Re: [Asterisk-Users] 5,000 concurrent calls system rollout question

[Leo Ann Boon]

Vic wrote:

> Hi,
>
> we are currently considering different options for rolling out a large
> scale IP PBX to handle around 3,000 + concurrent calls.
>
> Can this be done with Asterisk? Has it been done before?
>
> I really would like an input on this.
>
> Thanks!
>
>  
>
Check out Signate Telephony Server,
http://www.signate.com/pbx.php
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question

[Rob Lith]

What's you mix of calls going SIP/IAXand to PSTN?We've done  some benchmark experiments on a 3GHz HT box with 1GB of ram, mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls over 4 PRI spans. Its running
MusicOnHold into 60 of the channels, playing various GSM prompts into the other 60. The "user" cpu usage is about 25%, the "system" cpu about 25% also. We can add to that 5000 registered SIP peers and 5000 registered IAX2 peers - total of about 100 registration refreshes per second. That adds about 40% more user CPU and pretty much fills up CPU. Audio quality is still perfectly fine, and PRI slips few and far between. Load average for the whole mix sits between 5 and 10. About 550Kb/s out and 400Kb/s out on the ethernet for the registration traffic.
Also on www.voip-info.org - search for dimensioningRobOn 1/28/06, Vic <
[EMAIL PROTECTED]> wrote:
Hi,
we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls.
Can this be done with Asterisk? Has it been done before?
I really would like an input on this.
Thanks!


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Re: [Asterisk-Users] G729 Commercial Licenses.

[Rob Lith]

Read towards the bottom of http://www.digium.com/downloads/ftp/asterisk/g729/READMERobOn 1/28/06, 
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hi all,
 
I have purchased 2 licenses of G729 from digium and has done the registration. It works well and is quite fine with my 
[EMAIL PROTECTED]. Just want to clarify some licensing issues regarding them.

 
If i had to do a full reformat of my PC and reload [EMAIL PROTECTED] again will i be able to use the licenses again without re-registration?

 
If no. .Is there are limits for this? 
 
Please anyone clarify. 
 
Thanks 
 
Dan

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Re: [Asterisk-Users] Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?

[Rob Lith]

what difference does a non commercial installation make? On 1/28/06, Dean Collins <[EMAIL PROTECTED]> wrote:














Thanks but this is for a test, I didn't
buy the first one as it's a non commercial installation. I'm trying
to test bandwidth etc so I need to try out how 4 of them handle the link
simultaneously, I just don't know how to add a second test license.

 

 

Dean

 

 

 









From:

[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
[EMAIL PROTECTED]
Sent: Saturday, 28 January 2006
12:32 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?



 



Well its simple





 





Buy one more license...





 





Dan

 





On 28/01/06, Dean
Collins <[EMAIL PROTECTED]>
wrote: 

Does anyone know how to install more than one of these licenses on the
[EMAIL PROTECTED]

I installed one and works fine but of course when I try to make the second call
it says no lines are available 


Cheers,

Dean




-Original Message-
From: [EMAIL PROTECTED]
[mailto:
[EMAIL PROTECTED]] On Behalf Of Francesco Peeters
(Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: [EMAIL PROTECTED]; Asterisk Users
Mailing List - Non-Commercial Discussion 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?

On Sat, January 21, 2006 23:21, Franz Bräuer said: 
> Hi,
>
> MapsAir wrote:
>> Has anyone successfully Installing the none commercial intel g729
codecs
>> into [EMAIL PROTECTED] 2.2?
>
> Installed them today. Installing from source didn't work for me (Debian, 
> Asterisk 1.2 from svn) but just adding the binaries (see the wiki on
> voip.org) did the job. Have you already
tried the binaries?
>

Kewl! Those work like a treat! 

As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did:

cd /usr/lib/asterisk/modules/
wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so

wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so

After reloading, 'show translation' gives:
Translation times between
formats (in milliseconds) 
 Source Format (Rows)
Destination Format(Columns)

g723  
gsm  ulaw  alaw  g726 adpcm  slin
lpc10  g729 speex  ilbc
  g723
-22 8
817 8
724   115  
19897
   gsm   151
- 7
716 7
623   114   19796
  ulaw  
14616 -
111 2
118   109   19291
  alaw  
14616 1
-11 2
118   109  
19291 
  g726  
154241010
-10
926   117   20099
adpcm   14616
2 211
- 118  
109   19291
  slin  
14515 1
110 1
-17   108   19190

lpc10   161311717261716
-   124   207   106
  g729  
16939252534252441
-   215   114
speex  
16030161625161532  
123 -   105 
  ilbc  
17343292938292845  
136   219 -

Jolly good show, old chap!

--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 
   Cologne HFC-S pins #52, #54, #55 connected in parallel for
synching.
AMD Duron 1GHz - 1GB - * 1.2.1
2 Sweex HFC-PCI cards
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RE: [Asterisk-Users] DTMF's indescipherable, but voice clean!

[Nabeel Jafferali]

> After many hours today thinking that I had placed a bug into my dialplan,
> I realized that for some reason DTMF tones are simply not making it into
> asterisk! Calling into my pbx transmits crystal-clear audio in both
> directions. But dialing DTMF's from pstn->pbx is unsuccessful, while pbx-
> >pstn works fine. The tones simply don't make it through. Tiny brief
> fragments are all.

It might help to describe what interfaces your Asterisk PBX with the PSTN.
Is this a VoIP provider DID you are using, or a POTS line with an interface
card, or a PRI with a digital interface card?

Nabeel

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RE: [Asterisk-Users] Context for SIP incoming (newbie question?)

[Nabeel Jafferali]

If you have, in sip.conf, a register => blah:[EMAIL PROTECTED]/12345, you
would also have:

[blah]
…
host=sip.blah.com
context=from-blah
…

Then, in extensions.conf, you would have:

[from-blah]
exten => 12345,1,Dial(whatever)
...

Nabeel


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Mejía Evertsz
Sent: January 27, 2006 5:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Context for SIP incoming (newbie question?)

Please help me out with this…

To which context of the dial-plan does asterisk tries to match incoming
calls when acting as a sip client?
To be more specific:
In extensions.conf… Under which context should I place  “exten =>
648064,1,Dial(TECH/peer)” for an entry like this “register =>
648064:[EMAIL PROTECTED]/648064” ?

This is because I want to match one sip client to one context, and another
sip client into another context.
Is it possible?
What is the correct way to do it??

Thanks,
Alejandro

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Re: [Asterisk-Users] OT?: International number parsing

[Ron hotmail]

The short answer is no, you will never have a 
situation where the 'local' part of the term number is mistaken for part of the 
dialcode.
for example,
your customer dials 0119647701773352 (Iraq mobile 
number)
 
Iraq     
       011964
Iraq-Baghdad   0119641
Iraq-Mobile  0119647701
 
this would cause a match on Iraq, and Iraq-Mobile, 
but not on baghdad, the 'most' accurate match would be the dialcode with 
the most digits...
 
R 

  - Original Message - 
  From: 
  Damon Estep 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, January 27, 2006 11:56 
  PM
  Subject: RE: [Asterisk-Users] OT?: 
  International number parsing
  
  
  Have you seen 
  situations where a portion of the local number, when added to the country and 
  city code, result in a longer match then the actual country/city called and 
  therefore an inaccurate match?
   
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Ron hotmailSent: Friday, January 27, 2006 8:17 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] OT?: 
  International number parsing
   
  
  It realy is a pain in the 
  *ss.
  
  the problem is just how you 
  explained.  when trying to match the terminating number, there's no 
  SINGLE fixed pattern for the dialcodes.  so how do you know how many 
  digits of the term number to match against the dialcode? you dont.  you 
  have to match the dialcodes against the termnumbers then order by lenght of 
  dialcode (matched) and take the first record as the most accurate. (most 
  digits).
  
   
  
  R
  

- Original Message - 


From: Damon Estep 


To: Asterisk Users Mailing List - 
Non-Commercial Discussion 

Sent: Friday, 
January 27, 2006 6:32 PM

Subject: RE: 
[Asterisk-Users] OT?: International number 
parsing

 
Agreed, that is 
what I plan to do, but do you know if the numbering plans are such that a 
countrycode+citycode+”portion of a local number” could ever be mistaken for 
a different country/city combination?
 
Since international 
numbers vary in length, and country and city codes vary in length, there is 
no way to be sure unless the numbering plan is such that no combination of 
citycode plus the start of the local number could ever be mistaken for a 
different city code in the same country.
 
Likewise, there has 
to be assurance that no combination of countrycode+start of city code could 
be mistaken for another country code.
 
 
D
 





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Script HeadSent: Friday, January 27, 2006 3:47 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [Asterisk-Users] OT?: 
International number parsing
 
What you're trying to accomplish can be 
easily done with an SQL query. You need to create a table of all the 
prefixes (international dial+country code+city/carrier) and join by that 
prefix.

On 1/27/06, Damon 
Estep <[EMAIL PROTECTED]> 
wrote:
Can anyone shed some light on "rules" that might 
make the task ofparsing the country code and city codes from a dialed 
number in theCDRs?I know that there is almost never a case where 
a concatenated country and city code could overlap with another country 
code, but what aboutcity codes and local numbers? Is it possible for a 
concatenated citycode and local number to match another city code in the 
same country?I already have the table of country and city codes 
built.Are there holes in this theory;1. Starting after the 
international dialing code, find the longest matchfor country 
code.2. Starting after the country code from step 1, find the longest 
match for city code within that countries table of city codes.3. The 
rest is the local number.Are there known exceptions?Am I 
reinventing the wheel rather than finding the right alreadyexisting 
resource? 
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RE: [Asterisk-Users] Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?

[Dean Collins]

Thanks but this is for a test, I didn’t
buy the first one as it’s a non commercial installation. I’m trying
to test bandwidth etc so I need to try out how 4 of them handle the link
simultaneously, I just don’t know how to add a second test license.

 

 

Dean

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Saturday, 28 January 2006
12:32 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?



 



Well its simple





 





Buy one more license...





 





Dan

 





On 28/01/06, Dean
Collins <[EMAIL PROTECTED]>
wrote: 

Does anyone know how to install more than one of these licenses on the
[EMAIL PROTECTED]

I installed one and works fine but of course when I try to make the second call
it says no lines are available 


Cheers,

Dean




-Original Message-
From: [EMAIL PROTECTED]
[mailto:
[EMAIL PROTECTED]] On Behalf Of Francesco Peeters
(Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: [EMAIL PROTECTED]; Asterisk Users
Mailing List - Non-Commercial Discussion 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?

On Sat, January 21, 2006 23:21, Franz Bräuer said: 
> Hi,
>
> MapsAir wrote:
>> Has anyone successfully Installing the none commercial intel g729
codecs
>> into [EMAIL PROTECTED] 2.2?
>
> Installed them today. Installing from source didn't work for me (Debian, 
> Asterisk 1.2 from svn) but just adding the binaries (see the wiki on
> voip.org) did the job. Have you already
tried the binaries?
>

Kewl! Those work like a treat! 

As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did:

cd /usr/lib/asterisk/modules/
wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so

wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so

After reloading, 'show translation' gives:
Translation times between
formats (in milliseconds) 
 Source Format (Rows)
Destination Format(Columns)

g723  
gsm  ulaw  alaw  g726 adpcm  slin
lpc10  g729 speex  ilbc
  g723
-22 8
817 8
724   115  
19897
   gsm   151
- 7
716 7
623   114   19796
  ulaw  
14616 -
111 2
118   109   19291
  alaw  
14616 1
-11 2
118   109  
19291 
  g726  
154241010
-10
926   117   20099
adpcm   14616
2 211
- 118  
109   19291
  slin  
14515 1
110 1
-17   108   19190

lpc10   161311717261716
-   124   207   106
  g729  
16939252534252441
-   215   114
speex  
16030161625161532  
123 -   105 
  ilbc  
17343292938292845  
136   219 -

Jolly good show, old chap!

--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 
   Cologne HFC-S pins #52, #54, #55 connected in parallel for
synching.
AMD Duron 1GHz - 1GB - * 1.2.1
2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?

[[EMAIL PROTECTED]]

Well its simple
 
Buy one more license...
 
Dan 
On 28/01/06, Dean Collins <[EMAIL PROTECTED]> wrote:
Does anyone know how to install more than one of these licenses on the [EMAIL PROTECTED]I installed one and works fine but of course when I try to make the second call it says no lines are available
Cheers,Dean-Original Message-From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Francesco Peeters (Asterisk)Sent: Saturday, 21 January 2006 5:34 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?On Sat, January 21, 2006 23:21, Franz Bräuer said:
> Hi,>> MapsAir wrote:>> Has anyone successfully Installing the none commercial intel g729 codecs>> into [EMAIL PROTECTED] 2.2?>> Installed them today. Installing from source didn't work for me (Debian,
> Asterisk 1.2 from svn) but just adding the binaries (see the wiki on> voip.org) did the job. Have you already tried the binaries?>Kewl! Those work like a treat!
As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did:cd /usr/lib/asterisk/modules/wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so
wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.soAfter reloading, 'show translation' gives:Translation times between formats (in milliseconds)
 Source Format (Rows) Destination Format(Columns)g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc  g723 -22 8 817 8 724   115   19897
   gsm   151 - 7 716 7 623   114   19796  ulaw   14616 - 111 2 118   109   19291  alaw   14616 1 -11 2 118   109   19291
  g726   154241010 -10 926   117   20099adpcm   14616 2 211 - 118   109   19291  slin   14515 1 110 1 -17   108   19190
lpc10   161311717261716 -   124   207   106  g729   16939252534252441 -   215   114speex   16030161625161532   123 -   105
  ilbc   17343292938292845   136   219 -Jolly good show, old chap!--F PeetersPIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
   Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.AMD Duron 1GHz - 1GB - * 1.2.12 Sweex HFC-PCI cards___--Bandwidth and Colocation provided by 
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[Asterisk-Users] Name/username (sip show peers)

[Ronald Wiplinger]

How can I make it more readable?

Name/username
601/601
123456789/123456789
voipbuster/abcd


601 = hotline
123456789 = Peter Pan

only voipbuster/abcd  is easy read/understandable!


bye

Ronald Wiplinger

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[Asterisk-Users] shared fxo line

[James Harper]

My home asterisk system has failed the wife test (still too much echo
with the current hardware... my voice seems okay but when she talks she
complains of echo), but I'd still like to be able to use it to send and
receive fixed line sms messages to and from my mobile.

My x100p card has 2 ports on it, a 'line' port and a 'phone' port, so
there is provision there to plug the household phone into the phone
port, which is what I've done for testing.

Is this generally held to me a good idea? How will asterisk cope with
it? My dialplan is:

[fxo]
exten => s/0198339100,1,Goto(sms_rx,s,1)
exten => s,1,NoOp()
exten => t,1,NoOp()

[sms_rx]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,SMS(${CALLERIDNUM},a)
exten => s,4,System(/usr/local/bin/sms_rx)
exten => s,5,Hangup

fxo is the default context of my x100p
0198339100 is the number of the sms message center
/usr/local/bin/sms_rx is a wrapper script to mail the sms message to me

This seems to work okay for incoming calls, Asterisk doesn't answer the
phone unless the call originates from the sms message center, which is
what I want. Without the 's' and 't' NoOp() calls, asterisk complains
that it doesn't know what to do with the call.

The only thing I can think of that might cause problems is if Asterisk
tries to send an sms while someone else is using the phone line. Can
asterisk detect that the line is otherwise not in use, eg take phone off
hook, detect dialtone, fail if no dialtone?


Sound good? Anything else that might bite me?

Thanks

James

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[Asterisk-Users] G729 Commercial Licenses.

[[EMAIL PROTECTED]]

Hi all,
 
I have purchased 2 licenses of G729 from digium and has done the registration. It works well and is quite fine with my [EMAIL PROTECTED]. Just want to clarify some licensing issues regarding them.

 
If i had to do a full reformat of my PC and reload [EMAIL PROTECTED] again will i be able to use the licenses again without re-registration?
 
If no. .Is there are limits for this? 
 
Please anyone clarify. 
 
Thanks 
 
Dan
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RE: [Asterisk-Users] OT?: International number parsing

[Damon Estep]

Have you seen situations where a portion
of the local number, when added to the country and city code, result in a
longer match then the actual country/city called and therefore an inaccurate match?

 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron hotmail
Sent: Friday, January 27, 2006
8:17 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT?:
International number parsing



 



It realy is a pain in the *ss.





the problem is just how you explained.  when trying to
match the terminating number, there's no SINGLE fixed pattern for the
dialcodes.  so how do you know how many digits of the term number to match
against the dialcode? you dont.  you have to match the dialcodes against
the termnumbers then order by lenght of dialcode (matched) and take the first
record as the most accurate. (most digits).





 





R







- Original Message - 





From: Damon
Estep 





To: Asterisk Users Mailing List -
Non-Commercial Discussion 





Sent: Friday, January 27,
2006 6:32 PM





Subject: RE:
[Asterisk-Users] OT?: International number parsing





 



Agreed, that is what I plan to do, but do
you know if the numbering plans are such that a
countrycode+citycode+”portion of a local number” could ever be
mistaken for a different country/city combination?

