s,1,SetCIDName(Company 2)
 exten = s,2,Dial(SIP/300SIP/301etc.,15) ; company 2's phones
 exten = s,3,Dial(SIP/300SIP/301SIP/200SIP/201etc.,15) ; 
 comapny 2's  1's phones
 exten = s,4,Voicemail(su300)

Thanks for the replies folks.

My concern is that the SIP phones in question (GXP-2000s) tend to take a
second or two to realise they're no longer ringing. If phones are ringing
from the first dial statement and still think they're ringing when the
second dial statement is executed, they will all report busy to asterisk and
not ring at all.

I suppose I could insert a Wait(2) or something like that between the two
dial statements, but I can see it causing problems with users picking the
phone up and finding nothing on the end, then when the second dial kicks in,
their phone reports busy because it's off-hook. What I'm really after is a
method of starting the second dial on company 2's phones without
interrupting the dial on company 1's phones.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] AGI Problem

2005-10-17 Thread Obelix

Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:

In my experience most AGI problems I had came from other info sent to the
terminal via verbose commands and other stdout output. There is some info on
the voip-info wiki about using AGI.

I use the phpagi 2 library, and carefully setting up the agi-verbose commmands
fixes my 510 problems


 Hmm still have problems with the get variable with PHP i have this error
 now separated with a script:

 Sending string GET VARIABLE CALLERIDNUM\n to Asterisk...
 Wroten bytes to STDOUT: 25

 Reading 80 bytes response from Asterisk...
 Received response: 510 Invalid or unknown command



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Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

2005-10-17 Thread Obelix

Quoting Chuck Bunn [EMAIL PROTECTED]:

Check your firewall configuration. New versions of Linux come with tighter
default firewall configurations.

Check these notes from Redhat to see what processes if any are listening on the
relevant ports.

http://www.redhat.com/docs/manuals/linux/RHL-9-Manual/security-guide/s1-server-ports.html

Is your manager.conf properly configured? ;-)


 Hi,

 I cannot do the following:

 telnet 127.0.0.1 5038

 I get connection refused and this is preventing AMP from installing. I
 had this working when I was using FC3 but I had to upgrade to FC4 for
 another application. So I am running PHP5, MYSQL 4 with FC4 and asterisk
 is running (I had this problem before with FC3 and it turned out
 asterisk was not running) I am using 1.2.0 beta1 Asterisk code.

 Thanks
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[Asterisk-Users] Incoming call problem - ringing SIP devices report busy

bump from last week

Hi all,

I have 12 SIP phones at a particular site all connected to a local asterisk
server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming
calls. An IAX gateway is used for outbound calls. At the moment, when an
incoming call comes in, asterisk dials every SIP phone like so:
Dial (SIP/1SIP/2etc.)
This has worked fine for some months, but I noticed a few days ago that if 2
calls come in only a second or two apart, the first one will cause the dial
command to be executed, and when the second call comes in, it'll go to
voicemail because *all* the SIP phones report themselves as busy (because
they're ringing for the first call).

Is there any way around this problem whilst keeping the same incoming call
behaviour (i.e. call comes in, all phones ring)?

Would it be better to do something like this using queues on a ringall
strategy (and would it solve the problem)?

Is it even possible to use queues without asterisk answering the call until
it's been connected to a human being?

Thanks in advance.

Regards,

Chris
-- 
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This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Delayed ringing on some SIP phones

2005-10-17 Thread jurczak

Why dont you make a special extension where you could provide the delay and 
the numbers you want to dial?

for example

exten = _900X,1,Wait(${EXTEN:4:2})
exten = _900X,2,Dial(SIP/${EXTEN:5})

then in the incoming context you could dial 

exten = s,1,Dial(SIP/200SIP/201LOCAL/90015300LOCAL/90015301)

in the above example 200 and 201 extension will ring immediately, and 300 and 
301 will start ringing after 15 seconds.
after to 900 the first two digits are for the delay before start ringing
and the last three are the extension that should be called.


On Mon, 17 Oct 2005 02:50:46 -0600, Rich Adamson wrote
  One of the buildings I have an asterisk box deployed in is used by two 
small
  companies on two floors. They have an agreement between them whereby 
they'll
  answer each other's incoming calls and take messages if the office is 
empty
  / everyone is on the phone.
  
  Each of them has an  ISDN BRI delivered to asterisk via zaphfc, then 
dropped
  into a context as follows:
  exten = s,1,SetCIDName(Company 1)
  exten = s,2,Dial(SIP/200SIP/201etc.,30)
  exten = s,3,Voicemail(su200)
  
  Each company is able to see on the LCD on their SIP phones whether the 
call
  is for them or the folks up/downstairs.
  
  What I'd like to do is implement a delayed ringing strategy - i.e. if the
  call comes in for Company 1, only their SIP phones will ring for the first
  15 seconds, then if there's not been an answer, company 2's SIP phones 
will
  also start ringing.
  
  Is there any way to do this without stopping Company 1's phones ringing
  (i.e. timing out the dial statement after 15 seconds)?
 
 Either this is a very simple question or I'm missing something...
 
 Wouldn't something like this work for you?
 
 [incoming-bri-one]
 exten = s,1,SetCIDName(Company 1)
 exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones
 exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; company 
 1's  2's phones exten = s,4,Voicemail(su200)
 
 [incoming-bri-two]
 exten = s,1,SetCIDName(Company 2)
 exten = s,2,Dial(SIP/300SIP/301etc.,15) ; company 2's phones
 exten = s,3,Dial(SIP/300SIP/301SIP/200SIP/201etc.,15) ; comapny 
 2's  1's phones exten = s,4,Voicemail(su300)
 
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AW: [Asterisk-Users] AGI Problem

2005-10-17 Thread René Enskat [Teamware GmbH]

I have the phpagi 2 library too.
So what did you change in details there to mute the vebrose things?




 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Im Auftrag von Obelix
 Gesendet: Montag, 17. Oktober 2005 11:02
 An: Asterisk Users Mailing List - Non-Commercial Discussion
 Betreff: Re: [Asterisk-Users] AGI Problem

 Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:

 In my experience most AGI problems I had came from other info
 sent to the terminal via verbose commands and other stdout
 output. There is some info on the voip-info wiki about using AGI.

 I use the phpagi 2 library, and carefully setting up the
 agi-verbose commmands fixes my 510 problems

 
  Hmm still have problems with the get variable with PHP i have this
  error now separated with a script:
 
  Sending string GET VARIABLE CALLERIDNUM\n to Asterisk...
  Wroten bytes to STDOUT: 25
 
  Reading 80 bytes response from Asterisk...
  Received response: 510 Invalid or unknown command
 
 
 
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Re: [Asterisk-Users] Routing landline calls to asterisk.

2005-10-17 Thread Peter Ankerstål

On Mon, 17 Oct 2005 00:05:39 -0400
Tom Rymes [EMAIL PROTECTED] wrote:

 Your other options include FXO gateways like the sipura 3000 9which  
 is an ATA, too), Digium TDM400p PCI card, or a T1 card and a channel  
 bank.
 
 The appropriate piece of equipment depends on the number of lines you  
 will need.
 
Ok, but if I get a ISDN-modem there will be no problem? I can easily get one
for about $5 or something..

-- 
MVH
Peter Ankerstål.
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[Asterisk-Users] AVM B1

2005-10-17 Thread Steve Foy

Hi,

I'm trying to get Asterisk working with the AVM B1 card. I've tried every
instruction set I can find, but to no avail.

I think I'm getting mixed up as to what Zaptel/CAPI configuration to use.

If someone is currently using one of these cards, would it be possible to
mail me a few bits of the important config files?

Sorry if I should have included something else too, but let me know and I'll
provide it!