 

Since international numbers vary in
length, and country and city codes vary in length, there is no way to be sure
unless the numbering plan is such that no combination of citycode plus the
start of the local number could ever be mistaken for a different city code in
the same country.

 

Likewise, there has to be assurance that
no combination of countrycode+start of city code could be mistaken for another
country code.

 

 

D

 











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Script Head
Sent: Friday, January 27, 2006
3:47 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT?:
International number parsing



 

What you're trying to
accomplish can be easily done with an SQL query. You need to create a table of
all the prefixes (international dial+country code+city/carrier) and join by
that prefix.






On 1/27/06, Damon Estep <[EMAIL PROTECTED]>
wrote:

Can anyone shed some light on "rules" that might make the
task of
parsing the country code and city codes from a dialed number in the
CDRs?

I know that there is almost never a case where a concatenated country 
and city code could overlap with another country code, but what about
city codes and local numbers? Is it possible for a concatenated city
code and local number to match another city code in the same country?

I already have the table of country and city codes built.

Are there holes in this theory;

1. Starting after the international dialing code, find the longest match
for country code.
2. Starting after the country code from step 1, find the longest match 
for city code within that countries table of city codes.
3. The rest is the local number.

Are there known exceptions?

Am I reinventing the wheel rather than finding the right already
existing resource? 


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[Asterisk-Users] 5,000 concurrent calls system rollout question

[Vic]

Hi,
we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls.
Can this be done with Asterisk? Has it been done before?
I really would like an input on this.
Thanks!

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Re: [Asterisk-Users] DTMF's indescipherable, but voice clean!

[Andrew Kohlsmith]

On Friday 27 January 2006 22:40, OK Computer wrote:
> After many hours today thinking that I had placed a bug into my dialplan, I
> realized that for some reason DTMF tones are simply not making it into
> asterisk! Calling into my pbx transmits crystal-clear audio in both
> directions. But dialing DTMF's from pstn->pbx is unsuccessful, while
> pbx->pstn works fine. The tones simply don't make it through. Tiny brief
> fragments are all.

How are you connected to the PSTN?

-A.
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Re: [Asterisk-Users] AAH out bound routing problem

[ram]

Hi
 
i have added the Dial pattern as you said
 
1NXXNXX9|.NXXNXXNXX
 
but still i get all circuits are busy now call later
 
ram 
On 1/28/06, J.Raborg <[EMAIL PROTECTED]> wrote:
Go to the setup/
Outbound Routing/ pick the one you use for out must be 0 9out  by default and addin the dial patterns  9|. <-including the dotThen try to dial 919197543700JRram wrote: 


Hi rajeev
 
i have posted the extension.conf before
now iam posting extension_additional.conf
 
[EMAIL PROTECTED] asterisk]# more extensions_additional.conf[globals]#include globals_custom.confVM_PREFIX = *RINGTIMER = 15REGTIME = 7:55-17:05REGDAYS = mon-friRECORDEXTEN = ""
PARKNOTIFY = SIP/200OUT_2 = SIP/easycallOUTPREFIX_2 =OUTMAXCHANS_2 = 1
OUTCID_2 = outside account
OPERATOR =NULL = ""IN_OVERRIDE = forcereghoursINCOMING = group-allFAX_RX_EMAIL = [EMAIL PROTECTED]
FAX_RX = systemFAX =DIRECTORY_OPTS =DIRECTORY = last DIAL_OUT = 9DIAL_OPTIONS = trDIALOUTIDS = 2/CALLFILENAME = ""AFTER_INCOMING =
[ext-local]include => ext-local-customexten => 1000,1,Macro(exten-vm,1000,1000)exten => ${VM_PREFIX}1000,1,Macro(vm,1000)exten => 1000,hint,SIP/1000
[outbound-allroutes]include => outbound-allroutes-custominclude => outrt-001-longdistance
[outrt-001-longdistance]include => outrt-001-longdistance-customexten => _1NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)exten => _1NXXNXX,2,Macro(outisbusy)    ; No available circuitsexten => _NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},) 
exten => _NXXNXX,2,Macro(outisbusy) ; No available circuitsexten => _NXX,1,Macro(dialout-trunk,2,${EXTEN},)exten => _NXX,2,Macro(outisbusy)    ; No available circuits
 
 
ram  
On 1/27/06, Rajeev Natarajan <[EMAIL PROTECTED]
> wrote: 
if you are using AAH, please post extensions.conf,extensions_additional.conf - also send us more info on your phones. 
thanksrajeevram wrote:> Hi>> all of them thanks for the quick reply>> i was tried adding 9 as well as 00> but i get number invalid if i put any of the digits 
>> what kind of config files need to post here to resolve the problem>> please assists>> ram>>> On 1/27/06, *Michael Collins* <
 [EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>> Ram,
 On my AAH the stock dial plan requires a 9 first.  For kicks, try > dialing 919197543700 and see what you get. -MC>>>
> >> *From:* 
[EMAIL PROTECTED]> [EMAIL PROTECTED]
 >> [mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED] > ] *On Behalf Of *ram> *Sent:* Friday, January 27, 2006 6:14 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion> *Subject:* [Asterisk-Users] AAH out bound routing problem  Hi all I have installed AAH 
2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
 and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 
> i get message that, all circuits are busy now, please try your call> later and when i see in the console i get this mesage> 
>>> any help Called easycall/19197543700> -- Got SIP response 488 "Not acceptable here" back from (PeerIP)> -- SIP/easycall-838e is circuit-busy 
 ram>>> ___> --Bandwidth and Colocation provided by 
Easynews.com> < http://easynews.com/> -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users>
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Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>
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RE: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?

[Dean Collins]

Does anyone know how to install more than one of these licenses on the [EMAIL 
PROTECTED] 

I installed one and works fine but of course when I try to make the second call 
it says no lines are available


Cheers,

Dean




-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco 
Peeters (Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs 
into [EMAIL PROTECTED] 2.2?

On Sat, January 21, 2006 23:21, Franz Bräuer said:
> Hi,
>
> MapsAir wrote:
>> Has anyone successfully Installing the none commercial intel g729 codecs
>> into [EMAIL PROTECTED] 2.2?
>
> Installed them today. Installing from source didn't work for me (Debian,
> Asterisk 1.2 from svn) but just adding the binaries (see the wiki on
> voip.org) did the job. Have you already tried the binaries?
>

Kewl! Those work like a treat!

As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did:

cd /usr/lib/asterisk/modules/
wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so
wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so

After reloading, 'show translation' gives:
 Translation times between formats (in milliseconds)
  Source Format (Rows) Destination Format(Columns)

 g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
   g723 -22 8 817 8 724   115   19897
gsm   151 - 7 716 7 623   114   19796
   ulaw   14616 - 111 2 118   109   19291
   alaw   14616 1 -11 2 118   109   19291
   g726   154241010 -10 926   117   20099
  adpcm   14616 2 211 - 118   109   19291
   slin   14515 1 110 1 -17   108   19190
  lpc10   161311717261716 -   124   207   106
   g729   16939252534252441 -   215   114
  speex   16030161625161532   123 -   105
   ilbc   17343292938292845   136   219 -

Jolly good show, old chap!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
  AMD Duron 1GHz - 1GB - * 1.2.1
  2 Sweex HFC-PCI cards
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[Asterisk-Users] DTMF's indescipherable, but voice clean!

[OK Computer]

After many hours today thinking that I had placed a bug into my dialplan, I realized that for some reason DTMF tones are simply not making it into asterisk! Calling into my pbx transmits crystal-clear audio in both directions. But dialing DTMF's from pstn->pbx is unsuccessful, while pbx->pstn works fine. The tones simply don't make it through. Tiny brief fragments are all. 
Please listen to the linked audio clip if this doesn't make any sense. I don't know what to do. http://grace.evergreen.edu/~hergab13/msg.wav
thanks,Gabe Herbert
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Re: [Asterisk-Users] OT?: International number parsing

[Ron hotmail]

It realy is a pain in the *ss.
the problem is just how you explained.  when 
trying to match the terminating number, there's no SINGLE fixed pattern for the 
dialcodes.  so how do you know how many digits of the term number to match 
against the dialcode? you dont.  you have to match the dialcodes against 
the termnumbers then order by lenght of dialcode (matched) and take the first 
record as the most accurate. (most digits).
 
R

  - Original Message - 
  From: 
  Damon Estep 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, January 27, 2006 6:32 
  PM
  Subject: RE: [Asterisk-Users] OT?: 
  International number parsing
  
  
  Agreed, that is what 
  I plan to do, but do you know if the numbering plans are such that a 
  countrycode+citycode+”portion of a local number” could ever be mistaken for a 
  different country/city combination?
   
  Since international 
  numbers vary in length, and country and city codes vary in length, there is no 
  way to be sure unless the numbering plan is such that no combination of 
  citycode plus the start of the local number could ever be mistaken for a 
  different city code in the same country.
   
  Likewise, there has 
  to be assurance that no combination of countrycode+start of city code could be 
  mistaken for another country code.
   
   
  D
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Script HeadSent: Friday, January 27, 2006 3:47 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] OT?: 
  International number parsing
   
  What you're trying to accomplish can be 
  easily done with an SQL query. You need to create a table of all the prefixes 
  (international dial+country code+city/carrier) and join by that 
  prefix.
  
  On 1/27/06, Damon 
  Estep <[EMAIL PROTECTED]> 
  wrote:
  Can anyone shed some light on "rules" that might make 
  the task ofparsing the country code and city codes from a dialed number in 
  theCDRs?I know that there is almost never a case where a 
  concatenated country and city code could overlap with another country 
  code, but what aboutcity codes and local numbers? Is it possible for a 
  concatenated citycode and local number to match another city code in the 
  same country?I already have the table of country and city codes 
  built.Are there holes in this theory;1. Starting after the 
  international dialing code, find the longest matchfor country code.2. 
  Starting after the country code from step 1, find the longest match for 
  city code within that countries table of city codes.3. The rest is the 
  local number.Are there known exceptions?Am I reinventing the 
  wheel rather than finding the right alreadyexisting resource? 
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Re: [Asterisk-Users] 802.1p

[Jerry Jones]

On Jan 27, 2006, at 11:33 AM, Mimmus wrote:


Hi,
I'm trying to configure some Quality Of Service among an Asterisk  
server

with RedHat3 and some IP phones on my LAN.
I read about 802.1p (level 2) QoS, using 3 bits of VLAN tag.

Two questions:
- do I need to use tagged links (trunks) end-to-end? In other  
words, do all

ports on all switches from phones to server need to be configured as
'tagged'?


If you wish to use tagging, then yes all ports would have to support  
it for all devices which are using it. Although in your case it  
sounds fairly simple setup, I would suggest just setting your server  
and voice ports to high priority in your switch and leave it at that.  
Not understanding all the ins and outs of tagging may cause more  
problems than you have now. Most switches with management that  
support vlans will also support port based priority.
- how can I configure ethernet card on the Red Hat server  
(Broadcom, tg3
driver) to support tagged traffic and to mark outgoing packets with  
priority

6?

Use port based priority and dont worry about it:)



Thanks in advance for any help
Mimmus

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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Peter Dean]

Alternatively you can issue the following command (which deletes all
the .so modules excluding the g729 codec module) prior to make install

find /usr/lib/asterisk/modules/ -name "*.so" -a ! -name
"codec_g729a.so" -exec rm -f {} \;


On 1/28/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> Hmm - I'd do as others have suggested and move the
> /usr/lib/asterisk/modules  directory to another, and do a make
> clean;make;make install
>
> If you have app_rxfax.so installed then you must have customised your
> original makefile, and not the 1.2.3 makefile, which would suggest that
> these modules are from a previous asterisk version.
>
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Re: [Asterisk-Users] sip qualify=yes interval

[BJ Weschke]

 Yes. If you're looking to change them, you can modify DEFAULT_FREQ_OK
and DEFAULT_FREQ_NOTOK

On 1/27/06, Damon Estep <[EMAIL PROTECTED]> wrote:
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of BJ Weschke
> > Sent: Friday, January 27, 2006 6:18 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] sip qualify=yes interval
> >
> > On 1/27/06, Damon Estep <[EMAIL PROTECTED]> wrote:
> > > In an earlier thread Andrew Kohlsmith enlightened me on the use of
> > > qualify in sip.conf to deal with a peer that is down.
> > >
> > > Since then I have been searching for information on how the behavior
> of
> > > qualify can be tuned.
> > >
> > > The wiki is vague on this;
> > >
> > > " Syntax:
> > >
> > >  qualify=xxx|no|yes
> > >
> > > where XXX is the number of milliseconds used. If yes the default
> timeout
> > > is used, 2 seconds.
> > >
> > > If you turn on qualify in the configuration of a SIP device in
> sip.conf,
> > > Asterisk will send a SIP OPTIONS command regularly to check that the
> > > device is still online. If the device does not answer within the
> > > configured (or default) period (in ms) Asterisk considers the device
> > > off-line for future calls. "
> > >
> > > So;
> > > qualify=1000|yes
> > > means query for SIP OPTIONS, then take then unregister the peer if
> no
> > > response in 1000ms.
> > >
> > > But, how do you set/determine the frequency at which a peer is
> queried?
> > > Does this go on indefinitely after a peer fails to respond to make
> sure
> > > the peer is re-registered when available again? Can the interval be
> set
> > > on a per peer basis?
> > >
> > > Any documentation on this that you can point me to?
> >
> >  It should actually be qualify=1000 if you'd like for the peer to be
> > made unavailable when we don't get a response to SIP OPTIONs within
> > 1000ms (1 second).
>
> Figured that out, thanks.
> >
> >  If the host is reachable, the next SIP OPTION attempt will not come
> > until 60 seconds later. If the host isn't reachable, it will proceed
> > to schedule SIP OPTION attempts every 10 seconds.
> >
> >  These are defined constants in chan_sip.c
>
> Seems silly to make these constant, I can think of many situations where
> you might want to change them (heavily loaded system, many, many peers),
> but I assume the sip options exchange is only a few packets... easy
> enough to change the constants in chan_sip.c I suppose.
> >
> Thanks for the info!
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RE: [Asterisk-Users] sip qualify=yes interval

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of BJ Weschke
> Sent: Friday, January 27, 2006 6:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] sip qualify=yes interval
> 
> On 1/27/06, Damon Estep <[EMAIL PROTECTED]> wrote:
> > In an earlier thread Andrew Kohlsmith enlightened me on the use of
> > qualify in sip.conf to deal with a peer that is down.
> >
> > Since then I have been searching for information on how the behavior
of
> > qualify can be tuned.
> >
> > The wiki is vague on this;
> >
> > " Syntax:
> >
> >  qualify=xxx|no|yes
> >
> > where XXX is the number of milliseconds used. If yes the default
timeout
> > is used, 2 seconds.
> >
> > If you turn on qualify in the configuration of a SIP device in
sip.conf,
> > Asterisk will send a SIP OPTIONS command regularly to check that the
> > device is still online. If the device does not answer within the
> > configured (or default) period (in ms) Asterisk considers the device
> > off-line for future calls. "
> >
> > So;
> > qualify=1000|yes
> > means query for SIP OPTIONS, then take then unregister the peer if
no
> > response in 1000ms.
> >
> > But, how do you set/determine the frequency at which a peer is
queried?
> > Does this go on indefinitely after a peer fails to respond to make
sure
> > the peer is re-registered when available again? Can the interval be
set
> > on a per peer basis?
> >
> > Any documentation on this that you can point me to?
> 
>  It should actually be qualify=1000 if you'd like for the peer to be
> made unavailable when we don't get a response to SIP OPTIONs within
> 1000ms (1 second).

Figured that out, thanks.
> 
>  If the host is reachable, the next SIP OPTION attempt will not come
> until 60 seconds later. If the host isn't reachable, it will proceed
> to schedule SIP OPTION attempts every 10 seconds.
> 
>  These are defined constants in chan_sip.c

Seems silly to make these constant, I can think of many situations where
you might want to change them (heavily loaded system, many, many peers),
but I assume the sip options exchange is only a few packets... easy
enough to change the constants in chan_sip.c I suppose.
> 
Thanks for the info!
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Re: [Asterisk-Users] sip qualify=yes interval

[BJ Weschke]

On 1/27/06, Damon Estep <[EMAIL PROTECTED]> wrote:
> In an earlier thread Andrew Kohlsmith enlightened me on the use of
> qualify in sip.conf to deal with a peer that is down.
>
> Since then I have been searching for information on how the behavior of
> qualify can be tuned.
>
> The wiki is vague on this;
>
> " Syntax:
>
>  qualify=xxx|no|yes
>
> where XXX is the number of milliseconds used. If yes the default timeout
> is used, 2 seconds.
>
> If you turn on qualify in the configuration of a SIP device in sip.conf,
> Asterisk will send a SIP OPTIONS command regularly to check that the
> device is still online. If the device does not answer within the
> configured (or default) period (in ms) Asterisk considers the device
> off-line for future calls. "
>
> So;
> qualify=1000|yes
> means query for SIP OPTIONS, then take then unregister the peer if no
> response in 1000ms.
>
> But, how do you set/determine the frequency at which a peer is queried?
> Does this go on indefinitely after a peer fails to respond to make sure
> the peer is re-registered when available again? Can the interval be set
> on a per peer basis?
>
> Any documentation on this that you can point me to?