Steve


My capi.conf is as follows:

# card  fileproto   io  irq mem cardnr  options
#b1isa  b1.t4   DSS10x150   7   -   -   P2P
b1pci   b1.t4   DSS1-   -   -   -
c4  c4.bin  DSS1-   -   -   -
c4  -   DSS1-   -   -   -
c4  -   DSS1-   -   -   -   P2P
c4  -   DSS1-   -   -   -   P2P
#c2 c2.bin  DSS1-   -   -   -
#c2 -   DSS1-   -   -   -
#t1isa  t1.t4   DSS10x340   9   -   0
#t1pci  t1.t4   DSS1-   -   -   -
#fcpci  -   -   -   -   -   -
#fcclassic  -   -   0x150   10  -   -


CAPI seems to be installed correctly:

talky:/home/steve# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-03  (49.19)
Serial Number: 3904932
BChannels: 2
Global Options: 0x0039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x401f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
B2 protocols support: 0x0b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x803f
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions

  0100
  0200
  3900
  1f40
  1b0b
  3f80
       
  0101 0002   

Supplementary services support: 0x03ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS
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Re: AW: [Asterisk-Users] AGI Problem

2005-10-17 Thread Obelix

Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:

What I normally do now with agi-verbose is to pass it a variable using
print_r($outputvariable, true).

thus if I want to output a string  or an array of some sort it goes out in
the form

$output = 

$agi-verbose(print_r($output, true))

What I suggest now is to suppress screen output as much as you can and see if
the 510 errors go away.

I also realised after using phpagi 1 before that the variable hashes had changed
in phpagi 2. So if you are adapting some code from phpagi 1 check the hashes. Do
a print_r on the result variables and see if the hashes are what you expect them
to be.

 I have the phpagi 2 library too.
 So what did you change in details there to mute the vebrose things?




  -Ursprüngliche Nachricht-
  Von: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Im Auftrag von Obelix
  Gesendet: Montag, 17. Oktober 2005 11:02
  An: Asterisk Users Mailing List - Non-Commercial Discussion
  Betreff: Re: [Asterisk-Users] AGI Problem
 
  Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:
 
  In my experience most AGI problems I had came from other info
  sent to the terminal via verbose commands and other stdout
  output. There is some info on the voip-info wiki about using AGI.
 
  I use the phpagi 2 library, and carefully setting up the
  agi-verbose commmands fixes my 510 problems
 
  
   Hmm still have problems with the get variable with PHP i have this
   error now separated with a script:
  
   Sending string GET VARIABLE CALLERIDNUM\n to Asterisk...
   Wroten bytes to STDOUT: 25
  
   Reading 80 bytes response from Asterisk...
   Received response: 510 Invalid or unknown command
  
  
  
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[Asterisk-Users] AstBill-0.9.0.7 with many new features Released

2005-10-17 Thread Are

The AstBill project has released version 0.9.0.7 of its open-source billing and VOIP management platform for Asterisk. There are many new features in this release.
AstBill 0.9.0.7 is also a maintenance release that fixes problems reported using the forums and the bug tracking system.
We STRONGLY recommend you to update to the latest version as
AstBill. The software is under a very fast development schedule mainly
thanks to feedback from the fast growing user community.
AstBill is a Web Based Billing, Routing and Management Software for
Asterisk and VOIP based on Drupal and MySQL. AstBill Provides pre and
post Paid VOIP Billing Services. The aim of AstBill is to completely
automate Asterisk, call management and VOIP billing from start to
finish.
Key benefits is Open Source, Credit Control on outgoing calls, ease
of use and the User Management and call routing module. AstBill is
fully themeable and skinable.
AstBill is not only a web-based, user friendly billing interface for
Asterisk and VOIP. It is also an Asterisk configuration and GUI
management tool and a standardized implementation of Asterisk using
REALTIME and static configuration as you please.
There is also an AstBill Live CD available. This allows you to run Asterisk and AstBill from your CD drive. No installation needed.
Some of the new and improved features of AstBill-0.9.0.7
Implemented full support for H323Improved web interface for accounts management. You can now choose
between DISABLED, REALTIME, STATIC and ANI/CLI authentication.Improvements in web interface Fixed problems reported using the forums and the bug tracking system.Improved Debug output on Perl agi scripts
Minor Update to MySQL database schemaUpdated extensions.conf added example used when Asterisk and AstBill integrates with SERImplemented stronger caller authentication securityImproved Multi Tenant functionality
Rate Table in Currency of choiceCall Data Records including cost of each call and time based billingCall Data Records in his Currency of choiceSwitchboard (Displays live status of user's phones and ongoing calls)
Allows one click calling from GUI and direct to phoneAre Casilla
http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants
http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP
AstBill DEMO: http://demo.astbill.com

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Re: [Asterisk-Users] Pass variable to context (NOT macro)


Samy Antoun wrote:

--- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:


That is the default.  Once you set a variable it should exist for the
life of the channel.  Now, if you are wanting to access that variable
when one channel spawns another channel (like chan_local does), then 
prefix the name of the variable by _ or __ .  I'm sure it's
documented 
somewhere, but I have no idea where.  The underscore and double 
underscore prefix feature is NOT available in 1.0.x, only in CVS-HEAD



Eric,
If I have two contexts:

[context1]
exten = s,1,Answer
exten = s,2,SetVar(MYVAR=1)
exten = s,3,Goto(context2,s,1)
[context2]
exten = s,1,NoOp(${MYVAR})

The NoOp in context2 will return 1?


It should.

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[Asterisk-Users] ooh323c and calls to pri

2005-10-17 Thread Coufal Bohuslav

Hi I have a trouble with calls coming form ooh323c channels and going to PRI. 
This calls are rejected by telecom. Incoming calls form PRI and going to 
ooh323c works good. When i spoke with man on telecom thay said to me that 
there is wrong in something called information element. Does anybody knows if 
i can change some values for it or what i can do.

Thanks,

Bob.
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[Asterisk-Users] integrating asterisk smoothly

Hello List!

I would like to integrate a Asterisk box in my current (german) telephone
setup. Right now it looks like this:

Provider -- DSL/ISDN Splitter -- Telephone-System(Box) or TK-Anlage ;)

I have read that you can put Asterisk between my Splitter and the
Telephone-System-Box, so that for the beginning asterisk will just forward
the incoming and outgoing calls. The rest of the original Telephone Setup
should stay the same.

I think i will need a TDM400P for this.

On the FXO Port i will have to plug in the telephone line coming from my
DSL/ISDN Splitter.

Is this correct so far?

If yes, how do i get from here to my Telephone-System(Box) also known as 
TK-Anlage.(whats TK-Anlage translated correctly?)


Thanks, Mario

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Re: [Asterisk-Users] Pass variable to context (NOT macro)

2005-10-17 Thread Bob Goddard

On Monday 17 Oct 2005 04:11, Kevin P. Fleming wrote:
 Samy Antoun wrote:
  [context1]
  exten = s,1,Answer
  exten = s,2,SetVar(MYVAR=1)
  exten = s,3,Goto(context2,s,1)
  [context2]
  exten = s,1,NoOp(${MYVAR})
 
  The NoOp in context2 will return 1?

 Variables are set on the channel itself, they aren't related to contexts
 at all.

There are 2 channels involved though, are there not,
source and destination?

I'd like to see these variables set for the source channel,
not the destination. To me, that seems more logical, especially
when dialling multiple phones at once.


B
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Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread Troy Settle




Julian Lyndon-Smith wrote:

Tom Rymes wrote:

That's a good idea, but it does not help when the agent receives a  
call from the queue. If an agent has call-waiting enabled (at least  
on our 7940 Ciscos...) the queue will send another incoming call  
while the agent is still on the phone withthe last call sent to them  
from the queue.


Is that not the case? Have I misconfigured something?



The Queue should not be sending a call to an agent that is marked as 
paused, that is what the pause was desigined for. Are you using more 
than 1 queue with the same agent ?


When accepting a call from the queue, what mechanism is there to pause 
the queue member?


Yes, it's possible to pause the agent when she places an outbound call 
or when recieving a direct-dialed or extention-dialed call, but how do 
you pause the agent when she accepts a call from the queue?


To the OP:

We too use Cisco 7940s for our office, and what I ended up doing, was 
turning off call waiting completely, then using the first line 
appearance for the user's actual extension, and the second line 
appearance for the call queue.  It's just as annoying as call waiting 
without getting slammed by queue calls.