 It should actually be qualify=1000 if you'd like for the peer to be
made unavailable when we don't get a response to SIP OPTIONs within
1000ms (1 second).

 If the host is reachable, the next SIP OPTION attempt will not come
until 60 seconds later. If the host isn't reachable, it will proceed
to schedule SIP OPTION attempts every 10 seconds.

 These are defined constants in chan_sip.c

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS

[Andrew McRory]

On 1/26/06, Eric Bishop <[EMAIL PROTECTED]> wrote:


Do you have step by step instructions on how you created these RPMs. I would
like to create a few of my own but compiled for my own custom kernel and
patchea and am not very familiar with RPM packaging


Eric,

Install the srpms, change directory to /usr/src/redhat/SPECS, modify the 
spec as required and run


rpmbuild -ba 

Hope that halps.

Andrew McRory - President/CTO
Linux Systems Engineers, Inc. - http://www.linuxsys.com
Located in beautiful Tallahassee, Florida
Office  850-224-5737
Office  850-575-7213
Mobile  850-294-7567
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RE: [Asterisk-Users] Agent counts

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kevin Smith
> Sent: Friday, January 27, 2006 4:45 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Agent counts
> 
> Hey everyone,
> 
> I am having a little trouble getting this section of the dial plan
> configured. Does anyone know of a way I can get the number of agents
> that are currently logged into a queue? My goal is if no agent is
logged
> in the queue, it gives customers the message we are closed depending
on
> the queue they dial in to. Any suggestions would be great.
> 
> Thanks,
> Kevin


Leavewhenempty=yes

http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf

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Re: [Asterisk-Users] Nagios and Asterisk

[JP Carballo]

Patrick wrote:



I'm in Europe and pagers died here (if they ever lived) when bellbottoms
went out of style but perhaps this link is of help:
http://www.qpage.org/

Regards,
Patrick
 


Lol!
Pagers were being improved upon and were still pretty much in use in 
Asia 4 years ago.
The last one I played with had mp3 capabilities and games and a tiny 
thumbpad.
Their extinction was caused by the widespread adoption of prepaid SIM 
cards for cellphones and SMS.


It's funny though that the last time I saw a bellbottom there was about 
the same time.


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] app_rxfax.so and app_txfax.so

[Doug Lytle]

Support Internet.net wrote:
 
In verbose mode, it say:

 -- Executing Answer("SIP/18193408704-b95b", "") in new stack
 -- Executing RxFAX("SIP/18193408704-b95b", "/temp/test.tif") in new stack
 
It heard the fax tone and don't take the call.

Do I something of wrong?


I'm seeing the same thing, but figured it was because I'm running the 
SVN version.  I'm planning on installing stable this weekend to see if 
it takes care of it.


Doug

--
Ben Franklin quote:

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


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[Asterisk-Users] Agent counts

[Kevin Smith]

Hey everyone,

I am having a little trouble getting this section of the dial plan 
configured. Does anyone know of a way I can get the number of agents 
that are currently logged into a queue? My goal is if no agent is logged 
in the queue, it gives customers the message we are closed depending on 
the queue they dial in to. Any suggestions would be great.


Thanks,
Kevin
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[Asterisk-Users] sip qualify=yes interval

[Damon Estep]

In an earlier thread Andrew Kohlsmith enlightened me on the use of
qualify in sip.conf to deal with a peer that is down.

Since then I have been searching for information on how the behavior of
qualify can be tuned.

The wiki is vague on this;

" Syntax: 

  qualify=xxx|no|yes 

where XXX is the number of milliseconds used. If yes the default timeout
is used, 2 seconds. 

If you turn on qualify in the configuration of a SIP device in sip.conf,
Asterisk will send a SIP OPTIONS command regularly to check that the
device is still online. If the device does not answer within the
configured (or default) period (in ms) Asterisk considers the device
off-line for future calls. "

So;
qualify=1000|yes
means query for SIP OPTIONS, then take then unregister the peer if no
response in 1000ms.

But, how do you set/determine the frequency at which a peer is queried?
Does this go on indefinitely after a peer fails to respond to make sure
the peer is re-registered when available again? Can the interval be set
on a per peer basis?

Any documentation on this that you can point me to?
Thx!


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Re: [Asterisk-Users] Calls pickup

[Mojo with Horan & Company, LLC]

I pickup ringing PSTN lines from both SIP and IAX2 phones :)

Mimmus wrote:

Hi,
is it possible pickup calls (with *8) between different channels (SIP and
IAX)?


Thanks
Mimmus

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] OT?: International number parsing

[Damon Estep]

Agreed, that is what I plan to do, but do
you know if the numbering plans are such that a countrycode+citycode+”portion
of a local number” could ever be mistaken for a different country/city
combination?

 

Since international numbers vary in
length, and country and city codes vary in length, there is no way to be sure
unless the numbering plan is such that no combination of citycode plus the
start of the local number could ever be mistaken for a different city code in
the same country.

 

Likewise, there has to be assurance that
no combination of countrycode+start of city code could be mistaken for another
country code.

 

 

D

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Script Head
Sent: Friday, January 27, 2006
3:47 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT?:
International number parsing



 

What you're trying to
accomplish can be easily done with an SQL query. You need to create a table of
all the prefixes (international dial+country code+city/carrier) and join by
that prefix.







On 1/27/06, Damon Estep <[EMAIL PROTECTED]>
wrote:

Can anyone shed some light on "rules" that might make the
task of
parsing the country code and city codes from a dialed number in the
CDRs?

I know that there is almost never a case where a concatenated country 
and city code could overlap with another country code, but what about
city codes and local numbers? Is it possible for a concatenated city
code and local number to match another city code in the same country?

I already have the table of country and city codes built.

Are there holes in this theory;

1. Starting after the international dialing code, find the longest match
for country code.
2. Starting after the country code from step 1, find the longest match 
for city code within that countries table of city codes.
3. The rest is the local number.

Are there known exceptions?

Am I reinventing the wheel rather than finding the right already
existing resource? 


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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Julian Lyndon-Smith]

Hmm - I'd do as others have suggested and move the 
/usr/lib/asterisk/modules  directory to another, and do a make 
clean;make;make install


If you have app_rxfax.so installed then you must have customised your 
original makefile, and not the 1.2.3 makefile, which would suggest that 
these modules are from a previous asterisk version.


Let us know how you get on.

Julian.

Dan Littlejohn wrote:

On 1/27/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:

These modules are not part of the standard 1.2.3 release - did you also
install the 1.2.3 release of the asterisk-addons package ?

If * is loading older modules (which it probably is because of your
config files) then it may cause grief ;)

My .2p worth. Probably not helpful, but maybe, just maybe 

Julian

Dan Littlejohn wrote:

On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote:

Hi Brent -


Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.

Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server would reply to a
ping, but no ssh login, no local console login - just locked up. This ain't
good for business.

We've been doing fine with 1.2.3 so far.  No problems reported, though I
only have it deployed in a small office.  Definitely no lock-ups.

On the asterisk side, just a basic question - did you make sure to remove
the old modules so the new 1.2.3 versions got installed?

As far as the lockups, maybe it is coincidental?  I've never had asterisk
(even the crazy CVS versions) lock a whole OS like that.  I have had
machines running asterisk lock up, but it always turned out to be caused by
something else like bad hardware, or unrelated network problems.

- Noah

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I was confused about the modules.

Got this warning when upgrading to 1.2.3 even when using the most
current asterisk-addons and even svn asterisk-addons.

 WARNING WARNING WARNING

 Your Asterisk modules directory, located at
 /usr/lib/asterisk/modules
 contains modules that were not installed by this
 version of Asterisk. Please ensure that these
 modules are compatible with this version before
 attempting to run Asterisk.

   app_addon_sql_mysql.so
   app_rxfax.so
   app_saycountpl.so
   app_striplsd.so
   app_substring.so
   app_txfax.so
   cdr_addon_mysql.so
   chan_modem_aopen.so
   chan_modem_bestdata.so
   chan_modem_i4l.so
   chan_modem.so
   format_mp3.so
   res_config_mysql.so

 WARNING WARNING WARNING

Do not understand how to fix this?  Do not know if that would also be
related to the ops crashing.

Dan
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There is no asterisk-addons 1.2.3.  Only 1.2.1 and I tried that and
svn and still get this warning?
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Re: [Asterisk-Users] fxo/fxs cards with 8 ports

[Time Bandit]

On 1/27/06, roswel ajf <[EMAIL PROTECTED]> wrote:
> we have got asterisk 1.0 (over 1 yrs old) version and very old zaptel
> version. That code is working only with 8 or less ports (accumulative) on
> digium fxs/fxo cards (2 cards with 4 ports each).
A lot of improvements/bug-fixes as gone in Asterisk and Zaptel in a
year, so I would at least update to 1.0.10, the last version of the
1.0 branch.

But, if you don't have any problem as-is, why risk upgrading

> the questoin is, what if we want 12 ports?..well, really, i don't understand
> the limitations? is it simply zaptel driver code fix? or kernel fix? or
In short, a TDM400P generates 1000 Interrupt Request to your CPU each
second. Multiply that by the number of card you put in the server and
I think you can see where the problem lies...

As a side note, some people have reported running 3 or 4 cards in a
server without problems. YMMV

Personnaly, I would upgrade to a TDM2400P, this card can have up to 24
ports and will generate only 1000 Interrupt Request per second. You
can sell your old TDM400 or recycle them in another *

I just installed 2 TDM2400P with hardware echo-canceller and I'm
really satisfied with them up to now.

> technology limitation? donno any tips would help. we are though planning to
> move to latest asterisk 1.2.3 on linux 2.4.
>
> thanks, very much appreciate any comments.

You're welcome

hth
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RE: [Asterisk-Users] CDR reporting between two Asterisk servers

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of JP Carballo
> Sent: Friday, January 27, 2006 3:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] CDR reporting between two Asterisk
servers
> 
> [EMAIL PROTECTED] wrote:
> 
> > Damon Estep wrote:
> >
> >> Use cdr_mysql
> >>
> >> Log your CDRs to a common database
> >> Query as needed from either server using realtime() or from an
external
> >> app
> >>
> >
> > Yeah, I thought about that.  If it works how I think it would
> > work though I would have two CDR records for one call though.
> > I would have one record from the remote server and one from
> > the local. Correlating one record with another could be a pain.
> 
> There are several fields you can use to sync records. You could also
set
> the accountcode and/or userfield to a value you generate per call..
> 
How would you sync the generated code between two servers?
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RE: [Asterisk-Users] Nagios and Asterisk

[Patrick]

On Sat, 2006-01-28 at 09:20 +1100, James Harper wrote:
[snip]
> Along the same lines, does anyone know of any snpp servers that are
> compatible with app_sms? I have nagios on another server and would like
> to send pages via app_sms and so an snpp server running on the asterisk
> server would be a good way to go about it.

I'm in Europe and pagers died here (if they ever lived) when bellbottoms
went out of style but perhaps this link is of help:
http://www.qpage.org/

Regards,
Patrick

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RE: [Asterisk-Users] Polycom 501 horrible echo

[gw]

Hello Chad,
Where did you get 1.6.4.0064?  Site says latest is 1.63.0067.  Also my
supplier only has 1.63.0067.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad
Osmond
Sent: Friday, January 27, 2006 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom 501 horrible echo

I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.

Chad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Herring
Sent: January 26, 2006 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom 501 horrible echo

Now I'm really confused...
1.6.3 is on the Polycom Website as the latest...

I'm running 1.6.2.0041 according to my phone.

Which firmware "worked" for you?

At 04:04 PM 1/26/2006, Ron Senykoff wrote:
> > We also have noticed a poor server config can cause this in testing.
> >
> > Noticed when I had one person building * servers using Debian. Had 
> > them rebuilt with FC4 and have no issues - yet:)
>
>I recently upgraded all our phones to the latest Polycom firmware
>1.6.2 and went from great speakerphone to tons of feedback. I would 
>hate to have to go back to the old firmware. Although Polycom 
>recommends keeping the older bootrom unless you need https 
>provisioning, I'm going to try the new bootrom and see if it fixes the 
>problem.
>
>This is being experienced across 3 corporate offices with 3 separate 
>Asterisk servers. And I have to reiterate... all was good until the 
>firmware upgrade.
>
>-Ron
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Re: [Asterisk-Users] logging performance, important impact?

[Zoa]

If you really need it, save it on a remote server (nfs or so), that 
should minimize the problems


Zoa
---
www.asteriskguru.com


Simone Cittadini wrote:


Moises Silva ha scritto:

How important is the impact i could have if I have a single entry log 
file in /etc/asterisk/logger.conf wich loggs everything, even debug 
level. This is sometimes important to us because it helps us to make 
a track of the issues some times we have with the system. I just want 
to know if there is a considerable impact in performance because of 
the writing of the logs.




I haven't made benchmarks, but speaking out of my experience and 
knowing that asterisk debug level is very verbose I think it will have 
a sensible impact.
I can remember a very slow samba installation due to the sysadmin 
forgetting to turn off the debug level of logging, it made the 
difference between "we can use it" and "we switch back to windows", 
and I'm talking about a dozen of users, not big numbers.
Are you sure debug level will help you tracking the issues ? Usually 
debug level info is for debug like "what is the bottleneck ?", "why my 
prepaid agi isn't doing the update on hangup ?", nothing you need to 
keep tracking once you are in production.



Is better to log as few expected stuff as possible and as much 
unexpected stuff as possible.



Anyway autoanswering your question is pretty simple, put an agi which 
timestamps the first line of each extension and one for the last one, 
send a lot of calls in the system with and without debugging and look 
at the results.

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RE: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

[gw]

Strange though it's only effecting since build 8000...

Here's the snippet:

exten => s,1,LookupCIDName
exten =>
s,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/incoming/${IncomingLine
}/In-${STRFTIME(${EPOCH},,%Y%m% . . .
exten => s,3,Monitor(wav,${CALLFILENAME})
exten => s,4,GotoIf($["${INCOMINGLINE}" = "9146930821"]?9:5);
exten => s,5,GotoIfTime(20:01-7:59|mon-sun|*|*?9)
exten => s,6,Dial(${ADCOMDAYRINGTO},25,t);All
exten => s,7,NoOp(${DIALSTATUS})
exten => s,8,Goto(adcomincoming,s,11)
exten => s,9,Dial(${ADCOMNIGHTRINGTO},25,t);Cisco,Ping,Poly,SPA841
exten => s,10,NoOp(${DIALSTATUS})  
exten => s,107,Answer
exten => s,108,Wait(1)
exten => s,109,BackGround(adcom1/thankyou); Thank you for calling ADCOM
Corp.
exten => s,110,Playback(busy-pls-hold)
exten => s,111,Queue(adcomgwqueue)
exten => s,11,Answer ; Answer the line
exten => s,12,Wait(1)

... Menu plays.

ADCOMDAYRINGTO =
${C79601L1}&${OFFICE3}&${POLY1L1}&SIP/344&SIP/345&SIP/364&${SOMERSADCOM}
; SIP/355&SIP/342 SP
ADCOMNIGHTRINGTO =
${C79601L1}&${POLY1L1}&SIP/344&SIP/345&SIP/364&${SOMERSADCOM} 

So could it have something to do with the dialstring?  I would think
asterisk would say something first before doing the dials.

I'll try it later with a simple dialstring.  I'm going to rebuild it
anyhow.

I am looking to use a global variable in like a switch setup, to direct
calls to particular setups based on a menu.  For example, someone dials
ext 333, and they get a menu for day mode, night mode, holiday, away
from office, etc and the dialplan will ring different devices depending
on the choice...

We'll see what happens...

On a side note, I believe it works if I dial right into the menu
playback.  But if it's the dialstring that's wrong, I would think
asterisk should complain about it.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: Friday, January 27, 2006 4:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

This is not a problem of the ISDN line (or chan_capi), Asterisk is just
not doing anything after

  -- Executing
GotoIfTime("CAPI/ISDNL1/5912211-0","20:01-7:59|mon-sun|*|*?9") in new
stack

and without further commands (like Ringing(), Answer(), ...) the ISDN
line timed out and disconnects.

So either your dialplan is buggy, or Asterisk is not doing what you
want.
What should be done according your extensions.conf in that state ?