--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  866.477.5638


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Re: [Asterisk-Users] Asterisk and Fedora

2005-10-17 Thread Rudolf Ladyzhenskii

Here is a link to get you going:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

Rudolf

- Original Message - 
From: Luke Kearney [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Sunday, October 16, 2005 11:20 PM
Subject: [Asterisk-Users] Asterisk and Fedora

Hello List,
I have been beating my head against the wall for a little while now  
trying to get my TDM400 card to work with Fedora Core 4. Not a great  
deal of success even after successful builds of zaptel and the other  
required componentry. The machine doesn't even recognize the card.  
Using Debian it was recognised but as a newbie to Asterisk most of  
the documentation I can find and understand is written with RH or  
Fedora in mind it seems. Deb does things ever so slightly  
differently. I read on the list a little while ago comments that  
indicated that Fedora Core 4 was not suitable and am contemplating  
going back to Fedora Core 3. The hardware is pretty generic Intel  
Pentium 4 based hardware. Has anyone had good experiences with Fed  
Core 3 ? Or is it something to stay away from ?

Kind Regards,

-
Luke Kearney
 [EMAIL PROTECTED] 

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Re: [Asterisk-Users] ooh323c and calls to pri

2005-10-17 Thread Coufal Bohuslav

The next information is that calls send from ooh323 to PRI has packet mode and 
it shall be circuit.

Bob.

P.S. - I did use old H323 driver form asterisk up to now and it works fine.

Dne pondělí 17 říjen 2005 13:10 Coufal Bohuslav napsal(a):
 Hi I have a trouble with calls coming form ooh323c channels and going to
 PRI. This calls are rejected by telecom. Incoming calls form PRI and going
 to ooh323c works good. When i spoke with man on telecom thay said to me
 that there is wrong in something called information element. Does anybody
 knows if i can change some values for it or what i can do.

 Thanks,

 Bob.
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[Asterisk-Users] fax - conversion problem

I am having a strange problem.

On one * box I setup the fax recive, via spandsp -app_rxfax

I have no problem here.

On a second box I did the same. The resulting PDF appear corrupt.

If I transmit the same fax to both * box, the tiff files received are the
same.

A deeper analysis shows the only problem is the width and heigth of the
document

In the first PDF, I see
/Type/Page/MediaBox [0 0 595 842]./Rotate 0/Parent 3 0

In the second PDF, I see
/Type/Page/MediaBox [0 0 8.5 11]./Rotate 0/Parent 3 0

If in the second  PDF, I replace the width and height according to the
first, it becomes OK

So it seeems that the second file does not convert the width and height
information (8.5 X 11 inches) in pixels

The fist box is a Suse Linux 8.1
The second box is a Suse Linux 9.2

the producers are
/Producer(ESP Ghostscript 7.05) on the first box
/Producer(ESP Ghostscript 7.07) on the second box

The configured language is u.s english on both boxes

any help will be greatly appreciated,

thanks in advance,

Andrea



Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [Asterisk-Users] ooh323c and calls to pri

2005-10-17 Thread Coufal Bohuslav

And the next information is that in header of call is information about 
transfer rate zero and should be 64k (codec ulaw).

Bob.

Dne pondělí 17 říjen 2005 13:36 Coufal Bohuslav napsal(a):
 The next information is that calls send from ooh323 to PRI has packet mode
 and it shall be circuit.

 Bob.

 P.S. - I did use old H323 driver form asterisk up to now and it works fine.

 Dne pondělí 17 říjen 2005 13:10 Coufal Bohuslav napsal(a):
  Hi I have a trouble with calls coming form ooh323c channels and going to
  PRI. This calls are rejected by telecom. Incoming calls form PRI and
  going to ooh323c works good. When i spoke with man on telecom thay said
  to me that there is wrong in something called information element. Does
  anybody knows if i can change some values for it or what i can do.
 
  Thanks,
 
  Bob.
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Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread J Thomas

Using Asterisk agents.

Not recognizing that an agent has made an outgoing call IS THE PROBLEM.
Only workaround I see is to take the agent out of queue on all outgoing
(and direct dialed incoming) calls and put him back in the queue at the
completion of the call. That seems too kloodgy.

Hence the proper behavior has to come through feature request only.

-- jt

On Mon, 2005-10-17 at 04:30, Lenz wrote:
 Hello,
 are you using Asteriks agents or dialing straight to extensions? because  
 if you are using agents for incoming calls and then you dial straight  
 out of Asterisk, Asterisk will not know that the agent is busy. One  
 possible workaround would be to make a call to the agent using a .call  
 file, so that the agent is busy and the queue system recognizes it.
 (It's just an idea, I have never tried this)
 Thanks
 l.


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Re: [Asterisk-Users] Pass variable to context (NOT macro)


Bob Goddard wrote:

On Monday 17 Oct 2005 04:11, Kevin P. Fleming wrote:


Samy Antoun wrote:


[context1]
exten = s,1,Answer
exten = s,2,SetVar(MYVAR=1)
exten = s,3,Goto(context2,s,1)
[context2]
exten = s,1,NoOp(${MYVAR})

The NoOp in context2 will return 1?


Variables are set on the channel itself, they aren't related to contexts
at all.



There are 2 channels involved though, are there not,
source and destination?

I'd like to see these variables set for the source channel,
not the destination. To me, that seems more logical, especially
when dialling multiple phones at once.


When dealing with channel variables, you can consider both legs of a 
call to be one channel

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Re: [Asterisk-Users] AVM B1


Steve Foy schrieb:

Hi,

I'm trying to get Asterisk working with the AVM B1 card. I've tried every
instruction set I can find, but to no avail.


You should use the chan_capi or chan_capi-cm. I used to use an old B1 
ISA card, which worked without much trouble, after I got the CAPI itself 
running. Asterisk ontop of CAPI worked just fine.


You don't need to configure anything for the Zaptel, except if you want 
to use Asterisk functionality that needs a timing source. But that 
should be the second step.

--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA

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Re: [Asterisk-Users] integrating asterisk smoothly


[EMAIL PROTECTED] schrieb:

Hello List!

I would like to integrate a Asterisk box in my current (german) telephone
setup. Right now it looks like this:

Provider -- DSL/ISDN Splitter -- Telephone-System(Box) or TK-Anlage ;)

I have read that you can put Asterisk between my Splitter and the
Telephone-System-Box, so that for the beginning asterisk will just forward
the incoming and outgoing calls. The rest of the original Telephone Setup
should stay the same.

I think i will need a TDM400P for this.


Depends on your current PBX (TK-Anlage). If it is connected to regular 
ISDN line (Mehrgeräteanschluss) you need a BRI-type ISDN card (Fritz or 
HFC-S, if you need more lines, quadbri or octobri come to mind).

--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA

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[Asterisk-Users] RxFax dropping line

2005-10-17 Thread Paul Hussein



Hi, I am running a build of [EMAIL PROTECTED] with asterisk 1.2beta1 and am 
trying to diagnose RxFax with a Voip incoming trunk. I am running the 
latest spandsp and rxfax with libtiff 3.7.


Switching on debug IU can see the call come in, but after a small time 
the fax connection drops and the sending fax (paper doc ) has not moved 
in the machine.


I guess it must be dropping in the negotiation prelim stuff.

I do have some rtp stuff about time differences and some stuff about 
converting from a slin codec to ulaw and dropped frames, I just dont 
know what is significant if anything.


Can someone help me to diagnose why the line is just dropping, I have 
included the debug below.


Cheers

Paul.




Oct 17 11:10:24 DEBUG[3088] pbx.c: Launching 'RxFAX'
Oct 17 11:10:24 VERBOSE[3088] logger.c: -- Executing 
RxFAX(SIP/6969021653-dc67, 
/var/spool/asterisk/fax/1129540223.28.tif) in new stack
Oct 17 11:10:24 DEBUG[3088] channel.c: Set channel SIP/6969021653-dc67 
to read format slin
Oct 17 11:10:24 DEBUG[3088] channel.c: Set channel SIP/6969021653-dc67 
to write format slin
Oct 17 11:10:24 NOTICE[3088] channel.c: Dropping incompatible voice 
frame on SIP/6969021653-dc67 of format slin since our native format has 
changed to ulaw

Oct 17 11:10:27 DEBUG[3088] rtp.c: Difference is 23528, ms is 2961
Oct 17 11:10:30 DEBUG[3088] rtp.c: Difference is 22416, ms is 2822
Oct 17 11:10:31 DEBUG[3088] rtp.c: Difference is 1920, ms is 260
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Allocating new SIP dialog for 
[EMAIL PROTECTED] - REGISTER (No RTP)
Oct 17 11:10:35 DEBUG[3088] acl.c: # Testing 80.87.16.11 with 
192.168.0.0
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Target address 80.87.16.11 is 
not local, substituting externip
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Scheduled a registration timeout 
for sip.vira.it id  #420
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: REGISTER sip:sip.vira.it 
SIP/2.0 (32)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Via: SIP/2.0/UDP 
82.106.48.104:5060;branch=z9hG4bK683546c8 (58)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: From: 
sip:[EMAIL PROTECTED];tag=as0edb555e (49)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: To: 
sip:[EMAIL PROTECTED] (32)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Call-ID: 
[EMAIL PROTECTED] (54)

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: CSeq: 104 REGISTER (18)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: User-Agent: Asterisk PBX 
(24)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Authorization: Digest 
username=6969021653, realm=asterisk, algorithm=MD5, 
uri=sip:sip.vira.it, nonce=3d5b52e3, 
response=8fa16c4b34cdbed367257edb0cac9fa1, opaque= (173)

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Expires: 120 (12)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Contact: 
sip:[EMAIL PROTECTED] (39)

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Event: registration (19)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Content-Length: 0 (17)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header:  (0)
Oct 17 11:10:35 VERBOSE[3088] logger.c: REGISTER attempt 1 to 
[EMAIL PROTECTED]

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: SIP/2.0 100 Trying (18)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Via: SIP/2.0/UDP 
82.106.48.104:5060;branch=z9hG4bK683546c8;received=82.106.48.104;rport=5060 
(92)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: From: 
sip:[EMAIL PROTECTED];tag=as0edb555e (49)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: To: 
sip:[EMAIL PROTECTED] (32)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Call-ID: 
[EMAIL PROTECTED] (54)

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: CSeq: 104 REGISTER (18)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: User-Agent: Asterisk PBX 
(24)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Allow: INVITE, ACK, 
CANCEL, OPTIONS, BYE, REFER, NOTIFY (55)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Contact: 
sip:[EMAIL PROTECTED] (37)

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Content-Length: 0 (17)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header:  (0)Oct 17 11:10:35 
DEBUG[3088] chan_sip.c: (Provisional) Stopping retransmission (but 
retaining packet) on '[EMAIL PROTECTED]' 
Request 104: Found
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: SIP/2.0 401 Unauthorized 
(24)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Via: SIP/2.0/UDP 
82.106.48.104:5060;branch=z9hG4bK683546c8;received=82.106.48.104;rport=5060 
(92)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: From: 
sip:[EMAIL PROTECTED];tag=as0edb555e (49)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: To: 
sip:[EMAIL PROTECTED];tag=as33f90cb4 (47)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Call-ID: 
[EMAIL PROTECTED] (54)

Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: CSeq: 104 REGISTER (18)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: User-Agent: Asterisk PBX 
(24)
Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Header: Allow: INVITE, ACK, 
CANCEL, OPTIONS, BYE, REFER, NOTIFY (55)
Oct 17 11:10:35 DEBUG[3088]

RE: [Asterisk-Users] ooh323c and calls to pri

2005-10-17 Thread Bohuslav Coufal

Does anybody has more information about internal structure of ooh323c and 
should tell me how can i setup startup information about transfer rate of call?

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Coufal Bohuslav
Sent: Monday, October 17, 2005 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ooh323c and calls to pri

And the next information is that in header of call is information about 
transfer rate zero and should be 64k (codec ulaw).

Bob.

Dne pondělí 17 říjen 2005 13:36 Coufal Bohuslav napsal(a):
 The next information is that calls send from ooh323 to PRI has packet mode
 and it shall be circuit.

 Bob.

 P.S. - I did use old H323 driver form asterisk up to now and it works fine.

 Dne pondělí 17 říjen 2005 13:10 Coufal Bohuslav napsal(a):
  Hi I have a trouble with calls coming form ooh323c channels and going to
  PRI. This calls are rejected by telecom. Incoming calls form PRI and
  going to ooh323c works good. When i spoke with man on telecom thay said
  to me that there is wrong in something called information element. Does
  anybody knows if i can change some values for it or what i can do.

  Thanks,

  Bob.
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Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread Julian Lyndon-Smith


Ah, ok, I see what you are getting at.

However, could you not run a macro on connection (to the agents device) 
that then pauses the queue member so that no more calls will come 
through until they are unpaused ?


Julian.

Troy Settle wrote:




Julian Lyndon-Smith wrote:


Tom Rymes wrote:

That's a good idea, but it does not help when the agent receives a  
call from the queue. If an agent has call-waiting enabled (at least  
on our 7940 Ciscos...) the queue will send another incoming call  
while the agent is still on the phone withthe last call sent to 
them  from the queue.


Is that not the case? Have I misconfigured something?




The Queue should not be sending a call to an agent that is marked as 
paused, that is what the pause was desigined for. Are you using 
more than 1 queue with the same agent ?



When accepting a call from the queue, what mechanism is there to pause 
the queue member?


Yes, it's possible to pause the agent when she places an outbound call 
or when recieving a direct-dialed or extention-dialed call, but how do 
you pause the agent when she accepts a call from the queue?


To the OP:

We too use Cisco 7940s for our office, and what I ended up doing, was 
turning off call waiting completely, then using the first line 
appearance for the user's actual extension, and the second line 
appearance for the call queue.  It's just as annoying as call waiting 
without getting slammed by queue calls.





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AW: AW: [Asterisk-Users] AGI Problem

2005-10-17 Thread René Enskat [Teamware GmbH]

Hmm sorry can't follow you in the way.
You can say me how i have to change my script to that what do you mea?

#!/usr/bin/php -q

?php
include(/var/lib/asterisk/agi-bin/phpagi.php);
$agi = new AGI();

$ID = $agi-get_variable(SIPUSER);
if ($ID[result] == 0) {
$agi-verbose(SIPUSER not set -- nothing to do);
   exit(1);
}
$number = $ID[data];

$agi-set_variable(MSN, exec(/var/lib/asterisk/agi-bin/msn4sip 111
222 333 $number));
?



 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Im Auftrag von Obelix
 Gesendet: Montag, 17. Oktober 2005 12:29
 An: Asterisk Users Mailing List - Non-Commercial Discussion
 Betreff: Re: AW: [Asterisk-Users] AGI Problem

 Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:

 What I normally do now with agi-verbose is to pass it a
 variable using print_r($outputvariable, true).

 thus if I want to output a string  or an array of some
 sort it goes out in the form

 $output = 

 $agi-verbose(print_r($output, true))

 What I suggest now is to suppress screen output as much as
 you can and see if the 510 errors go away.

 I also realised after using phpagi 1 before that the variable
 hashes had changed in phpagi 2. So if you are adapting some
 code from phpagi 1 check the hashes. Do a print_r on the
 result variables and see if the hashes are what you expect them to be.

  I have the phpagi 2 library too.
  So what did you change in details there to mute the vebrose things?
 
 
 
 
   -Ursprüngliche Nachricht-
   Von: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] Im Auftrag von
   Obelix
   Gesendet: Montag, 17. Oktober 2005 11:02
   An: Asterisk Users Mailing List - Non-Commercial Discussion
   Betreff: Re: [Asterisk-Users] AGI Problem
  
   Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:
  
   In my experience most AGI problems I had came from other
 info sent
   to the terminal via verbose commands and other stdout
 output. There
   is some info on the voip-info wiki about using AGI.
  
   I use the phpagi 2 library, and carefully setting up the
   agi-verbose commmands fixes my 510 problems
  
   
Hmm still have problems with the get variable with PHP
 i have this
error now separated with a script:
   
Sending string GET VARIABLE CALLERIDNUM\n to Asterisk...
Wroten bytes to STDOUT: 25
   
Reading 80 bytes response from Asterisk...
Received response: 510 Invalid or unknown command
   
   
   
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RE: [Asterisk-Users] Delayed ringing on some SIP phones

  Wouldn't something like this work for you?
  [incoming-bri-one]
  exten = s,1,SetCIDName(Company 1)
  exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones
  exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; 
  company 1's  2's phones
  exten = s,4,Voicemail(su200)
  [incoming-bri-two]
  exten = s,1,SetCIDName(Company 2)
  exten = s,2,Dial(SIP/300SIP/301etc.,15) ; company 2's phones
  exten = s,3,Dial(SIP/300SIP/301SIP/200SIP/201etc.,15) ; 
  comapny 2's  1's phones
  exten = s,4,Voicemail(su300)
 
 Thanks for the replies folks.
 