Armin

On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote:
>  /etc/init.d/asterisk stop
> Stopping Asterisk PBX: .
> censys:/usr/src/asterisk-8632#  cd ..
> censys:/usr/src# asterisk -vc
> 
>   == Parsing '/etc/asterisk/asterisk.conf': Found
> 
>   == Parsing '/etc/asterisk/extconfig.conf': Found
> 
> Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and 
> others.
> 
> Created by Mark Spencer <[EMAIL PROTECTED]>
> 
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for 
> details.
> 
> This is free software, with components licensed under the GNU General 
> Public
> 
> License version 2 and other licenses; you are welcome to redistribute 
> it under
> 
> certain conditions. Type 'show license' for details.
> 
> ==
> ==
> =
> 
>   == Parsing '/etc/asterisk/logger.conf': Found
> 
> Asterisk Event Logger Started /var/log/asterisk/event_log
> 
> Asterisk Dynamic Loader loading preload modules:
> 
> CLIP
>  [chan_capi.so] => (Common ISDN API for Asterisk)
> 
>   == Parsing '/etc/asterisk/capi.conf': Found
> 
>   == This box has 1 capi controller(s).
> 
> -- CAPI/contr1 supports DTMF
> 
> -- CAPI/contr1 supports echo cancellation
> 
> -- CAPI/contr1 supports line interconnect
> 
> -- CAPI/contr1 supports supplementary services
> 
>> supplementary services : 0x010f
> 
>> HOLD/RETRIEVE
> 
>> TERMINAL PORTABILITY
> 
>> ECT
> 
>> 3PTY
> 
>> MWI
> 
>   == Reading config for ISDNL1
> 
> -- capi_pvt ISDNL1-pseudo-D (5912211,capi-in-5912211,0,2) (1,4,64)
> 
> -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)
> 
> -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)
> 
>   == Reading config for ISDNL2
> 
> -- capi_pvt ISDNL2-pseudo-D (6930821,capi-in-6930821,0,2) (0,0,64)
> 
> -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)
> 
> -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)
> 
> -- listening on contr1 CIPmask = 0x1fff03ff
> 
>   == Registered channel type 'CAPI' (Common ISDN API Driver (cm-0.6.3)

> )
> 
>   == Registered application 'capiCommand'
> 
>   == Registered custom function VANITYNUMBER
> 
> CLIP
> 
> Asterisk Ready.
> *CLI> capi debug CAPI Debugging Enabled
> *CLI> -- Saved useragent
> "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 364
> 
> -- Executing Set("SIP/366-11b2",

Re: [Asterisk-Users] OT?: International number parsing

[Script Head]

What you're trying to accomplish can be easily done with an SQL query. You need to create a table of all the prefixes (international dial+country code+city/carrier) and join by that prefix.
On 1/27/06, Damon Estep <[EMAIL PROTECTED]> wrote:
Can anyone shed some light on "rules" that might make the task ofparsing the country code and city codes from a dialed number in theCDRs?I know that there is almost never a case where a concatenated country
and city code could overlap with another country code, but what aboutcity codes and local numbers? Is it possible for a concatenated citycode and local number to match another city code in the same country?
I already have the table of country and city codes built.Are there holes in this theory;1. Starting after the international dialing code, find the longest matchfor country code.2. Starting after the country code from step 1, find the longest match
for city code within that countries table of city codes.3. The rest is the local number.Are there known exceptions?Am I reinventing the wheel rather than finding the right alreadyexisting resource?
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Re: [Asterisk-Users] CDR reporting between two Asterisk servers

[JP Carballo]

[EMAIL PROTECTED] wrote:


Damon Estep wrote:


Use cdr_mysql

Log your CDRs to a common database
Query as needed from either server using realtime() or from an external
app



Yeah, I thought about that.  If it works how I think it would
work though I would have two CDR records for one call though.
I would have one record from the remote server and one from
the local. Correlating one record with another could be a pain.


There are several fields you can use to sync records. You could also set 
the accountcode and/or userfield to a value you generate per call..


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] CDR reporting between two Asterisk servers

[Script Head]

You have to write a CDR normalization script that would sift thru the calls and remove duplicate entries. It's also quite easy to do with a stored procedure, this way every time CDR gets written, it eliminates a duplicate.
On 1/27/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Damon Estep wrote:> Use cdr_mysql>> Log your CDRs to a common database
> Query as needed from either server using realtime() or from an external> app>Yeah, I thought about that.  If it works how I think it wouldwork though I would have two CDR records for one call though.
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[Asterisk-Users] Context for SIP incoming (newbie question?)

[Alejandro Mejía Evertsz]

Please help me out with this…

 

To which context of the dial-plan does asterisk tries to
match incoming calls when acting as a sip client?

To be more specific:

In extensions.conf… Under which context should I place
 “exten => 648064,1,Dial(TECH/peer)”
for an entry like this “register => 648064:[EMAIL PROTECTED]/648064” ?

 

This is because I want to match one sip client to one
context, and another sip client into another context.

Is it possible?

What is the correct way to do it??

 

Thanks,

Alejandro






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Re: [Asterisk-Users] VOXEE Caller ID..

[Ben Higley]

I use voicepulse as well... but just curious if maybe they dont allow it?
or what.



> I'm running Asterisk 1.2.1.  You're supposed to have to set callerid this
> way:
>
> Set(CALLERID(num)=9315551212)
>
> In fact, doing this with voicepulse works fine.  However it doesn't
> with voxee (at least for me).  I have to set callerid the old
> fashioned way:
>
> SetCallerID(9315551212)
>
> I even tried setting it using both methods, the correct method
> followed by the old method, and it still wouldn't work (at least for
> me).  The old way still works for voicepulse too, so I just left it
> set that way.
>
> Joseph Tanner
>
> On 1/27/06, Ben Higley <[EMAIL PROTECTED]> wrote:
>> I cannot find any means of passing my own Callerid using Voxee. It
>> always
>> comes across as NO ID, or nothing, or unknown.
>>
>> I could not find anything on their website about setting your own caller
>> id in  the system either. (their web account pages).
>>
>> Is anyone here using their own Callerid information through Voxee?
>>
>> thanks
>>
>>
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 18, Issue 181

[Noah Miller]

Hi Dan - 

> Got this warning when upgrading to 1.2.3 even when using the most
> current asterisk-addons and even svn asterisk-addons.
> 
>  WARNING WARNING WARNING
> 
>  Your Asterisk modules directory, located at
>  /usr/lib/asterisk/modules
>  contains modules that were not installed by this
>  version of Asterisk. Please ensure that these
>  modules are compatible with this version before
>  attempting to run Asterisk.
> 
>app_addon_sql_mysql.so
>app_rxfax.so
>app_saycountpl.so
>app_striplsd.so
>app_substring.so
>app_txfax.so
>cdr_addon_mysql.so
>chan_modem_aopen.so
>chan_modem_bestdata.so
>chan_modem_i4l.so
>chan_modem.so
>format_mp3.so
>res_config_mysql.so
> 
>  WARNING WARNING WARNING
> 
> Do not understand how to fix this?  Do not know if that would also be
> related to the ops crashing.

Yeah, as Julian said, the asterisk installer does a check to see if there
are any modules in /usr/lib/asterisk/modules.  If there are, it will return
that warning.  This includes if you install the addons prior to installing
asterisk itself.  I've always made sure that the modules directory was
cleared (moved all the old modules to another folder in case I need to
revert to the old version), then installed asterisk, then any addons.

If you're using the latest versions of everything, though, you should be
fine.  The error message is there to make sure you don't try to use
older/incompatible versions of modules with a newer version of asterisk.

- Noah

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RE: [Asterisk-Users] app_rxfax.so and app_txfax.so

[Support Internet.net]

You where right! now i'm running Asterisk with 
SpanDSP modules.
 
But I have another problem:
 
When I use these applications, it's doing nothing, 
I explain me:
 
My extensions.conf:
 
exten => s,1,Answer()
exten => s,2,RxFAX(/temp/test.tif)
 
 
In verbose mode, it say: 
 -- Executing Answer("SIP/18193408704-b95b", 
"") in new stack
 -- Executing RxFAX("SIP/18193408704-b95b", 
"/temp/test.tif") in new stack
 
It heard the fax tone and don't take the 
call.
Do I something of wrong?
 
Loic Foucault

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RE: [Asterisk-Users] Nagios and Asterisk

[James Harper]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Darrell Long
> Sent: Saturday, 28 January 2006 05:37
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Nagios and Asterisk
> 
> Is anyone using Asterisk (and Festival) to make calls to appropriate
> persons (techs, etc. ) when Nagios generates a particular type of
alert?
> 
> If so, I would love to hear how people are doing it.

I'm not doing that but dropping a call file in should do the trick
shouldn't it?

Along the same lines, does anyone know of any snpp servers that are
compatible with app_sms? I have nagios on another server and would like
to send pages via app_sms and so an snpp server running on the asterisk
server would be a good way to go about it.

Thanks

James
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Re: [Asterisk-Users] How's the best way to set up agents...

[pdhales]

Base your internal dialplan around Dial(Agent/xxx) rather than DIAL(SIP/xxx) 
for calling extensions.

later,

PaulH

> Ben Ferguson <[EMAIL PROTECTED]> wrote:
> 
> So I'm trying to set up queues and agents and am trying to figure out 
> the
> best way to set up what I need to do.  And what I need to do is 
> basically
> get Asterisk to mimic my company's current phone system.  As close as
> possible of course.  And my main problem is queues and agents.  
> Currently,
> for our queues and agents, a person is assigned a hot-desk extension, 
> which
> they use to login to any phone and then they can send and receive calls 
> at
> that extension.  There is no seperate extension and agent id--they are
> pretty much the same thing.  But the extension moves around with them to
> wherever they log in.  The advantage is that they always have the same
> extension.  When no one is logged into a phone, the phone is assigned a
> catch all username called "no user" which has limited dialing 
> capabilities.
> With Asterisk, when you log in an agent, they assume the extension of 
> the
> phone that they have just logged in under.  Yes, if they are a member of 
> a
> queue, they will always receive calls from that queue regardless of what
> extension they are at, but for DID and internal calls, you would never 
> know
> which extension to dial to reach a person setup in such a way.
>  
> So here's what I've come up with (but I, of course, still have
> questions...):  Match the agent ID to an extension.  Assign an agent 
> their
> ID and then assign a certain working area, and a assign certain phone to
> that working area and assign that phone an extension that is the same as
> their agent id.  The pitfall here is that if you do it this way, only 
> one
> person could utilize that working area and that phone.  Agents working 
> in
> shifts at the same work area and same phone would not work--unless 
> multiple
> agents used the same extension, which again kills your DID and internal
> calling.  Of course, you could assign the extension the phone based on 
> time,
> but what if you want to have open seating and you want any agent to sit 
> in
> any work area at any given time?  You can see it starts to get messy 
> when
> you try to work it out this way.  Also, if you did do the matched 
> extension
> and agent id, if a person was re-assigned to a new work area, they 
> either
> have to take their phone with them, or you get to setup a new phone with
> their extension.  Perhaps there is something in Asterisk that I don't 
> know
> about that could benefit me here?
>  
> I'm thinking another way to do something like what I need is via XML, 
> but
> I'm not exactly sure how to do it this way.  Can you assign a phone a
> certain extension and then give it an option for "logging in" using 
> their
> agent id and then based on their agent id, push a new XML file out that
> assigns their specific extension.  Can you re-assign a new extension to 
> a
> phone this way?  I believe this would be a decent way to set this up (if 
> the
> XML files aren't too complicated) but I'm not exactly sure how to do it. 
>  
>  
> Any suggestions, pointers, directions...?
>  
> Thanks!
> Ben Ferguson
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[Asterisk-Users] fxo/fxs cards with 8 ports

[roswel ajf]

we have got asterisk 1.0 (over 1 yrs old) version and very old zaptel 
version. That code is working only with 8 or less ports (accumulative) on 
digium fxs/fxo cards (2 cards with 4 ports each).


the questoin is, what if we want 12 ports?..well, really, i don't understand 
the limitations? is it simply zaptel driver code fix? or kernel fix? or 
technology limitation? donno any tips would help. we are though planning to 
move to latest asterisk 1.2.3 on linux 2.4.


thanks, very much appreciate any comments.


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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Dan Littlejohn]

On 1/27/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> These modules are not part of the standard 1.2.3 release - did you also
> install the 1.2.3 release of the asterisk-addons package ?
>
> If * is loading older modules (which it probably is because of your
> config files) then it may cause grief ;)
>
> My .2p worth. Probably not helpful, but maybe, just maybe 
>
> Julian
>
> Dan Littlejohn wrote:
> > On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote:
> >> Hi Brent -
> >>
> >>> Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
> >>> the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
> >>> hours or so.
> >>>
> >>> Since upgrading to 1.2.3, though, the whole system has locked up twice. 
> >>> Once
> >>> on Thursday, and then about a half hour ago. The server would reply to a
> >>> ping, but no ssh login, no local console login - just locked up. This 
> >>> ain't
> >>> good for business.
> >>
> >> We've been doing fine with 1.2.3 so far.  No problems reported, though I
> >> only have it deployed in a small office.  Definitely no lock-ups.
> >>
> >> On the asterisk side, just a basic question - did you make sure to remove
> >> the old modules so the new 1.2.3 versions got installed?
> >>
> >> As far as the lockups, maybe it is coincidental?  I've never had asterisk
> >> (even the crazy CVS versions) lock a whole OS like that.  I have had
> >> machines running asterisk lock up, but it always turned out to be caused by
> >> something else like bad hardware, or unrelated network problems.
> >>
> >> - Noah
> >>
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> >> Asterisk-Users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > I was confused about the modules.
> >
> > Got this warning when upgrading to 1.2.3 even when using the most
> > current asterisk-addons and even svn asterisk-addons.
> >
> >  WARNING WARNING WARNING
> >
> >  Your Asterisk modules directory, located at
> >  /usr/lib/asterisk/modules
> >  contains modules that were not installed by this
> >  version of Asterisk. Please ensure that these
> >  modules are compatible with this version before
> >  attempting to run Asterisk.
> >
> >app_addon_sql_mysql.so
> >app_rxfax.so
> >app_saycountpl.so
> >app_striplsd.so
> >app_substring.so
> >app_txfax.so
> >cdr_addon_mysql.so
> >chan_modem_aopen.so
> >chan_modem_bestdata.so
> >chan_modem_i4l.so
> >chan_modem.so
> >format_mp3.so
> >res_config_mysql.so
> >
> >  WARNING WARNING WARNING
> >
> > Do not understand how to fix this?  Do not know if that would also be
> > related to the ops crashing.
> >
> > Dan
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
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There is no asterisk-addons 1.2.3.  Only 1.2.1 and I tried that and
svn and still get this warning?
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RE: [Asterisk-Users] CDR reporting between two Asterisk servers

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Friday, January 27, 2006 2:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] CDR reporting between two Asterisk
servers
> 
> Damon Estep wrote:
> > Use cdr_mysql
> >
> > Log your CDRs to a common database
> > Query as needed from either server using realtime() or from an
external
> > app
> >
> 
> Yeah, I thought about that.  If it works how I think it would
> work though I would have two CDR records for one call though.
> I would have one record from the remote server and one from
> the local. Correlating one record with another could be a pain.
> 
> 
True, but the CDR just logs what happens, and in your scenario you
actually do have 2 calls...
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Re: [Asterisk-Users] VOXEE Caller ID..

[Joseph Tanner]

I'm running Asterisk 1.2.1.  You're supposed to have to set callerid this way:

Set(CALLERID(num)=9315551212)

In fact, doing this with voicepulse works fine.  However it doesn't
with voxee (at least for me).  I have to set callerid the old
fashioned way:

SetCallerID(9315551212)

I even tried setting it using both methods, the correct method
followed by the old method, and it still wouldn't work (at least for
me).  The old way still works for voicepulse too, so I just left it
set that way.

Joseph Tanner

On 1/27/06, Ben Higley <[EMAIL PROTECTED]> wrote:
> I cannot find any means of passing my own Callerid using Voxee. It always
> comes across as NO ID, or nothing, or unknown.
>
> I could not find anything on their website about setting your own caller
> id in  the system either. (their web account pages).
>
> Is anyone here using their own Callerid information through Voxee?
>
> thanks
>
>
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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Joseph Tanner]

Got my commands mixed up.  The last should be "cp -r
/usr/lib/asterisk/modules.bak /usr/lib/asterisk/modules", it shouldn't
be /usr/lib/modules.  Sorry bout that, wasn't thinking clearly.