 My concern is that the SIP phones in question (GXP-2000s) tend to take a
 second or two to realise they're no longer ringing. If phones are ringing
 from the first dial statement and still think they're ringing when the
 second dial statement is executed, they will all report busy to asterisk and
 not ring at all.
 
 I suppose I could insert a Wait(2) or something like that between the two
 dial statements, but I can see it causing problems with users picking the
 phone up and finding nothing on the end, then when the second dial kicks in,
 their phone reports busy because it's off-hook. What I'm really after is a
 method of starting the second dial on company 2's phones without
 interrupting the dial on company 1's phones.

If one thinks about how sip phones are actually caused to ring (eg, sending
a sip packet with Ring in it), and think about the ring timing sequence
implemented within the phone itself, then one should be able to pick a
timeout value (probably not 15 seconds) to minimize the above probability.

Example... in the US the ring cycle is approximately seven seconds. If the
timeout is set to 15 seconds as opposed to 12 seconds, the probability of
picking up the phone during that unwanted period is different.

Pure guess on my part given the internal speed of asterisk, but I'd suggest
that time interval from leaving one dial statement and moving into the
seocnd is very likely measured in milliseconds. That would imply the user
would have to pick up the phone at just the exact moment covered by that
duration for the unwanted condition to occur. Since the phone system (and
people) are not very busy (one pstn line each), this would appear to be
something that technical folks would argu forever with no real business-
world impact. Even if it did impact the user, call pickup would address it.

Am I really all that far off base with that thought process?


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Re: [Asterisk-Users] pri release cause code mismatch

2005-10-17 Thread Michael Toop


Hi Guys,

Been following this discussion, do you think these triggers could be a 
cause of our Nortel dropping some of our digits? ...keep getting: -- 
ACKing all packets from 118 to (but not including) 119  ..on an intense 
debug?


Thanks,

Michael

Johann Steinwendtner wrote:


Tirpák Miklós schrieb:



Yes. 34 is required by the Nortel to send the call to an alternative 
destination.




Cause 38 or 42 triggers the rerouting also for both options.

Hans

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Fax  011 656 1342
Mobile  083 364 2370
Web  www.bizcall.co.za

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Re: [Asterisk-Users] fax - conversion problem

The problem is in the tiff2ps, not in the ps2pdf.
I found that if I remove the -h and -w parameter everything is OK

in extensions.conf I replaced :

[ext-fax]
exten = s,1,Answer
exten = s,2,Goto(in_fax,1)
exten = in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1)
exten = in_fax,2,Macro(faxreceive)
;exten = in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf -
${FAXFILE}.pdf) ; this line does not work
exten = in_fax,3,system(tiff2ps -2eaz ${FAXFILE} | ps2pdf -
${FAXFILE}.pdf) ; this line is ok

Actually I don't know what is the problem: this is a workaround

Andrea



   
 [EMAIL PROTECTED] 
 .it   
 Sent by:   To 
 asterisk-users-bo asterisk-users@lists.digium.com 
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   [Asterisk-Users] fax - conversion   
 17/10/2005 13.41  problem 
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




I am having a strange problem.

On one * box I setup the fax recive, via spandsp -app_rxfax

I have no problem here.

On a second box I did the same. The resulting PDF appear corrupt.

If I transmit the same fax to both * box, the tiff files received are the
same.

A deeper analysis shows the only problem is the width and heigth of the
document

In the first PDF, I see
/Type/Page/MediaBox [0 0 595 842]./Rotate 0/Parent 3 0

In the second PDF, I see
/Type/Page/MediaBox [0 0 8.5 11]./Rotate 0/Parent 3 0

If in the second  PDF, I replace the width and height according to the
first, it becomes OK

So it seeems that the second file does not convert the width and height
information (8.5 X 11 inches) in pixels

The fist box is a Suse Linux 8.1
The second box is a Suse Linux 9.2

the producers are
/Producer(ESP Ghostscript 7.05) on the first box
/Producer(ESP Ghostscript 7.07) on the second box

The configured language is u.s english on both boxes

any help will be greatly appreciated,

thanks in advance,

Andrea



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Re: [Asterisk-Users] Ask for config files of Nortell Meridian Op 11 Asterisk for PRI

2005-10-17 Thread Michael Toop





Hi,

There is a page on the wiki:
(http://www.voip-info.org/wiki-Asterisk+legacy+integration) a .pdf
for this. We are also trying to get it right  struggeling. Let
me know if you get it right! ...would love to hear how you did it.

Kind Regards,

Michael

Alvaro Parres wrote:

  Hi list, any one can let me his config files for interconecting
a Meridian Op 11 and Asterisk
  via a E1 PRI CARD. 
  
  Actually i need the nortell config part, becouse my client
nortell provider doesn't know
  how to config the PRI card at his part.
  
  Thanks all.
  
  
  

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Tel  011 602 9300
Fax  011 656 1342
Mobile  083 364 2370
Web  www.bizcall.co.za


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[Asterisk-Users] module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format

2005-10-17 Thread Giorgio Incantalupo


Hi,
I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) 
using gcc 4.0.2.
Compilation does not give me errors so after a 'make install' I try to 
load zaptel module with insmod but the following error arise:


*insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid 
module format*

Is there anybody who can help me??

TIA

Giorgio

--


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Tel. +39 02 9350 4780 (104)

FGA Software
20017 Rho - Via Puccini, 8

E-Mail :
[EMAIL PROTECTED]
Internet:
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Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread Lenz



Yes, but if an agent is connected through the Agent module even on an
outboiund call, * will consider it unavailable and therefore will not
route calls to it, as if the agent was answering some inbound call.
Just my $0.02 :-)
l.



On Mon, 17 Oct 2005 14:01:27 +0200, J Thomas [EMAIL PROTECTED] wrote:


Using Asterisk agents.

Not recognizing that an agent has made an outgoing call IS THE PROBLEM.
Only workaround I see is to take the agent out of queue on all outgoing
(and direct dialed incoming) calls and put him back in the queue at the
completion of the call. That seems too kloodgy.

Hence the proper behavior has to come through feature request only.

-- jt

On Mon, 2005-10-17 at 04:30, Lenz wrote:

Hello,
are you using Asteriks agents or dialing straight to extensions? because
if you are using agents for incoming calls and then you dial straight
out of Asterisk, Asterisk will not know that the agent is busy. One
possible workaround would be to make a call to the agent using a .call
file, so that the agent is busy and the queue system recognizes it.
(It's just an idea, I have never tried this)
Thanks
l.









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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)


tim panton wrote:

By the way, there is a reason for this. It ensures that there is  
traffic (initiated by the client) often
enough to keep the 'connection' in a NATing firewall's map of ports.  
This means that a
'new' call (ie incoming) message from asterisk to the client will be  
seen by the firewall as part of that

'recent' conversation and allowed through (and correctly forwarded).


Ostensibly that was the reason, yes, but it's flawed... 'qualify' is 
much better for that purpose, for three reasons:


1) It is initiated from the server end instead of the peer end, so there 
is no chance the firewall will drop the association.
2) It is far less work on the server; registrations require 
authentication and database updates.
3) It will also make your Asterisk server aware of when the peer becomes 
unreachable.


Personally, I'd recommend changing the minexpiry time to something like 
300 seconds or longer, and using 'qualify' to keep the NAT mapping alive.

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Re: [Asterisk-Users] module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format

2005-10-17 Thread Giorgio Incantalupo

Sorry, I forgot to say I'm using Asterisk 1.2.0-beta and the same are 
the zaptel and libpri version.


Giorgio

Giorgio Incantalupo wrote:


Hi,
I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x 
kernel) using gcc 4.0.2.
Compilation does not give me errors so after a 'make install' I try to 
load zaptel module with insmod but the following error arise:


*insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 
Invalid module format*


Is there anybody who can help me??

TIA

Giorgio




--


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Tel. +39 02 9350 4780 (104)

FGA Software
20017 Rho - Via Puccini, 8

E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com

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[Asterisk-Users] compiling Asterisk 1.2 with zaptel and h.323

2005-10-17 Thread Lenz


Hello list,
I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with  
a TDM400 card and H.323.

You can find it at http://www.oinko.net/astrecipes/index.php?n=102

Any comment / suggestion / modification /bugfix is welcome!

I was wondering: is there any way to build a version of Bristuff for 1.2  
beta 1?


Bye for now,
l.


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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)

  By the way, there is a reason for this. It ensures that there is  
  traffic (initiated by the client) often
  enough to keep the 'connection' in a NATing firewall's map of ports.  
  This means that a
  'new' call (ie incoming) message from asterisk to the client will be  
  seen by the firewall as part of that
  'recent' conversation and allowed through (and correctly forwarded).
 
 Ostensibly that was the reason, yes, but it's flawed... 'qualify' is 
 much better for that purpose, for three reasons:
 
 1) It is initiated from the server end instead of the peer end, so there 
 is no chance the firewall will drop the association.
 2) It is far less work on the server; registrations require 
 authentication and database updates.
 3) It will also make your Asterisk server aware of when the peer becomes 
 unreachable.
 
 Personally, I'd recommend changing the minexpiry time to something like 
 300 seconds or longer, and using 'qualify' to keep the NAT mapping alive.

The only issue I see with that approach is that customers tend to buy
crap for firewalls without any knowledge/experience relative to nat
timeouts, etc. We've seen some that never timeout the nat entries (unless
the nat table becomes full), and others with very short duration timeouts.
Using the server-based qualify assumes you either know the nat table
timeout value, or, one must pick a very short duration qualify generating
wasteful traffic. 

I'm not arguing or proposing alternatives, just simply stating actual 
observations.



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[Asterisk-Users] Asterisk Busy Detect

2005-10-17 Thread Matt

Hi,
Is my CLEC feeding me a line, or is this really how it has to be?

When I dial a number on their system, if it's busy I get a busy back right away.

If I dial a number on say Verizon's system, the CLEC sends me a call
preceding event, and then I get a BUSY back from Verizon.

Should the CLEC be able to only send me ahead if Verizon says go
head and give me a BUSY if Verizon gives a BUSY?  Rather then just
blindly passing me off to Verizon?
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[Asterisk-Users] Double Ringing for PRI Calls

2005-10-17 Thread aza

Hi,

We've recently upgraded to 1.2 and for outgoing PRI calls we are now getting a
SIP 180 Ringing message generated by asterisk along with the RTP audio stream
with the PRI ring tone. This creates a double ring tone on most SIP devices
(Cisco 7960s are an exception and ignore the 180) that people find a bit
annoying.

Anybody know how to stop the 180 message being generated?

Thanks,

Aaron
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[Asterisk-Users] fax receive problem on zapata channel

I have a TE110P Digium card.
I am trying to receive fax, but the fax are almost always unreadable, i.e


==
Oct 17 15:21:20 DEBUG[4054]: Pages transferred:  1
Oct 17 15:21:20 DEBUG[4054]: Image size: 1728 x 1157
Oct 17 15:21:20 DEBUG[4054]: Image resolution7700 x 3850
Oct 17 15:21:20 DEBUG[4054]: Transfer Rate:  9600
Oct 17 15:21:20 DEBUG[4054]: Bad rows26
Oct 17 15:21:20 DEBUG[4054]: Longest bad row run 5
Oct 17 15:21:20 DEBUG[4054]: Compression type2
Oct 17 15:21:20 DEBUG[4054]: Image size (bytes)  0
Oct 17 15:21:20 DEBUG[4054]:
==
Oct 17 15:21:24 DEBUG[4054]:
==
Oct 17 15:21:24 DEBUG[4054]: Fax successfully received.
Oct 17 15:21:24 DEBUG[4054]: Remote station id: 0108680549
Oct 17 15:21:24 DEBUG[4054]: Local station id:
Oct 17 15:21:24 DEBUG[4054]: Pages transferred: 1
Oct 17 15:21:24 DEBUG[4054]: Image resolution:  7700 x 3850
Oct 17 15:21:24 DEBUG[4054]: Transfer Rate: 9600
Oct 17 15:21:24 DEBUG[4054]:
==

this is one of the best (!!) received.

I tried to put, in zapata.conf,
faxdetect=incoming

but nothing changed.
also, the callerid does not appear in the $CALLERID variable (but in
ordinary phones, not fax, it appears)

callerid=asreceived
usecallerid=yes

Is there anything I can try ? i.e. raise/decrease the RX gain ?

thanks in  advance,

Andrea

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[Asterisk-Users] SIP Phones

2005-10-17 Thread James Courtier-Dutton

Hi,

I wish to set up a simple network of about 20 SIP phones. This will be
a stand alone VoIP network, without any links to the internet or
standard PSTN networks.
For SIP phones to work, one needs a SIP server so I thought that
Asterisk might be a good choice.
Does anyone have a list of SIP IP phones that have been tested with
Asterisk and known to work reliably?
Also, when making a SIP call, does the voice bearer also pass through
Asterisk, or is it just the SIP call setup that passes through
Asterisk?
If the voice bearer passes through the Asterisk box, is it possible to
record every voice call?

James
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[Asterisk-Users] Transfering calls. Dial plan

2005-10-17 Thread Arne Morten Johansen









Hi there.



We have lots of internal phones and we use
the transfer option very often. But how do I set up the dial plan so that when a
user transfers the call to someone else and that person is unavailable/busy
(etc), the call returns to the user after a couple of seconds or so.



I was thinking about using the CallerIdNum
but the callerid is not always the number of the phone that transfers the
phone. Sometimes its the other party of the converstation.



This is basicly what i want to do

External User calls in - One of us
(person A) answers - Transfer call to correct person(B) - If person B
unavailable, transfer back to person A.





Thanks.








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Re: [Asterisk-Users] Double Ringing for PRI Calls

2005-10-17 Thread Matt

Yes,
Go into sip.conf and add this line:
progressinband=no


On 10/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hi,

 We've recently upgraded to 1.2 and for outgoing PRI calls we are now getting a
 SIP 180 Ringing message generated by asterisk along with the RTP audio stream
 with the PRI ring tone. This creates a double ring tone on most SIP devices
 (Cisco 7960s are an exception and ignore the 180) that people find a bit
 annoying.

 Anybody know how to stop the 180 message being generated?

 Thanks,

 Aaron
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Re: [Asterisk-Users] compiling Asterisk 1.2 with zaptel and h.323

2005-10-17 Thread Simone Cittadini


Lenz ha scritto:


Hello list,
I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 
with  a TDM400 card and H.323.

You can find it at http://www.oinko.net/astrecipes/index.php?n=102

Any comment / suggestion / modification /bugfix is welcome!



I've found that when you compile zaptel in debian you must link
/usr/src/kernel-headers-2.4.whatever to /usr/src/linux and
zaptel-1.2 dir to /usr/src/zaptel, and make zaptel from there or it 
won't find a lot of stuff ...
where kernel-headers-2.4.whatever is from the package specific to your 
architecture, generic deb won't do


no need to modprobe zaptel  and modprobe wctdm since zaptel is required 
by wc, just modprobe wc

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Re: [Asterisk-Users] Polycom MWI

2005-10-17 Thread Chris Coulthurst

I think I have an idea of what dto do here.  Look in your sip.cfg file for a 
line starting with MSG_WAITING under the CALLPROGRESS section.  It defines 
the tone chirp you hear for message waiting notification.   I'll bet if you 
zero out the values it would stop alerting you.


P.S. It might be in ipmid.cfg if you have that file instead

Chris Coulthurst
[EMAIL PROTECTED]


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

Sent: Monday, October 17, 2005 12:33 AM
Subject: [Asterisk-Users] Polycom MWI


Hi,

I have lookedaround and don't see this anywhere. Is there a way to
tell the ip500 to not make the aural MWI blips?
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Re: [Asterisk-Users] SIP Phones

2005-10-17 Thread Mark Phillips

Check out voip supply.com. All their SIP phone have been tested with 
Asterisk.