Joseph Tanner

On 1/27/06, Joseph Tanner <[EMAIL PROTECTED]> wrote:
> Quick and dirty solution:
>
> mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.bak
>
> Then go into the asterisk source directory (in my case,
> /usr/src/asterisk) and do a make install.  Might as well re-install
> the asterisk-addons too, if you need anything there.  Try running
> asterisk now and put it through its paces.  If you're missing any
> functionality, try to put it back in (probably a module included in
> asterisk-addons).  If you can't get it working and time is critical,
> just stop asterisk, do a "mv /usr/lib/asterisk/modules
> /usr/lib/asterisk/modules.new" and then a "cp -r
> /usr/lib/asterisk/modules.bak /usr/lib/modules" and restart asterisk
> and try to figure out what went wrong.  The modules.new directory has
> all the new modules, modules.bak still has the old ones.
>
> Joseph Tanner
>
> On 1/27/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote:
> > On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote:
> > > Hi Brent -
> > >
> > > > Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course 
> > > > had
> > > > the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 
> > > > 24
> > > > hours or so.
> > > >
> > > > Since upgrading to 1.2.3, though, the whole system has locked up twice. 
> > > > Once
> > > > on Thursday, and then about a half hour ago. The server would reply to a
> > > > ping, but no ssh login, no local console login - just locked up. This 
> > > > ain't
> > > > good for business.
> > >
> > >
> > > We've been doing fine with 1.2.3 so far.  No problems reported, though I
> > > only have it deployed in a small office.  Definitely no lock-ups.
> > >
> > > On the asterisk side, just a basic question - did you make sure to remove
> > > the old modules so the new 1.2.3 versions got installed?
> > >
> > > As far as the lockups, maybe it is coincidental?  I've never had asterisk
> > > (even the crazy CVS versions) lock a whole OS like that.  I have had
> > > machines running asterisk lock up, but it always turned out to be caused 
> > > by
> > > something else like bad hardware, or unrelated network problems.
> > >
> > > - Noah
> > >
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > I was confused about the modules.
> >
> > Got this warning when upgrading to 1.2.3 even when using the most
> > current asterisk-addons and even svn asterisk-addons.
> >
> >  WARNING WARNING WARNING
> >
> >  Your Asterisk modules directory, located at
> >  /usr/lib/asterisk/modules
> >  contains modules that were not installed by this
> >  version of Asterisk. Please ensure that these
> >  modules are compatible with this version before
> >  attempting to run Asterisk.
> >
> >app_addon_sql_mysql.so
> >app_rxfax.so
> >app_saycountpl.so
> >app_striplsd.so
> >app_substring.so
> >app_txfax.so
> >cdr_addon_mysql.so
> >chan_modem_aopen.so
> >chan_modem_bestdata.so
> >chan_modem_i4l.so
> >chan_modem.so
> >format_mp3.so
> >res_config_mysql.so
> >
> >  WARNING WARNING WARNING
> >
> > Do not understand how to fix this?  Do not know if that would also be
> > related to the ops crashing.
> >
> > Dan
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Joseph Tanner]

Quick and dirty solution:

mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.bak

Then go into the asterisk source directory (in my case,
/usr/src/asterisk) and do a make install.  Might as well re-install
the asterisk-addons too, if you need anything there.  Try running
asterisk now and put it through its paces.  If you're missing any
functionality, try to put it back in (probably a module included in
asterisk-addons).  If you can't get it working and time is critical,
just stop asterisk, do a "mv /usr/lib/asterisk/modules
/usr/lib/asterisk/modules.new" and then a "cp -r
/usr/lib/asterisk/modules.bak /usr/lib/modules" and restart asterisk
and try to figure out what went wrong.  The modules.new directory has
all the new modules, modules.bak still has the old ones.

Joseph Tanner

On 1/27/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote:
> On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote:
> > Hi Brent -
> >
> > > Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
> > > the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
> > > hours or so.
> > >
> > > Since upgrading to 1.2.3, though, the whole system has locked up twice. 
> > > Once
> > > on Thursday, and then about a half hour ago. The server would reply to a
> > > ping, but no ssh login, no local console login - just locked up. This 
> > > ain't
> > > good for business.
> >
> >
> > We've been doing fine with 1.2.3 so far.  No problems reported, though I
> > only have it deployed in a small office.  Definitely no lock-ups.
> >
> > On the asterisk side, just a basic question - did you make sure to remove
> > the old modules so the new 1.2.3 versions got installed?
> >
> > As far as the lockups, maybe it is coincidental?  I've never had asterisk
> > (even the crazy CVS versions) lock a whole OS like that.  I have had
> > machines running asterisk lock up, but it always turned out to be caused by
> > something else like bad hardware, or unrelated network problems.
> >
> > - Noah
> >
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> I was confused about the modules.
>
> Got this warning when upgrading to 1.2.3 even when using the most
> current asterisk-addons and even svn asterisk-addons.
>
>  WARNING WARNING WARNING
>
>  Your Asterisk modules directory, located at
>  /usr/lib/asterisk/modules
>  contains modules that were not installed by this
>  version of Asterisk. Please ensure that these
>  modules are compatible with this version before
>  attempting to run Asterisk.
>
>app_addon_sql_mysql.so
>app_rxfax.so
>app_saycountpl.so
>app_striplsd.so
>app_substring.so
>app_txfax.so
>cdr_addon_mysql.so
>chan_modem_aopen.so
>chan_modem_bestdata.so
>chan_modem_i4l.so
>chan_modem.so
>format_mp3.so
>res_config_mysql.so
>
>  WARNING WARNING WARNING
>
> Do not understand how to fix this?  Do not know if that would also be
> related to the ops crashing.
>
> Dan
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RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 2:45 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection
failure
> 
> On Friday 27 January 2006 16:24, Damon Estep wrote:
> > If you have qualify=yes I assume that triggers a sip query to get
> > channel capabilities from the peer? What is the qualify timeout? Can
it
> > be manipulated?
> 
> qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for
a
> response.  If it does not receive one within 1000ms (by default) and
> qualifysmoothing is not enabled, it will flag the peer as UNREACHABLE
> which
> means that any attempts to Dial() the peer will fail immediately with
> CHANUNAVAIL.  Asterisk continues to send these "pings" until it
receives a
> response within the accepted timeframe and once it gets responses
again it
> will flag the peer as being available once again.
> 
> There are some other tuning parameters which can be used to modify
this
> behaviour slightly but this is what qualify does in a nutshell.

Since your original hint on qualify=yes  have been hunting for the
parameter tuning capabilities of this feature - to no avail. Are you
aware of any reference anywhere on tuning the qualify frequency and
timeout? I assume this (tuning) does not require code changes. Correct?
> 
> > If the goal was strictly to try one provider, and if the channel
fails
> > qualify, then try the next, is the macro you posted needed?
> 
> Correct.
> 
> > Couldn't you just;
> >
> > Exten => ,1,Dial(SIP/[EMAIL PROTECTED]
> > Exten => ,2,Dial(SIP/[EMAIL PROTECTED]
> > Exten => ,3,Congestion(15)
> > Exnte => ,4,Hangup
> 
> Well I've never been a fan of just letting things "fall off the edge"
and
> expecting them to work reliably.  I use the 'g' Dial() option so that
I
> can
> handle failover and call completion correctly or properly -- instead
of
> just
> letting it do "whatever svn trunk deems right at this point" I
> specifically
> do things based on how the call terminated.  It's just a nicer way of
> doing
> what you've provided, and ends up being more robust to code policy
> changes.

Sounds like words of wisdom to me :)
Thanks a million

D
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Re: [Asterisk-Users] Help with Congestion error

[Joseph Tanner]

I think it's a problem on Voicepulse's end, I'm having the same
problems.  If there is a problem on your end, let me know, maybe I
have the same problem.

Personally, I'm using voxee for all my outbound calls with voicepulse
as a backup.  I'd probably pick someone else as a backup (their rates
are a bit high), but I like their auto-fill feature.  If I forget to
refill a prepaid account, I could be in big trouble.  But in my case
the call'd just fail, then try voicepulse next, which will always have
a positive balance.

Quick question, anyone recommend any other decent providers with a
credit card auto-fill option?  Rates aren't as big a deal as
reliability, I'll use the fly-by-night, untested, prepaid-only
providers as the first provider, and a reliable one as the backup.

On 1/27/06, Naren Koka <[EMAIL PROTECTED]> wrote:
> I am using Asterisk with Connect.VoicePulse.  Of late, we are getting too
> many congestion errors. Chris Icide has helped me before in setting up the
> server. He has done a wonderful job. It has worked very well until about 2
> months ago. Now I need some help to fix this issue. I appreciate the help.
>
> Sincerely,
> Naren Koka
> (480) 829-0479
>
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[Asterisk-Users] FlashTransfer to Bridge

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I have a merlin legend that is connected to my asterisk PBX via te110p
- -> 100D on the Legend.  Life is good, except some of their context
features do not work because the legend system does allow those features
from PRI.  (eg: voicemail, call pickup, paging...)

To resolve this I installed a channel back on my second T1/PRI port on
the te110p and connected a couple of FXO ports to the Legend.  I can
dial those features with no problem at all, however it ties up the
analog line of which I only have 3.

What I would like to do is dial,flash,sendtdmf to tranfer back to the
dialing extension via  UDP over the PRI.  How do you break off the
existing channel to pickup the line dial and transfer?

Sean
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFD2pdsy9wPyZpnL2URAiH9AJ4pKKpxC3XtjR/ukwDl5Iact1RMNwCeNMND
w/ke3Gn5qBRx1zO+rvbe5jo=
=MUCD
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Re: [Asterisk-Users] CDR reporting between two Asterisk servers

[kwijibo]

Damon Estep wrote:

Use cdr_mysql

Log your CDRs to a common database
Query as needed from either server using realtime() or from an external
app



Yeah, I thought about that.  If it works how I think it would
work though I would have two CDR records for one call though.
I would have one record from the remote server and one from
the local. Correlating one record with another could be a pain.



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[Asterisk-Users] Page() and Asterisk 1.2.3 Problems?

[Jeremiah Millay]

Has anyone else had problems with the Page() application not working 
under Asterisk 1.2.3?
We use Cisco 7960 phones and set one of the lines to auto answer. When 
someone dials the paging extension it calls the page app and invites all 
the lines on the phones that are set to auto answer into a meetme 
conference where all the members are muted except the original caller.


When I try to use the same dialplan logic that I used in Asterisk 1.2.1 
for paging under Asterisk 1.2.3 it doesn't work. Basically the 
auto-answer lines get called but no one can hear anything. Then within a 
minute or two the asterisk console spits out these messages:


Jan 27 15:09:13 WARNING[5429] app_meetme.c: Unable to write frame to 
channel: Resource temporarily unavailable
Jan 27 15:12:11 WARNING[6228] app_meetme.c: Unable to write frame to 
channel: Inappropriate ioctl for device


Jan 27 15:21:32 DEBUG[6502] app_meetme.c: Ooh, something swapped out 
under us, starting over


The first line a of output above is spammed over and over. The second 
and third show up as well but not as much.


Here is some snippents from extensions.conf

[globals]
INTERCOM=Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL 
PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]

[macro-page]
; Paging macro:
; Check to see if SIP device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
; ${ARG2} - Other line (not paging line...we don't want to disturb 
their other line)

;

exten => s,1,ChanIsAvail(${ARG2}|js) ; j is for dump and s is for ANY 
call. Check to see if line 1 is available

;exten => s,n,Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,1,NoOp(${AVAILSTATUS})
exten => s,n,SIPAddHeader(Call Info: Anwser-After=0) ; This is for the 
Snoms and Others

exten => s,n,NoOp() ; Add others here
exten => s,n,Dial(${ARG1}||)
exten => s,n,Hangup
exten => s,102,Hangup()

[page] ; Paging context

; Two line phone set to auto answer
exten => 2201_com,1,Macro(page,SIP/2201_com,SIP/2201)
exten => 2202_com,1,Macro(page,SIP/2202_com,SIP/2202)
exten => 2203_com,1,Macro(page,SIP/2203_com,SIP/2203)
exten => 2205_com,1,Macro(page,SIP/2205_com,SIP/2205)
exten => 2220_com,1,Macro(page,SIP/2220,SIP/2220) ; One line phone set 
to auto-answer


[home]
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(${INTERCOM}|)



Like I said before this was working with 1.2.1 but 1.2.3 doesn't seem to 
like it. Any help / suggestions would be appreciated if you see a coding 
error.


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[Asterisk-Users] VOXEE Caller ID..

[Ben Higley]

I cannot find any means of passing my own Callerid using Voxee. It always
comes across as NO ID, or nothing, or unknown.

I could not find anything on their website about setting your own caller
id in  the system either. (their web account pages).

Is anyone here using their own Callerid information through Voxee?

thanks


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Re: [Asterisk-Users] Nagios and Asterisk

[JP Carballo]

Darrell Long wrote:

Is anyone using Asterisk (and Festival) to make calls to appropriate 
persons (techs, etc. ) when Nagios generates a particular type of alert?


If so, I would love to hear how people are doing it.

Thanks,

It's as simple as defining a host or service notification command 
consisting of a .call file generating script.

The context it drops to will contain your festival routine.
You'll find the pertinent info on voip-info.org

You then give Nagios the proper contact host or service notification 
options so that voice won't nag you when everything's fine.


I admit I no longer have festival running.
It's a good idea at first but after you hear the voice one too many 
times for an error message you will find yourself switching to email/sms :)
I just keep the option activated for the possibility that my internet 
provider will fail, but I use my own recorded message now.


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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[Asterisk-Users] moh & clock

[Dov Bigio]

Hi,
 
I had a wct1xxp in my asterisk server, but I 
migrated to a cisco sip gateway, and then unplugged the e1.
I then changed zaptel's Makefile to include ztdummy 
and ran modprobe ztdummy
 
Music on hold for queues is not working well... it 
is simply mute.
 
I realized that, while waiting on a Queue, if I ran 
a reload, the music on hold starts being played for a few seconds and then 
stops, until I reload again.
 
I am using 1.2.3, but this happens to me since 
1.2.0 (it worked well on 1.0.10).
 
When I ran lsmod, I see
 
usb-uhci   
26860   0  
[ztdummy]zaptel    
183680  78  [ztdummy wcusb wct1xxp]
Does this make sense? Should I recompile zaptel? 
How do I remove wct1xxp? (the card is actually there, but it has no E1 in it 
anymore.
 
Thank you very much 
Dov
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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure

[Andrew Kohlsmith]

On Friday 27 January 2006 16:24, Damon Estep wrote:
> If you have qualify=yes I assume that triggers a sip query to get
> channel capabilities from the peer? What is the qualify timeout? Can it
> be manipulated?

qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for a 
response.  If it does not receive one within 1000ms (by default) and 
qualifysmoothing is not enabled, it will flag the peer as UNREACHABLE which 
means that any attempts to Dial() the peer will fail immediately with 
CHANUNAVAIL.  Asterisk continues to send these "pings" until it receives a 
response within the accepted timeframe and once it gets responses again it 
will flag the peer as being available once again.

There are some other tuning parameters which can be used to modify this 
behaviour slightly but this is what qualify does in a nutshell.

> If the goal was strictly to try one provider, and if the channel fails
> qualify, then try the next, is the macro you posted needed?

Correct.

> Couldn't you just;
>
> Exten => ,1,Dial(SIP/[EMAIL PROTECTED]
> Exten => ,2,Dial(SIP/[EMAIL PROTECTED]
> Exten => ,3,Congestion(15)
> Exnte => ,4,Hangup

Well I've never been a fan of just letting things "fall off the edge" and 
expecting them to work reliably.  I use the 'g' Dial() option so that I can 
handle failover and call completion correctly or properly -- instead of just 
letting it do "whatever svn trunk deems right at this point" I specifically 
do things based on how the call terminated.  It's just a nicer way of doing 
what you've provided, and ends up being more robust to code policy changes.

-A.
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[Asterisk-Users] How's the best way to set up agents...

[Ben Ferguson]

Title: Message



So I'm trying to set 
up queues and agents and am trying to figure out the best way to set up what I 
need to do.  And what I need to do is basically get Asterisk to mimic my 
company's current phone system.  As close as possible of course.  And 
my main problem is queues and agents.  Currently, for our queues 
and agents, a person is assigned a hot-desk extension, which they use to login 
to any phone and then they can send and receive calls at that 
extension.  There is no seperate extension 
and agent id--they are pretty much the same thing.  But the 
extension moves around with them to wherever they log in.  The 
advantage is that they always have the same extension.  When no one is 
logged into a phone, the phone is assigned a catch all username called "no user" 
which has limited dialing capabilities.  With Asterisk, when you log 
in an agent, they assume the extension of the phone that they have just 
logged in under.  Yes, if they are a member of a queue, they will 
always receive calls from that queue regardless of what extension they are at, 
but for DID and internal calls, you would never know which extension to 
dial to reach a person setup in such a way.
 
So here's what 
I've come up with (but I, of course, still have questions...):  
Match the 
agent ID to an extension.  Assign an agent their ID and then 
assign a certain working area, and a assign certain phone to that working 
area and assign that phone an extension that is the same as 
their agent id.  The pitfall here is that if you do it this 
way, only one person could utilize that working area and that phone.  
Agents working in shifts at the same work area and same phone would 
not work--unless multiple agents used the same extension, which again kills your 
DID and internal calling.  Of course, you could assign the extension the 
phone based on time, but what if you want to have open seating and you want any 
agent to sit in any work area at any given time?  You can see it starts to 
get messy when you try to work it out this way.  Also, if you did 
do the matched extension and agent id, if a person was re-assigned to a new work 
area, they either have to take their phone with them, or you get to 
setup a new phone with their extension.  Perhaps there is something in 
Asterisk that I don't know about that could benefit me here?
 