Asterisk can work in 2 ways when handling calls. It can set up the call 
and then step back and let the phones go peer to peer or it can stay 
involved in the call until its terminated.


Obviously the latter requires a larger CPU etc as it has more work to 
do. This is also required where there has to be code translation for 
example GM to LAW. As you are on a closed network this might not be 
relevant as you'll be using the same codes all around.


It is possible to record the calls when they pass through Asterisk This 
is very common for obvious reasons. Just make sure you have enough disk 
space.


plug
I do this for a living if you need some consultancy.
/plug

Mark

James Courtier-Dutton wrote:

Hi,

I wish to set up a simple network of about 20 SIP phones. This will be
a stand alone VoIP network, without any links to the internet or
standard PSTN networks.
For SIP phones to work, one needs a SIP server so I thought that
Asterisk might be a good choice.
Does anyone have a list of SIP IP phones that have been tested with
Asterisk and known to work reliably?
Also, when making a SIP call, does the voice bearer also pass through
Asterisk, or is it just the SIP call setup that passes through
Asterisk?
If the voice bearer passes through the Asterisk box, is it possible to
record every voice call?

James
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--

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Randolph, NJ
http://www.g7ltt.com
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[Asterisk-Users] Call transfer - atxfer

2005-10-17 Thread Andrew Nowrot

Hi,

I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:

[general]
parkext = 100
parkpos = 1-5
context = parkedcalls
parkingtime = 100
transferdigittimeout = 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark = yes
pickupexten = *8
[featuremap]
atxfer = *2
blindxfer = #
disconnect = *0
automon = *1

and when I press *2 console says something like this:

Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42
(*), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,42)
-- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50
(2), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,50)
-- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1

Does anyone know what's going on? What should I do to make attended
transfer works well?

Cheers

Andrew
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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)

2005-10-17 Thread tim panton



On 17 Oct 2005, at 15:06, Rich Adamson wrote:


By the way, there is a reason for this. It ensures that there is
traffic (initiated by the client) often
enough to keep the 'connection' in a NATing firewall's map of ports.
This means that a
'new' call (ie incoming) message from asterisk to the client will be
seen by the firewall as part of that
'recent' conversation and allowed through (and correctly forwarded).



Ostensibly that was the reason, yes, but it's flawed... 'qualify' is
much better for that purpose, for three reasons:

1) It is initiated from the server end instead of the peer end, so  
there

is no chance the firewall will drop the association.
2) It is far less work on the server; registrations require
authentication and database updates.
3) It will also make your Asterisk server aware of when the peer  
becomes

unreachable.

Personally, I'd recommend changing the minexpiry time to something  
like
300 seconds or longer, and using 'qualify' to keep the NAT mapping  
alive.




The only issue I see with that approach is that customers tend to buy
crap for firewalls without any knowledge/experience relative to nat
timeouts, etc. We've seen some that never timeout the nat entries  
(unless
the nat table becomes full), and others with very short duration  
timeouts.

Using the server-based qualify assumes you either know the nat table
timeout value, or, one must pick a very short duration qualify  
generating

wasteful traffic.

I'm not arguing or proposing alternatives, just simply stating actual
observations.



There is also the issue that if qualify ever misses a timeout (eg  
packet) and the

client's end firewall drops the map, then you will have to wait for
the next registration to initiate a new mapping since that firewall will
probably only allow new mappings to be triggered from the inside and
will ignore the server's next qualifying PING.

This is a reason not to make the registration timeout too long.

T.

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Re: [Asterisk-Users] Goto command question

2005-10-17 Thread Mohamed A. Gombolaty



Dear Eric,
You are totally right, I already know the information below
but I don't know why I couldn't see them, I certainly need a vacation,
anyway it worked like charm.
Thx
MAG
Eric \"ManxPower\" Wieling wrote:
Mohamed A. Gombolaty wrote:
> Dear All,
>
> I have this question regarding goto command, I amusing Asterisk cvs
head
> version, and I am trying to put a goto statement to send the user
to
> another extension that contains the extension he is dialing
here is how I
> am doing it :
>
> exten => 2x.,1,setgroup(outgoing)
> exten => 2x,2,checkgroup(2)
> exten => 2x.,3,goto(another-context, ${EXTEN},1)
> exten => 2x.,104,hangup
>
> but the result is always it hangs up I don't know if this goto statement
is
> correct or not, can anyone lead me to the right way to make this
statement?
First of all patterns must start with _
exten => _2X.,1,setgroup(outgoing)
Second you are using different patterns
exten => _2X.,1,setgroup(outgoing)
Is NOT the same as
exten => _2X,2,checkgroup(2)
The first pattern is _2X. the second pattern is _2X
Third, do not put spaces after commas.
Try this:
exten => _2X.,1,SetGroup(outgoing)
exten => _2X.,2,CheckGroup(2)
exten => _2X.,3,Goto(another-context,${EXTEN},1)
exten => _2X.,104,Hangup
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MAG

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Re: [Asterisk-Users] Call transfer - atxfer

Are you using 1.0.x?  DTMF Attended Transfer is not supported in 1.0.x. 
 Unless you have a brain dead phone, you should be able to use SIP 
attended transfer in 1.0.x.  (that would be the transfer key on the phone)


Andrew Nowrot wrote:

Hi,

I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:

[general]
parkext = 100
parkpos = 1-5
context = parkedcalls
parkingtime = 100
transferdigittimeout = 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark = yes
pickupexten = *8
[featuremap]
atxfer = *2
blindxfer = #
disconnect = *0
automon = *1

and when I press *2 console says something like this:

Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42
(*), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,42)
-- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50
(2), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,50)
-- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1

Does anyone know what's going on? What should I do to make attended
transfer works well?

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RE: [Asterisk-Users] Delayed ringing on some SIP phones

 Why dont you make a special extension where you could provide 
 the delay and the numbers you want to dial?
 exten = _900X,1,Wait(${EXTEN:4:2})
 exten = _900X,2,Dial(SIP/${EXTEN:5})
 then in the incoming context you could dial 
 exten = s,1,Dial(SIP/200SIP/201LOCAL/90015300LOCAL/90015301)

Just wanted to post back to the list and say this suggestion appears to work
fine - many thanks to the kind soul who suggested it.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Call transfer - atxfer

2005-10-17 Thread Andrew Nowrot

Hi,

Thank for the Email

I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?

Best wishes

Andrew
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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)


tim panton wrote:


This is a reason not to make the registration timeout too long.


Yep, that's why I suggested 5 minutes. It seems to be a reasonable 
compromise. Also keep in mind that qualify packets are sent far more 
often than the NAT timeout in most routers, so it would have to drop a 
number of packets before that would be an issue.

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Re: [Asterisk-Users] Call transfer - atxfer


Andrew Nowrot wrote:

Hi,

Thank for the Email

I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?


CVS-HEAD and 1.2Beta1 and later.
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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)


Rich Adamson wrote:


The only issue I see with that approach is that customers tend to buy
crap for firewalls without any knowledge/experience relative to nat
timeouts, etc. We've seen some that never timeout the nat entries (unless
the nat table becomes full), and others with very short duration timeouts.
Using the server-based qualify assumes you either know the nat table
timeout value, or, one must pick a very short duration qualify generating
wasteful traffic. 


Wouldn't the same be true of the registration interval though? If you 
need the NAT mapping to stay in effect, _something_ is going to have to 
generate two-way traffic before it expires... I don't see how it matters 
whether that is a registration or a qualify.

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RE: [Asterisk-Users] No Audio from Console but mpg123fromshellworksfine.

2005-10-17 Thread Jonathan k. Creasy

Thanks. I was only loading OSS. I installed the alsa development
libraries and then loaded alsa instead of oss and everything is working
now. 

Thanks!

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, October 16, 2005 9:00 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] No Audio from Console but
mpg123fromshellworksfine.

On 10/16/2005, Jonathan k. Creasy [EMAIL PROTECTED] wrote:
-Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
 Cohen
 Sent: Sunday, October 16, 2005 2:59 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from
 shellworksfine.

 Do you use ALSA or OSS for sound? What kernel version?

 ALSA. I used alsactl to reset the mixer controls as it was muted by
 default. I'm running CentOS 4.1, I don't remember the kernel version
 right off and I don't have access to that box here, I'll check it from
 work tomorrow. 

 [chan_oss.so] = (OSS Console Channel Driver)
  == Parsing '/etc/asterisk/oss.conf': Found
   == Registered channel type 'Console' (OSS Console Channel Driver)

 Asterisk grabs /dev/dsp . I figure you can't play anything at this
 point. Though you should get stuck at trying to open it.

Sigh...

From modules.conf

;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload = chan_alsa.so
;noload = chan_oss.so

Bet the problem is around here.

Brett
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RE: [Asterisk-Users] Polycom MWI

2005-10-17 Thread Andy Goss

Yes, if you look in the cfg files for the phone (either sip.cfg or
ipmid.cfg) you will see something similar to this (I use polycom 501s):

MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence se.pat.misc.1.inst.1.value=1
se.pat.misc.1.inst.2.type=silence se.pat.misc.1.inst.2.value=2
se.pat.misc.1.inst.3.type=silence se.pat.misc.1.inst.3.value=1/

I didn't bother taking out the unnecessary stuff, I just changed where
it said chord to silence, this way if I needed to bring it back I could
just change silence back to chord.

Hope this helps.

Andy

--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Chris Coulthurst
 Sent: Monday, October 17, 2005 10:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Polycom MWI
 
 I think I have an idea of what dto do here.  Look in your sip.cfg file
for
 a
 line starting with MSG_WAITING under the CALLPROGRESS section.  It
 defines
 the tone chirp you hear for message waiting notification.   I'll bet
if
 you
 zero out the values it would stop alerting you.
 
 P.S. It might be in ipmid.cfg if you have that file instead
 
 Chris Coulthurst
 [EMAIL PROTECTED]
 
 
 - Original Message -
 From: Wilson Pickett [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, October 17, 2005 12:33 AM
 Subject: [Asterisk-Users] Polycom MWI
 
 
 Hi,
 
 I have lookedaround and don't see this anywhere. Is there a way to
 tell the ip500 to not make the aural MWI blips?
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Re: [Asterisk-Users] Busy not jumping n + 101 anymore


Andrew Kohlsmith wrote:


0 : signals normal completion, and the dialplan continues
'0' - '9' or 'A' - 'F' or '#' or '*' : signals normal completion and
jump to that extension
anything else : signals failure and the call is hung up



Please explain the second result?  I don't understand.


Applications can return single-digit results to the PBX core to a simple 
extension jump, IIRC. I don't know if any applications currently use 
that feature.


I very very strongly disagree here, but I will wait for your response on my 
questions above.  The exensions.conf SPECIFICALLY states that the resultcode 
should be checked, but no way is provided.  This seems VERY counter-intuitive 
when you say that it should never be needed anyway.


Where do you see that in extensions.conf.sample? I only get one hit on 
'result', and nothing about return codes at all.

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Re: [Asterisk-Users] Restricting registration for peer '611' to 60 seconds (requested 1200)


  The only issue I see with that approach is that customers tend to buy
  crap for firewalls without any knowledge/experience relative to nat
  timeouts, etc. We've seen some that never timeout the nat entries (unless
  the nat table becomes full), and others with very short duration timeouts.
  Using the server-based qualify assumes you either know the nat table
  timeout value, or, one must pick a very short duration qualify generating
  wasteful traffic. 
 
 Wouldn't the same be true of the registration interval though? If you 
 need the NAT mapping to stay in effect, _something_ is going to have to 
 generate two-way traffic before it expires... I don't see how it matters 
 whether that is a registration or a qualify.

Sure, which is part of the logic behind a relatively short registration
period (re-opening the nat table entry). Likely the best approach to
maintain customer availability (and customer relationships) is a combination
of a relatively short registration period plus qualify.

This might be a rather poor example, but our company does a fair amount
of work with isp's and itsp's. We've purposefuly placed all customers
behind a Cisco 7206 nat router (customer's are very happy), but since some
still become infected from emails, etc, their PC's scan hundreds of IP's
on the Internet in an attempt to infect others. Since even a fully loaded
7206 will eventually run out of nat table space, we've had to reduce the
nat timeout (for udp) to 30 seconds. 

In this unusual case, a short duration qualify does handle the issue with
the exception of missed qualify attempts. When that happens, the typical
sip adapter/phone is out of service for a relatively long period. Not cool
from a customer satisfaction perspective.

Some of the Linksys products also have very short nat table timeouts.
In a recent case, a ten second qualify on a 784k residential dsl link
would occassionally be missed, and the Cisco 7960 became unreachable 
for lengthy periods of time. Cranking down both the registration and
qualify seems to have addressed it (waiting for recurrance).

It would seem on the surface that sip devices would benefit significantly
if the qualify-type function actually originated from the adapter/phone.
Then all of the above would become more of a non-issue. Obviously none of 
us can actually influence that approach, so we're kind of stuck addressing 
the issue with a combination as noted.


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RE: [Asterisk-Users] Calibrating both RX and TX gain?

2005-10-17 Thread tmassey


[EMAIL PROTECTED] wrote on 10/16/2005
07:49:38 AM:

 Here's a couple of ways to determine levels...
 
 1. using the model 4 transmission test set, attach the tone generator
 to one analog pstn line and the transmission level test jacks to a
 second pstn line. Dial from one line to other and measure the tone.
 Divide by two, and the result is the loss associated with a single
 analog pstn line from your location to your central office.

Remember, I'm not working with simple POTS lines.
I've got an Adtran TA 612 providing CO lines from a T1. There
is nothing that says that the RX and TX settings on the Adtran are the
same... Therefore, just dividing by 2 won't work.

Also, couldn't there be an issue on standard POTS
lines where the effect upon a singnal between TX and RX is different?

It seems you're just exchanging one set of assumptions
for another. But you're the expert! :)

 2. use one of those analog pstn lines to dial
the distant milliwatt
 generator (regardless of where its located), and measure the level
 of the tone. Subtract the loss determined from step #1 and now
you
 have the loss associated with facilities interconnecting your central
 office all the way to the distant milliwatt generator.

This doesn't address the problem above, correct?

 The end result will be whatever loss values you measure/calculate,
 you'll still have to play around with the rxgain  txgain to 
 minimize the echo while also maximizing the audio levels. The
 process will become a _qualitative_ eval process, not a quantitative
 one. It doesn't make any real difference which tools you use to get

 there or exactly where the milliwatt generator happens to reside.

So how important or valuable will getting a milliwatt
test number be?

Tim Massey
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[Asterisk-Users] Legacy PBX Integration and Zaptel.conf Timing Source

2005-10-17 Thread Geoff Manning

My Setup looks like this:

Mitel 200 SX (1st T1)  Bell South
 (2nd T1)
 |
 |
 |
Digium TE110P
  Asterisk


MITEL CONFIGURATION
Primary Timing Source: 1st T1 Card
Secondary Timing Source: 2nd T1 Card


ASTERISK CONFIGURATION
span=1,1,0,d4,ami (Look to the Span for timing)


We are getting a lot of Frame and Slip errors

TimeSlipFrame
7:00736 950
8:00690 1200
9:00437 762
10:00   500 913

and then the 2nd Mitel T1 card takes itself offline once the threshold
is hit (usually every 18 hours).

What is the proper way to setup timing in this scenario? I have tried
setting it both ways:

1) span=1,0,0,d4,ami (Provide timing for the span)
2) span=1,1,0,d4,ami (Look to the Span for timing)

And I get the same amount of errors either way.

(1) Asterisk set as timing source will cause problems with the 2nd T12 card
in Mitel since it will be receiving timing from the Asterisk as well as the
1st T1 card in the Mitel.

(2) Asterisk set to expect timing from the Mitel. The 2nd T1 card is
expecting timing from the 1st Mitel T1 card and the Asterisk is expecting
timing from the 2nd T1 card.

Does that sound like it could cause a problem? I don't think the Asterisk
server will try to get it's timing from the 1st Mitel T1 and I don't think
the 2nd Mitel T1 card will pass along it's timing from the 1st to the
Asterisk.

I think the solution is to remove the entry in the Mitel that sets the 2nd
T1 card as the secondary source for timing. Sound about right?

Thanks!


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RE: [Asterisk-Users] Calibrating both RX and TX gain?