I'm thinking another 
way to do something like what I need is via XML, but I'm not exactly sure how to 
do it this way.  Can you assign a phone a certain extension and then give 
it an option for "logging in" using their agent id and then based on their agent 
id, push a new XML file out that assigns their specific extension.  Can you 
re-assign a new extension to a phone this way?  I believe this would be a 
decent way to set this up (if the XML files aren't too complicated) but I'm not 
exactly sure how to do it.  
 
Any suggestions, 
pointers, directions...?
 
Thanks!
Ben 
Ferguson
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RE: [Asterisk-Users] CDR reporting between two Asterisk servers

[Damon Estep]

Use cdr_mysql

Log your CDRs to a common database
Query as needed from either server using realtime() or from an external
app

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Friday, January 27, 2006 2:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CDR reporting between two Asterisk servers
> 
> Anybody know if there is a way to get one Asterisk
> server to report CDR information to another Asterisk
> server?
> 
> I have a local Asterisk server and a remote Asterisk
> server.  They talk via IAX2.  I have calls that go
> as follows:
> 
> PSTN->remote server->IAX2->local server->IP phone
> ->IP phone
> 
> When a call comes in from the PSTN it rings phones
> at the remote site and the local site.  This
> call is bridged so whoever picks it up first gets
> the call.  I am trying to find a way for the local
> server to report back to the remote server with
> who answered the phone at the local site.
> 
> Is this even possible?
> 
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Re: [Asterisk-Users] * point to point t1 solution?

[Andrew Kohlsmith]

On Friday 27 January 2006 15:56, Damon Estep wrote:
> that was the goal - a end to end TDMoE path terminated at both ends at
> the t1 ports on the Digium card - aware of the idle bandwidth
> requirements (about 2.0mbps including packet overhead).

I don't think it's possible to get "no signaling awareness" with "idle 
bandwidth awareness" -- they don't mix.  You either deterimine idle channels 
by watching the signaling (CAS or CCS) or you don't and pass everything 
unmolested, including idle channel "data".

> I have not seen any information that says that it is possible to nail a
> t1 port to a TDMoE channel group on one end and vice versa so the
> signaling could be passed unmodified.

Admission: I have never used TDMoE, but I understand telephony and the concept 
behind TDMoE and TDMoIP very well.

TDMoE just takes the 1544000 bits per second (192 bits of 24 channel data + 1 
bit of T1 frame data, send 8000 times per second) that a T1 produces and 
encapsulates it in raw ethernet frames.  This is going from memory but I 
believe it takes the 24 timeslots (24 bytes) * 8 plus the 8 framing bits (+1 
byte) and stuffs it into an ethernet frame and sends it on its way.  That 
would certainly give you 1000pps as someone else mentioned.

Again (from memory, and from looking at ztd-eth driver it looks to be right) 
there is no interpretation or manipulation of the signaling data done at all.   
The bits are passed as-is, which means if you're using the MCDN/NAPN/SL1 
protocol between Norstar systems that it should all work.

> You have to decode the signaling, pass the media, and reproduce the
> signaling via asterisk at the remote end, correct? This means that id
> the signaling IS non-standard it is a no go.

Only if you're trying to convert the T1 channel data to SIP or otherwise 
manipulate the bits.  If you're only trying to get them to go from A to B 
then no, no interpretation is needed.

If you're looking for a straight T1-ethernet bridge, TDMoE should do the trick 
for you.  If your microwave link appears as an ethernet bridge (not an IP 
router) and it's robust enough and has enough excess (or the ability to 
prioritize ethernet frames based on MAC address) then this should work.

If your microwave link is indeed routing you should still be able to get it to 
work if you do some fancy tunneling.  Again, not impossible but you really 
need to do some testing.  

Not having done this specific implementation before I of course cannot make 
any guarantees.

> If it could be done, signaling matters none - the two connected devices
> would still understand each others foreign language (signaling) and the
> asterisk boxes would just provide a pure Nx64 data path.

Exactly correct, and as I said several times in this email I am not aware of 
TDMoE caring one iota about what the bits represent.  It merely passed them 
from A to B.

It's certainly an interesting application.

-A.
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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Julian Lyndon-Smith]

These modules are not part of the standard 1.2.3 release - did you also 
install the 1.2.3 release of the asterisk-addons package ?


If * is loading older modules (which it probably is because of your 
config files) then it may cause grief ;)


My .2p worth. Probably not helpful, but maybe, just maybe 

Julian

Dan Littlejohn wrote:

On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote:

Hi Brent -


Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.

Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server would reply to a
ping, but no ssh login, no local console login - just locked up. This ain't
good for business.


We've been doing fine with 1.2.3 so far.  No problems reported, though I
only have it deployed in a small office.  Definitely no lock-ups.

On the asterisk side, just a basic question - did you make sure to remove
the old modules so the new 1.2.3 versions got installed?

As far as the lockups, maybe it is coincidental?  I've never had asterisk
(even the crazy CVS versions) lock a whole OS like that.  I have had
machines running asterisk lock up, but it always turned out to be caused by
something else like bad hardware, or unrelated network problems.

- Noah

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I was confused about the modules.

Got this warning when upgrading to 1.2.3 even when using the most
current asterisk-addons and even svn asterisk-addons.

 WARNING WARNING WARNING

 Your Asterisk modules directory, located at
 /usr/lib/asterisk/modules
 contains modules that were not installed by this
 version of Asterisk. Please ensure that these
 modules are compatible with this version before
 attempting to run Asterisk.

   app_addon_sql_mysql.so
   app_rxfax.so
   app_saycountpl.so
   app_striplsd.so
   app_substring.so
   app_txfax.so
   cdr_addon_mysql.so
   chan_modem_aopen.so
   chan_modem_bestdata.so
   chan_modem_i4l.so
   chan_modem.so
   format_mp3.so
   res_config_mysql.so

 WARNING WARNING WARNING

Do not understand how to fix this?  Do not know if that would also be
related to the ops crashing.

Dan
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RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 2:07 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection
failure
> 
> On Friday 27 January 2006 16:00, Damon Estep wrote:
> > In the event that the first attempt DOES NOT RESPOND (is down) there
has
> > to be a timeout value to go to the next priority, correct? Otherwise
the
> > channels just sits silent waiting for a response.
> 
> That's what the qualify parameter in sip/iax.conf is for.  Never
terminate
> calls without it.  :-)  It won't *guarantee* that you'll never get
dead
> air,
> but it sure goes a long way to ensuring that it happens so
infrequently
> you'll think you misdialed.
> 
> > I think your macro assumes that you got a response from nufone, but
what
> > if they were dead in the water?
> 
> Then qualify would have failed and Dial() would have immediately
returned
> CHANUNAVAIL.
> 
> -A.

OK - starting to make sense now

Qualify=yes for the peer in sip.conf

If you have qualify=yes I assume that triggers a sip query to get
channel capabilities from the peer? What is the qualify timeout? Can it
be manipulated?

If the goal was strictly to try one provider, and if the channel fails
qualify, then try the next, is the macro you posted needed?

Couldn't you just;

Exten => ,1,Dial(SIP/[EMAIL PROTECTED]
Exten => ,2,Dial(SIP/[EMAIL PROTECTED]
Exten => ,3,Congestion(15)
Exnte => ,4,Hangup



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[Asterisk-Users] CDR reporting between two Asterisk servers

[kwijibo]

Anybody know if there is a way to get one Asterisk
server to report CDR information to another Asterisk
server?

I have a local Asterisk server and a remote Asterisk
server.  They talk via IAX2.  I have calls that go
as follows:

PSTN->remote server->IAX2->local server->IP phone
   ->IP phone

When a call comes in from the PSTN it rings phones
at the remote site and the local site.  This
call is bridged so whoever picks it up first gets
the call.  I am trying to find a way for the local
server to report back to the remote server with
who answered the phone at the local site.

Is this even possible?

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Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Dan Littlejohn]

On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote:
> Hi Brent -
>
> > Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
> > the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
> > hours or so.
> >
> > Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
> > on Thursday, and then about a half hour ago. The server would reply to a
> > ping, but no ssh login, no local console login - just locked up. This ain't
> > good for business.
>
>
> We've been doing fine with 1.2.3 so far.  No problems reported, though I
> only have it deployed in a small office.  Definitely no lock-ups.
>
> On the asterisk side, just a basic question - did you make sure to remove
> the old modules so the new 1.2.3 versions got installed?
>
> As far as the lockups, maybe it is coincidental?  I've never had asterisk
> (even the crazy CVS versions) lock a whole OS like that.  I have had
> machines running asterisk lock up, but it always turned out to be caused by
> something else like bad hardware, or unrelated network problems.
>
> - Noah
>
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>


I was confused about the modules.

Got this warning when upgrading to 1.2.3 even when using the most
current asterisk-addons and even svn asterisk-addons.

 WARNING WARNING WARNING

 Your Asterisk modules directory, located at
 /usr/lib/asterisk/modules
 contains modules that were not installed by this
 version of Asterisk. Please ensure that these
 modules are compatible with this version before
 attempting to run Asterisk.

   app_addon_sql_mysql.so
   app_rxfax.so
   app_saycountpl.so
   app_striplsd.so
   app_substring.so
   app_txfax.so
   cdr_addon_mysql.so
   chan_modem_aopen.so
   chan_modem_bestdata.so
   chan_modem_i4l.so
   chan_modem.so
   format_mp3.so
   res_config_mysql.so

 WARNING WARNING WARNING

Do not understand how to fix this?  Do not know if that would also be
related to the ops crashing.

Dan
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[Asterisk-Users] Help with Congestion error

[Naren Koka]

I am using Asterisk with Connect.VoicePulse.  Of late, we are getting too 
many congestion errors. Chris Icide has helped me before in setting up the 
server. He has done a wonderful job. It has worked very well until about 2 
months ago. Now I need some help to fix this issue. I appreciate the help.


Sincerely,
Naren Koka
(480) 829-0479

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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure

[Andrew Kohlsmith]

On Friday 27 January 2006 16:00, Damon Estep wrote:
> In the event that the first attempt DOES NOT RESPOND (is down) there has
> to be a timeout value to go to the next priority, correct? Otherwise the
> channels just sits silent waiting for a response.

That's what the qualify parameter in sip/iax.conf is for.  Never terminate 
calls without it.  :-)  It won't *guarantee* that you'll never get dead air, 
but it sure goes a long way to ensuring that it happens so infrequently 
you'll think you misdialed.

> I think your macro assumes that you got a response from nufone, but what
> if they were dead in the water?

Then qualify would have failed and Dial() would have immediately returned 
CHANUNAVAIL.

-A.
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[Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Noah Miller]

Hi Brent - 

> Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
> the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
> hours or so.
> 
> Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
> on Thursday, and then about a half hour ago. The server would reply to a
> ping, but no ssh login, no local console login - just locked up. This ain't
> good for business.


We've been doing fine with 1.2.3 so far.  No problems reported, though I
only have it deployed in a small office.  Definitely no lock-ups.

On the asterisk side, just a basic question - did you make sure to remove
the old modules so the new 1.2.3 versions got installed?

As far as the lockups, maybe it is coincidental?  I've never had asterisk
(even the crazy CVS versions) lock a whole OS like that.  I have had
machines running asterisk lock up, but it always turned out to be caused by
something else like bad hardware, or unrelated network problems.

- Noah

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RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure

[Damon Estep]

Andrew,

Thanks for this - I have also been looking for a way to "fail over"
calls to a second SIP path, but;

In the event that the first attempt DOES NOT RESPOND (is down) there has
to be a timeout value to go to the next priority, correct? Otherwise the
channels just sits silent waiting for a response.

I think your macro assumes that you got a response from nufone, but what
if they were dead in the water?

Have I missed something?

Is there a way to modify the relevant SIP timer so if the INVITE is not
ack'd in a specific period of time then the next priority is executed?

Damon

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 1:12 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection
failure
> 
> On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
> > I am trying to tweak my dial plan and I am running into a problem.
> > Sometimes my VoIP out bound calls do not complete on overseas
calls(busy
> > or just a hang-up).  Is there a way in the dial plan to
automatically
> > dial out of my PRI when something like this happens.  Either by time
> > limit by a failure event?
> 
> ; call $ARG1 through nufone, failing over to the PRI.
> [macro-nufone-dial]
> exten => s,1,Dial(SIP/[EMAIL PROTECTED],,go)
> exten => s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS
is
> ${DIALSTATUS})
> exten => s,n,Goto(dial-${DIALSTATUS},1)
> 
> exten => dial-CANCEL,1,Hangup
> exten => dial-ANSWER,1,Hangup
> exten => dial-NOANSWER,1,Hangup
> exten => dial-BUSY,1,Busy
> exten => dial-CONGESTION,1,Congestion
> exten => dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
> 
> It really is as simple as that.  :-)
> 
> -A.
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RE: [Asterisk-Users] * point to point t1 solution?

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 1:19 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] * point to point t1 solution?
> 
> On Thursday 26 January 2006 09:00, Damon Estep wrote:
> > The same trunking could be achieved with SIP or IAX, could it not
(with
> > higher latency)?
> 
> What signaling are you running on this ptp T1?  If it's MCDN (NAPN)
then
> forget it; Asterisk doesn't understand the signaling.  It's actually a
> side
> project of mine to bring that to Asterisk.  :-)
> 
> > This is what would be required to truly emulate a "signaling
un-aware"
> > point to point t1 like one that you would get from a telco if you
> ordered a
> > point to point esf/b8zs t1 from A location to Z location.
> 
> Not possible unless you want to transmit the entire T1 contents
(signaling
> and
> channels, inuse or not) continuously.  c.f. "TDMOE"  :-)
> 
> -A.


that was the goal - a end to end TDMoE path terminated at both ends at
the t1 ports on the Digium card - aware of the idle bandwidth
requirements (about 2.0mbps including packet overhead).

I have not seen any information that says that it is possible to nail a
t1 port to a TDMoE channel group on one end and vice versa so the
signaling could be passed unmodified.

You have to decode the signaling, pass the media, and reproduce the
signaling via asterisk at the remote end, correct? This means that id
the signaling IS non-standard it is a no go.

If it could be done, signaling matters none - the two connected devices
would still understand each others foreign language (signaling) and the
asterisk boxes would just provide a pure Nx64 data path.
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RE: [Asterisk-Users] Good provider of Polycom Phones (mostly for accessto latest/greatest firmware)

[Douglas Garstang]

Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a 
month ago and we're still waiting. Our account rep's communication with us on 
what the delay has been, has been terrible.

Doug.

-Original Message-
From: Gavin Adams [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 8:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Good provider of Polycom Phones (mostly for
accessto latest/greatest firmware)


Hi,

I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us maintenance and
access to the most current firmware.

What are some good resellers out there?

Regards,

--- Gavin Adams
VP of Technology
Promisant (USA) Inc.

Email: [EMAIL PROTECTED] 

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Re: [Asterisk-Users] AAH out bound routing problem

[J.Raborg]

Go to the 
setup/Outbound Routing/
pick the one you use for out must be 0 9out  by default and add
in the dial patterns  9|. <-including the dot

Then try to dial 919197543700

JR


ram wrote:

  Hi rajeev
   
  i have posted the extension.conf before
  now iam posting extension_additional.conf
   
  [EMAIL PROTECTED] asterisk]# more extensions_additional.conf
[globals]
#include globals_custom.conf
VM_PREFIX = *
RINGTIMER = 15
REGTIME = 7:55-17:05
REGDAYS = mon-fri
RECORDEXTEN = ""
PARKNOTIFY = SIP/200
OUT_2 = SIP/easycall
OUTPREFIX_2 =
OUTMAXCHANS_2 = 1
  OUTCID_2 = outside account
  OPERATOR =
NULL = ""
IN_OVERRIDE = forcereghours
INCOMING = group-all
FAX_RX_EMAIL = [EMAIL PROTECTED]
FAX_RX = system
FAX =
DIRECTORY_OPTS =
DIRECTORY = last
  
DIAL_OUT = 9
DIAL_OPTIONS = tr
DIALOUTIDS = 2/
CALLFILENAME = ""
AFTER_INCOMING =
  [ext-local]
include => ext-local-custom
exten => 1000,1,Macro(exten-vm,1000,1000)
exten => ${VM_PREFIX}1000,1,Macro(vm,1000)
exten => 1000,hint,SIP/1000
  [outbound-allroutes]
include => outbound-allroutes-custom
include => outrt-001-longdistance
  [outrt-001-longdistance]
include => outrt-001-longdistance-custom
exten => _1NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)
exten => _1NXXNXX,2,Macro(outisbusy)    ; No available
circuits
exten => _NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)
  
exten => _NXXNXX,2,Macro(outisbusy) ; No available circuits
exten => _NXX,1,Macro(dialout-trunk,2,${EXTEN},)
exten => _NXX,2,Macro(outisbusy)    ; No available circuits
  
  
 
   
  ram 
 
  On 1/27/06, Rajeev Natarajan <[EMAIL PROTECTED]>
wrote:
  if
you are using AAH, please post extensions.conf,
extensions_additional.conf - also send us more info on your phones.


thanks
rajeev


ram wrote:
> Hi
>
> all of them thanks for the quick reply
>
> i was tried adding 9 as well as 00
> but i get number invalid if i put any of the digits

>
> what kind of config files need to post here to resolve the problem
>
> please assists
>
> ram
>
>
> On 1/27/06, *Michael Collins* <
[EMAIL PROTECTED]
> [EMAIL PROTECTED]>>
wrote:
>
> Ram,
>
>
>
> On my AAH the stock dial plan requires a 9 first.  For kicks,
try

> dialing 919197543700 and see what you get.
>
>
>
> -MC
>
>
>
>

>
> *From:* [EMAIL PROTECTED]
> [EMAIL PROTECTED]
>
> [mailto:[EMAIL PROTECTED]
> [EMAIL PROTECTED]
> ] *On Behalf Of *ram
> *Sent:* Friday, January 27, 2006 6:14 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [Asterisk-Users] AAH out bound routing problem

>
>
>
> Hi all
>
>
>
> I have installed AAH 2.2 in my P4 PC
>
>
>
> following AAH handbook PDF and 
http://mundy.org/blog/index.php?p=62#amp
>
>
>
> and made as per the guide says
>
>
>
> and downloaded SJ Phone, and registered user
>
>
>
> and when i try to dial the 19197543700

>
>
>
>
> i get message that, all circuits are busy now, please try your
call
> later
>
>
>
> and when i see in the console i get this mesage
>

>
>
> any help
>
>
>
> Called easycall/19197543700
> -- Got SIP response 488 "Not acceptable here" back from
(PeerIP)
> -- SIP/easycall-838e is circuit-busy

>
>
>
> ram
>
>
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>
>
>

>
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[Asterisk-Users] starvox communications

[mezzmor]

Anyone have any experience with them...specifically, how to actually use their service from an asterisk server? They only do SIP, have no proxy, don't accept registrations, and only allow or disallow calls based on client IP address. 
 
I am at a loss as to how to make asterisk work this way.
 
Thanks
 
Jason

Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection.


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Re: [Asterisk-Users] * point to point t1 solution?

[Andrew Kohlsmith]

On Thursday 26 January 2006 09:00, Damon Estep wrote:
> The same trunking could be achieved with SIP or IAX, could it not (with
> higher latency)?

What signaling are you running on this ptp T1?  If it's MCDN (NAPN) then 
forget it; Asterisk doesn't understand the signaling.  It's actually a side 
project of mine to bring that to Asterisk.  :-)

> This is what would be required to truly emulate a "signaling un-aware"
> point to point t1 like one that you would get from a telco if you ordered a
> point to point esf/b8zs t1 from A location to Z location.

Not possible unless you want to transmit the entire T1 contents (signaling and 
channels, inuse or not) continuously.  c.f. "TDMOE"  :-)

-A.
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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure

[Andrew Kohlsmith]

On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
> I am trying to tweak my dial plan and I am running into a problem.
> Sometimes my VoIP out bound calls do not complete on overseas calls(busy
> or just a hang-up).  Is there a way in the dial plan to automatically
> dial out of my PRI when something like this happens.  Either by time
> limit by a failure event?

; call $ARG1 through nufone, failing over to the PRI.
[macro-nufone-dial]
exten => s,1,Dial(SIP/[EMAIL PROTECTED],,go)
exten => s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is 
${DIALSTATUS})
exten => s,n,Goto(dial-${DIALSTATUS},1)

exten => dial-CANCEL,1,Hangup
exten => dial-ANSWER,1,Hangup
exten => dial-NOANSWER,1,Hangup
exten => dial-BUSY,1,Busy
exten => dial-CONGESTION,1,Congestion
exten => dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})

It really is as simple as that.  :-)

-A.
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Re: [Asterisk-Users] Re: * point to point t1 solution? / alternatives

[Andrew Kohlsmith]

On Thursday 26 January 2006 16:10, Damon Estep wrote:
> The question has been answered; Asterisk is NOT capable of providing a
> clear PTP TDM path between two boxes configured with t1 cards and any
> type of trunking.

I don't know about that; I have been pushing phone and fax (yes fax) over a 
1-hop SDSL loop for the past 18 months.  I don't imagine your microwave link 
would be any different unless it was saturated.

Depending on what the link looks like to the computer, you very well might be 
able to use TDMoE to do what you are looking for.

-A.
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[Asterisk-Users] Lockups since upgrade 1.2.3 - anyone else? Any ideas?

[Brent Torrenga]

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.

Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server would reply to a
ping, but no ssh login, no local console login - just locked up. This ain't
good for business.

I have scoured the logs, and find no errors. Not even right before/around
the time of the crash.

I am worried that 1.2.3 is not as stable as 1.0.9 (or 1.0.10, though we
never ran that version). Is there a needed step aside from "make; make
install" that I missed when upgrading? Has anyone else had similar problems?
Or, if I submit other info, would someone have a clue as to what to look at?
We run a TDM400P with 3 FXO modules, and about 15 SIP Cisco 79XX phones
here. Any help is appreciated, this cannot continue.

Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com

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Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service

[RumaTech]

I tried through voipdiscount as well.
Even my older account through voipbuster started to behave this way and it 
used to be ok on IAX.


I would expect at least some reply.

Rudolf

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


Sent: Thursday, January 26, 2006 5:00 PM
Subject: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service


Didn't you read this from their Q&A?

I want to configure my own IAX/SIP device for calling with VoipBuster, is 
that possible?
It is possible to use your own IAX/SIP device, however we do not support it. 
We advise you to use SIP-Discount instead.


Do you have the same problem when you use their softphone? If not, why 
complaining.


The call to the UK is free only for VoIPstunt

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


Sent: Thursday, January 26, 2006 6:19 AM
Subject: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service



Hi, all

I am reallty pissed with their service. I wonder if this is common 
problem.

Firstly, all of my calls are terminated after 30s. And termination happens
in a strange way. My local asterisk server does not see the disconnection,
but remote party is disconnected. Basically, I am still on the phone, 
while

remote party was disconnected. When I hang up, I get something like that:

Apr 20 02:32:43 WARNING[4853]: chan_sip.c:8520 handle_response: Got
authentication request (401) on unknown BYE to
';tag=c9ebef50c90078c2c93eddc243d7352d6e04'

Secondly, they charged me for calls to UK that was supposed to be free.
And their customer service does not respond at all. Do they have a phone
number I can call?

Thanks,
Rudolf

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RE: [Asterisk-Users] Linksys SPA-941 multiple line appearences

[Kerry Garrison]

Accounts 3-4 are disabled, Account 1 is the only account. 
Thats it. Nothing special. If you have problems, try doing *70. 

-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  tracinetSent: Wednesday, January 25, 2006 8:57 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Linksys SPA-941 multiple line 
appearences
  This may sound odd, but I have the opposite problem.  I am 
  trying to disable call waiting by only allowing 1 call to come into the 
  SPA-941, but I am getting 2 calls per line key.  If have 1 extension set 
  up on all 4 line keys, the phone handles 8 incoming calls.  I have a few 
  customers that like that feature, but others that want to just have 1 call - 
  any tips on what you did to only get 1 call to ring through without doing any 
  status checking in the asterisk dialplan would be appreciated.- 
  Pedro
  On 1/25/06, Michael 
  Keyes <[EMAIL PROTECTED]> 
  wrote:
  I 
set up only 1 extension and set all 4 line appearences to point to 
thatextension.  I could place up to 4 outgoing calls as 
extension 1 no problem.The problem happened when there was an active 
call on line appearence 1 and someone called extension 
1.  Instead of ringing on line appearence 2(extension 1) it 
would busy out.  My question, is there something inAsterisk 
that needs to be adjusted to make the phone work properly or isthere a 
setting on the phone that needs to be adjusted?  Thank 
you.Michael K-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]]On 
Behalf Of KerryGarrisonSent: Tuesday, January 24, 2006 10:33 
PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: RE: [Asterisk-Users] Linksys SPA-941 multiple line 
appearencesYou only need to setup ONE account and all four call 
appearances will work.-Kerry> -Original 
Message-> From: [EMAIL PROTECTED]> 
[mailto:[EMAIL PROTECTED]] 
On Behalf Of > Michael Keyes> Sent: Tuesday, January 24, 2006 
3:41 PM> To: asterisk-users@lists.digium.com> 
Subject: [Asterisk-Users] Linksys SPA-941 multiple line appearences 
>> Has anyone had any experience with the Linksys SPA-941 
when> it comes to multiple line appearences?>> The 941 
comes with 4 line appearence buttons which can> individualy be 
configured to point at any extension.  The > phone is 
capable of 2 extensions out of the box with the> option to 
add  2 more for a license fee.>> The 941 manual 
states that, "The Call Waiting function is> activated when a device 
has a call in the active state and > another call is incoming. The 
phones in the SPA series do not> support multiple calls on the same 
Line Key.> Incoming calls are assigned to an unused Line Key, 
causing> the Line Key to quickly blink red. (Note that the Voice Mail 
> Waiting Indicator also blinks red whenever there is an> 
incoming call.) The phone will not ring. However, to alert> the user, 
the call waiting tone is played into the active> audio device." 
>> During testing I set up an Asterisk 1.2 box with a 941 
phone> using firmware ver. 4.1.8.  I configured 1 extension 
and set> all 4 line appearence buttons to point to that 
extension.  If> there was an active call in progress I 
could place that call > on hold and by pressing line appearence 
button 2 was able to> place an outgoing call.  That 
outgoing call would appear to> come from extension 1.  This 
is all working as desired.>> If an active call was in progress 
and someone called my > extension the product manual indicates that 
call should> appear on line appearence button 2.  During my 
testing> Asterisk would flag my extension 1 as busy and instead 
of> ringing the phone on line appearence 2 would send the call to 
> voicemail.>> Is anyone aware of any configuration 
setting needed on either> the Asterisk server or on the phone to make 
Call Waiting> function as described in the manual?  Thank 
you.>> Michael K>>> 
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Re: [Asterisk-Users] Nagios and Asterisk

[Joseph Tanner]

I have used both, just not together.  I have a possible idea though.
If they're running on separate servers, you can have nagios send an
email that the asterisk server receives.  Have different email aliases
for different alerts, or have a script parse the email to see what
kind of alert it is.  Have this script generate a .call file in
/var/spool/asterisk/outgoing based on the type of alert.  If they're
running on the same server you might be able to skip having to send an
email (but if not, then just have it send an email to a local user,
it'll work the same).

Personally, I just had Nagios send an email whenever there was a
problem.  If the tech is in front of their workstation, they'll get a
notice immediately.  If not, you could have a text message sent
instead.  Worked great for me.

On 1/27/06, Darrell Long <[EMAIL PROTECTED]> wrote:
> Is anyone using Asterisk (and Festival) to make calls to appropriate
> persons (techs, etc. ) when Nagios generates a particular type of alert?
>
> If so, I would love to hear how people are doing it.
>
> Thanks,
>
> --
> Darrell S. Long
> BestWeb Corporation
>
>
>
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Re: [Asterisk-Users] Re: Good provider of Polycom Phones (mostly for access to latest/greatest firmware)

[Mojo with Horan & Company, LLC]

amen! http://www.tritechcoa.com/ is a great supplier :)

Noah Miller wrote:
Hi Gavin - 




I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us maintenance and
access to the most current firmware.

What are some good resellers out there?



I love PC Connection for most of my ordering, but I've actually never used
them for Polycom hardware, even though we have a lot of it.

My Polycom supplier has been http://www.tritechcoa.com

They are extremely responsive, and have the best prices around on Polycom.
When I recently asked for the latest firmware, I got a response back in
about 2 minutes.  Now I actually have firmware newer than what is listed on
the Polycom website.  I haven't tested it out yet, though.

- Noah







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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] SIP channel not diconnecting on hangup

[Scott Bussinger]

I've got an SPA-841 SIP hardphone connecting to my asterisk server across 
the internet through a couple of NAT routers. Everything works great (I can 
initiate calls, receive calls, hear audio both ways, etc.) except for one 
thing. When I hang up the phone, the connection in asterisk doesn't 
disconnect. The phone is idle and things everything is fine, but Asterisk 
still show an open channel. It's like the phone isn't sending some sort of 
disconnect message to Asterisk. Can anyone provide some ideas on what might 
be going wrong?

As a test case, I call my echo() extension from the remote phone. The 
connection works fine but when I hangup the phone and get information from 
the Asterisk console here's what I see:

[Jan 27 10:27:00] -- Executing Playback("SIP/scottbhome-f4de", 
"demo-echotest") in new stack
[Jan 27 10:27:00] -- Playing 'demo-echotest' (language 'en')
[Jan 27 10:27:19] -- Executing Echo("SIP/scottbhome-f4de", "") in new 
stack

 I hangup the phone here 

pbx*CLI> show channels
Channel  Location State   Application(Data)
SIP/scottbhome-f4de  [EMAIL PROTECTED]:2  Up  Echo()
1 active channels
1 active calls

pbx*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last 
Message
xx.xx.xx.xxscottbhome  304dcbc8-5f  00101/00102  g729  No   Rx: 
INVITE
1 active SIP channels


So the connection initiates correctly, but nothing ever terminates it. I 
finally do a SOFT HANGUP to kill the connection.

Thanks for any help! 



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[Asterisk-Users] Nagios and Asterisk

[Darrell Long]

Is anyone using Asterisk (and Festival) to make calls to appropriate 
persons (techs, etc. ) when Nagios generates a particular type of alert?


If so, I would love to hear how people are doing it.

Thanks,

--
Darrell S. Long
BestWeb Corporation

 	  


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[Asterisk-Users] chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1

[Nilesh Londhe]

Editing subject line to reflect current status.

On 1/26/06, Nilesh Londhe <[EMAIL PROTECTED]> wrote:
> Since T616 is not answering (and incoming calls are going to Cingular
> voicemail after 30 sec,) I suspect the problem focus area is...
>
> >-- Executing Answer("BLT/T616", "") in new stack
>
> Is http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz
> tar xzf bluetoothfiles.tar.gz the latest source (r40?)
>
> On 1/26/06, Nilesh Londhe <[EMAIL PROTECTED]> wrote:
> > Here are my findings with my experiment using Sony Erisson T616 with
> > Cingular Service and connected to [EMAIL PROTECTED] 2.2 on a freshly
> > installed system and following the instructions
> > http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
> >
> > Outbound calls (Asterisk to T616 via bluetooth):
> >
> > Works OK via Dial(BLT/T616/8005551212)
> >
> > Inbound calling (T616 to asterisk via bluetooth):
> >
> > My configuration for inbound calls:
> >
> > [bluetooth]
> >  exten => s,1,Wait(1)
> >  exten => s,2,Answer
> >  exten => s,3,Dial(SIP/1007,15,rtT)
> >  exten => s,4,VoiceMail([EMAIL PROTECTED])
> >  exten => s,5,Hangup
> >
> > My observation:
> >
> > When I call my cell T616 from my landline, SIP/1007 rings for 2
> > seconds and the call is answered by Cingular voicemail not by asterisk
> > voicemail. My cingular voicemail is set to answer in 30 seconds after
> > first ring.
> >
> > Output on the asterisk CLI:
> >
> > [EMAIL PROTECTED] ~]# asterisk -r
> > Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
> > Written by Mark Spencer <[EMAIL PROTECTED]>
> > =
> > Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 3025)
> > Verbosity is at least 3
> >  [AG]   T616 > +CIEV: 2,4
> >  [AG]   T616 > +CIEV: 2,3
> >  [AG]   T616 > RING
> >  [AG]   T616 > +CLIP: "421212",161,,,"Landline"
> >-- Executing Wait("BLT/T616", "1") in new stack
> >-- Executing Answer("BLT/T616", "") in new stack
> >  [AG]   T616 < +CIEV: 2,1
> >  [AG]   T616 < +CIEV: 3,0
> >-- Executing Dial("BLT/T616", "SIP/1007|15|rtT") in new stack
> >-- Called 1007
> >-- SIP/1007-d97e is ringing
> >  == Spawn extension (bluetooth, s, 3) exited non-zero on 'BLT/T616'
> >  [AG]   T616 < ATH
> >  [AG]   T616 < AT+CHUP
> >  [AG]   T616 > ERROR
> >  [AG]   T616 > OK
> >  [AG]   T616 < AT+BRSF=23
> >  [AG]   T616 > ERROR
> >  [AG]   T616 < AT+CIND=?
> >  [AG]   T616 > +CIND:
> > ("battchg",(0-5)),("signal",(0-5)),("batterywarning",(0-1)),("chargerconnected",(0-1)),("service",(0-1)),("sounder",(0-1)),("message",(0-1)),("call",(0-1)),("roam",(0-1)),("smsfull",(0-1))
> >  [AG]   T616 > OK
> >  [AG]   T616 < AT+CIND?
> >  [AG]   T616 > +CIND: 5,3,0,1,1,0,0,0,0,0
> >  [AG]   T616 > OK
> >  [AG]   T616 < AT+CMER=3,0,0,1
> >  [AG]   T616 > OK
> >  [AG]   T616 < AT+CLIP=1
> >  [AG]   T616 > OK
> >  [AG]   T616 < AT+CGMI
> >  [AG]   T616 > SONY ERICSSON
> >  [AG]   T616 > OK
> >  [AG]   T616 < AT+CGMI
> >  [AG]   T616 > SONY ERICSSON
> >  [AG]   T616 > OK
> >  [AG]   T616 > +CIEV: 2,4
> >  [AG]   T616 > +CIEV: 2,3
> > asterisk1*CLI>
> >
> > On 1/26/06, Nilesh Londhe <[EMAIL PROTECTED]> wrote:
> > > BTW, I did get clear bidirectional audio when I succeded in dialing
> > > out...(with the channel = 3 in /etc/asterisk/bluetooth.conf) I have
> > > Sony Ericsson T616 connected to a cheap commodity bluetooth USB dongle
> > > that I bought ages ago from meritline.
> > >
> > > On 1/26/06, Nilesh Londhe <[EMAIL PROTECTED]> wrote:
> > > > Thanks a billion.
> > > >
> > > > Outbound bluetooth dialling on the lines of
> > > > Dial(BLT/DevName/8005551212) worked for me.
> > > >
> > > > Still trying out the inbound route. Before I created the [bluetooth]
> > > > context, it tried to reach the [default] context but then I began by
> > > > creating a new context [bluetooth] in extensions.conf and got my
> > > > internal SIP phone to ring when I received a call on my SE T616 cell
> > > > phone. However, I could not get the inbound line answered and I will
> > > > continue to work on this over the weekend and report back my progress.
> > > >
> > > > On 1/25/06, Joseph Tanner <[EMAIL PROTECTED]> wrote:
> > > > > Again, my documentation is still sparse.  I should have noted that the
> > > > > phone will recognize asterisk and connect even if the channel in
> > > > > bluetooth.conf is configured wrong.  You'll just get no audio, or
> > > > > disconnects, or what-not until it's set correctly.  So realize that
> > > > > later on when you're testing.  Also the usb dongle must have a CSR
> > > > > chipset, else it won't work (well, at least probably won't work, I'll
> > > > > provide instructions on how to tell if it should work or not later).
> > > > >
> > > > > Here's the relevant instructions on
> > > > > http://www.crazygreek.co.uk/content/chan_bluetooth for how

RE: [Asterisk-Users] Re: Polycom 501 horrible echo

[Chad Osmond]

I have had no problems running the Sip.cfg from 1.5.2 with 1.6.4 so far,
but I am looking to update in the next while.

Chad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: January 27, 2006 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Polycom 501 horrible echo

Hi - 

>>> I'm running 1.6.2.0041 according to my phone.
>>> 
>>> Which firmware "worked" for you?
>> 
>> It was the old firmware from when we first got the phones actually.
>> 1.4.x I think. Then I read that they fixed the CID issue and decided 
>> we needed an upgrade. I tried it out on my phone, but didn't really 
>> notice the problem until we had upgraded the rest. Oh well...
>> 
> 
> Also, these are IP 500 SIP.

We've been using Polycom phones since firmware version 1.3.0, and I've
used every version of the firmware since then in production on IP300s,
IP500s, IP501s, IP600s, IP601s and IP4000s.  I've never had this issue
on any of them.  I don't mean to downplay the issue, but it may be
possible that you did, in fact, get a bad batch of phones.  When I've
ordered these phones in quantity before, I've gotten many phones with
consecutive serial/mac addresses, so they were probably manufactured in
a bunch.  Maybe a bad batch of mics got installed on a group of phones?

One thing I was pondering: you are not, by chance, using the same
sip.cfg between version 1.4.1 and version 1.6.2 are you?  The file has
changed significantly between these versions, and certain acoustic
settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention
that ipmid.cfg and sip.cfg were merged in the 1.5.x release).


- Noah

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Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS

[Roderick A. Anderson]

Eric Bishop wrote:
Do you have step by step instructions on how you created these RPMs. I 
would like to create a few of my own but compiled for my own custom 
kernel and patchea and am not very familiar with RPM packaging


A good starting point is to download and install the source RPMs in:

ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0/asterisk-1.2.3/SRPMS

Install them and then tweak the spec file in '/usr/src/redhat/SPECS/' 
and do a rpmbuild on it.



Rod
--

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RE: [Asterisk-Users] Re: Random Disconnects

[Jean-François Rousseau]

Hi, we have the same problem here at 2 location that we just installed

Asterisk 1.2.1
P4 3.0Ghz
Motherboard ASUS P4S800-VM
2 SATA disk in software Raid-1

We use 2 nic, one (onboard) to talk to the network (1Gbps link that we use à
100Mbps) and the other realtek 8139 from  Startek that talk to the sipura on
a separate subnet.

Up to now I've tried going back to asterisk 1.0.9 with no success
Tried V2xx and V3xx of the sipura without success

Have you found something ?

Thanks in advance 


___
Jean-François Rousseau
www.sys-tech.net
[EMAIL PROTECTED]
Tél. 24h (418) 520-0739Télec. (418) 520-4554
1-877-969-tech
Ouverture Technologique

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Thczv F.
Thczv
Envoyé : 26 janvier 2006 14:12
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Re: Random Disconnects

On 1/26/06, C F <[EMAIL PROTECTED]> wrote:

> OK, some update on this. It's not related to the Sipuras (actualy the 
> sipuras are very good at this, since they will re-ring your call). I 
> changed my setup to a mediatrix 1204 and I still have the problem.
> Right now I'm looking at:
> 1. Changing the NIC.
> 2. Changing the machine asterisk is on.
> I will start with one, if that fails, then I'm going with a new 
> machine (such fun:P)
>
> BTW, what NIC are you using? what chipset is it? what module makes it 
> work? and/or what option in the kernle did you compile that loads it?
> A 'dmesg | grep eth' should give you some info.

I believe the NIC in the asterisk machine is a Netgear FA310TX.  I really
didn't do anything manually as part of the compile.  The [EMAIL PROTECTED] CD
took care of that for me (though I stripped out sip.conf and extensions.conf
and configured those myself).

Here is what "dmesg | grep eth" returns:

*
divert: allocating divert_blk for eth0
eth0: Lite-On 82c168 PNIC rev 32 at 0xc48db000, 00:A0:CC:D6:A9:47, IRQ 3.
divert: freeing divert_blk for eth0
divert: allocating divert_blk for eth0
eth0: Lite-On 82c168 PNIC rev 32 at 0xc496f000, 00:A0:CC:D6:A9:47, IRQ 3.
eth0: Setting full-duplex based on MII#1 link partner capability of 45e1.
*

Dave
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RE: [Asterisk-Users] extension to extension dialing

[Nora Lavelle]

Hmm.. I definitely have type=friend in the sip.conf and I added
qualify=yes but, I think that's default anyways.. When I call from the
outside and enter his extension it goes through to him fine but, when I
go extension to extension it automatically goes to voicemail.. Here are
the messages from the console:

-- Executing Macro("SIP/130-58df", "stdexten|SIP/124") in new stack
-- Executing Dial("SIP/130-58df", "SIP/124|20") in new stack
-- Called 124
Jan 27 10:27:10 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
  == No one is available to answer at this time
-- Executing Goto("SIP/130-58df", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail("SIP/130-58df", "u124") in new stack
-- Playing 'voicemail/default/124/greet' (language 'en')
Jan 27 10:27:10 NOTICE[28243]: sched.c:290 ast_sched_del: Attempted to
delete non-existant schedule entry 22838!
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing

In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.

About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..

On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
>
> Hi there gary. thanks so much for your help. we're using sipura-841
and snom 320s.
>
> Here's the sip show peers.. that's weird that extension 130 has port
2057.. could that be the problem ?
>
> -nora
>
> Name/usernameHostDyn Nat ACL Mask Port
Status
>
> 201/201  10.200.0.56  D  255.255.255.255  5060
Unmonitor
> ed
> 130/130  10.200.0.10  D  255.255.255.255  2057
Unmonitor
> ed
> 129/129  10.200.0.5   D  255.255.255.255  5060
Unmonitor
> ed
> 127/127  10.201.0.30  D  255.255.255.255  5060
Unmonitor
> ed
> 126/126  10.201.0.29  D  255.255.255.255  5060
Unmonitor
> ed
> 125/125  10.201.0.35  D  255.255.255.255  5060
Unmonitor
> ed
> 124/124  10.201.0.31  D  255.255.255.255  5060
Unmonitor
> ed
> 102/102  10.200.0.48  D  255.255.255.255  5060
Unmonitor
> ed
>
> -Original Message-
> From: [EMAIL PROTECTED] on behalf of Gary
Richardson
> Sent: Thu 1/26/2006 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's
help
> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo. yay!
Now I'm
> > having an issue with SIP extension to extension calling. Any time I
dial
> > another extension it goes right into voice mail.  My extensions.conf
is
> > pretty small and rough but, here's what I have right now. Most of it
was
> > taken from the voip-info website. Any help as always VERY
appreciated.
> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat extensions.conf
> >
> > [incoming]
> >
> > exten => s,1,Answer();
> >
> > exten => s,2,Background(ssn-greeting);
> >
> > exten => *,1,Directory(default)
> >
> > exten => 205,1,Wait(2)
> >
> > exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> >
> > exten => 205,3,Wait(2)
> >
> > exten => 205,4,Playback(/tmp/asterisk-recording)
> >
> > exten => 205,5,Wait(2)
> >
> > exten => 205,6,Hangup
> >
> >
> >
> > [internal]
> >
> > exten => 101,1,Macro(stdexten,SIP/101)
> >
> > exten => 102,1,Macro(stdexten,SIP/102)
> >
> > exten => 103,1,Macro(stdexten,SIP/103)
> >
> > exten => 123,1,Macro(stdexten,SIP/123)
> >
> > exten => 124,1,Macro(stdexten,SIP/124)
> >
> > exten => 125,1,Macro(stdexten,SIP/125)
> >
> > exten => 126,1,Macro(stdexten,SIP/126)
> >
> > exten => 127,1,Macro(stdexten,SIP/127)
> >
> > exten => 128,1,Macro(stdexten,SIP/128)
> >
> > exten => 129,1,Macro(stdexten,SIP/129)
> >
> > exten => 130,1,Macro(stdexten,SIP/130)
> >
> > exten => 135,1,Macro(stdexten,SIP/135)
> >
> > exten => 117,1,Macro(stdexten,SIP/117)
> >
> > exten => 201,1,Macro(stdexten,SIP/201)
> >
> >
> >
> > [voicemail]
> >
> > exten => 300,1,Answer
> >
> > exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
> >
> > exten => 300,3,Hangup
> >
> >
> >
> > [local]
> >
> > exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _9NX

Re: [Asterisk-Users] AAH out bound routing problem

[ram]

Hi rajeev
 
i have posted the extension.conf before
now iam posting extension_additional.conf
 
[EMAIL PROTECTED] asterisk]# more extensions_additional.conf[globals]#include globals_custom.confVM_PREFIX = *RINGTIMER = 15REGTIME = 7:55-17:05REGDAYS = mon-friRECORDEXTEN = ""
PARKNOTIFY = SIP/200OUT_2 = SIP/easycallOUTPREFIX_2 =OUTMAXCHANS_2 = 1
OUTCID_2 = outside account
OPERATOR =NULL = ""IN_OVERRIDE = forcereghoursINCOMING = group-allFAX_RX_EMAIL = [EMAIL PROTECTED]FAX_RX = systemFAX =DIRECTORY_OPTS =DIRECTORY = last
DIAL_OUT = 9DIAL_OPTIONS = trDIALOUTIDS = 2/CALLFILENAME = ""AFTER_INCOMING =
[ext-local]include => ext-local-customexten => 1000,1,Macro(exten-vm,1000,1000)exten => ${VM_PREFIX}1000,1,Macro(vm,1000)exten => 1000,hint,SIP/1000
[outbound-allroutes]include => outbound-allroutes-custominclude => outrt-001-longdistance
[outrt-001-longdistance]include => outrt-001-longdistance-customexten => _1NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)exten => _1NXXNXX,2,Macro(outisbusy)    ; No available circuitsexten => _NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)
exten => _NXXNXX,2,Macro(outisbusy) ; No available circuitsexten => _NXX,1,Macro(dialout-trunk,2,${EXTEN},)exten => _NXX,2,Macro(outisbusy)    ; No available circuits
 
 
ram  
On 1/27/06, Rajeev Natarajan <[EMAIL PROTECTED]> wrote:
if you are using AAH, please post extensions.conf,extensions_additional.conf - also send us more info on your phones.
thanksrajeevram wrote:> Hi>> all of them thanks for the quick reply>> i was tried adding 9 as well as 00> but i get number invalid if i put any of the digits
>> what kind of config files need to post here to resolve the problem>> please assists>> ram>>> On 1/27/06, *Michael Collins* <
[EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>> Ram, On my AAH the stock dial plan requires a 9 first.  For kicks, try
> dialing 919197543700 and see what you get. -MC >> *From:* 
[EMAIL PROTECTED]> [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED]
> ] *On Behalf Of *ram> *Sent:* Friday, January 27, 2006 6:14 AM> *To:* Asterisk Users Mailing List - Non-Commercial Discussion> *Subject:* [Asterisk-Users] AAH out bound routing problem
 Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and 
http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700
> i get message that, all circuits are busy now, please try your call> later and when i see in the console i get this mesage>
>>> any help Called easycall/19197543700> -- Got SIP response 488 "Not acceptable here" back from (PeerIP)> -- SIP/easycall-838e is circuit-busy
 ram>>> ___> --Bandwidth and Colocation provided by Easynews.com> <
http://easynews.com/> -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>   
http://lists.digium.com/mailman/listinfo/asterisk-users> >> ___
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RE: [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000strange problem

[Jean-François Rousseau]

Hi, I've got the exact same problem here with Asterisk 1.2.1, at 2
locations.

The first one have 8 sipura 3000 with 5 pstn lines  and 8 standard phones.
There is also 4 Mitel 5215 phones

The secon one have 8 sipura 3000 with 5 pstn lines and 8 standard phones.
There is also 8 Mitel 5215 phones

I have the same problem as yours on Mitel or standard phones at the 2
locations.

I also tried with Sipura v2 and v3 firmware... Same problem

It only happen 3 or 4 times a day but it's a big pain in the ass.

For the network config:

At each location all the sipura are on a network of their own directly
connected to the astersik box. All the mitel are on a secon network card.

The asterisk machines are P4 on ASUS Motherboard in software raid1 config.


Does somebody have a solution ?

I've found someone else with the same problem:

http://www.voipuser.org/index.php?name=PNphpBB2&file=viewtopic&t=4207&highli
ght=

And he also have another problem with the sipura that I have too.

http://www.voipuser.org/index.php?name=PNphpBB2&file=viewtopic&t=4193&highli
ght=

Thanks for your help...

___
Jean-François Rousseau
www.sys-tech.net
[EMAIL PROTECTED]
Tél. 24h (418) 520-0739Télec. (418) 520-4554
1-877-969-tech
Ouverture Technologique

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de C F
Envoyé : 28 décembre 2005 15:15
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura
SPA3000strange problem

In any case I'm trying to figure out if maybe someone else has seen this
problem. Or if they know what it might be.

On 12/28/05, C F <[EMAIL PROTECTED]> wrote:
> For somereason I think it's the polycom, which means I need logging 
> for the Polycom and not the spa.
>
> On 12/28/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> > > I have the follwoing setup:
> > > Asterisk  SVN-tag-1.2.1-r7367
> > > 6 Polycom 500 Sip version 1.5.x
> > > 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat 
> > > single network, no NAT, and no gateways to reach each other.
> > > Sometimes (happens around 3 times a day, but sometimes far more 
> > > often), while on the phone to an outside caller (on the PSTN using 
> > > the FXO on the spa3k), the call dissconects from the polycom and 
> > > goes thru the incoming extension for the sipura. In other words, 
> > > astrisk at least as far as I can see from what gets executed in 
> > > the DP (and maybe
> > > spa3k) sees this as if the follwoing has happened: 1. The polycom 
> > > user hungup, 2. A new call came in on the spa3k.
> > > The follwoing is part of the log that I think might help:
> > > Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from
> > > channel: SIP/201-8ba1
> > > Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging 
> > > channels
> > > SIP/201-8ba1 and SIP/804-fd83
> > >
> > > SIP/201 is the Polycom, while SIP/804 is the spa3k.
> > >
> > > If I'm losing a frame, is there a way to configure asterisk not to 
> > > drop the channel? Or is this something the Polycom/Sipura are doing?
> > >
> > > FYI, asterisk is running on a VIA/MPIA platform.
> >
> > Pure guess is that something happened (unknown what) and the error 
> > messages posted above are the result of that, and not the root 
> > cause. Finding the root cause may require you to implement the 
> > "syslog server" and "debug server" options in the spa3k, and compare 
> > those log entries to what * records for log messages during a failure.
> >
> > Implementing the log functions on the spa3k "does" require a reboot. 
> > Their log messages are rather cryptic, but looking at keywords and 
> > timestamps might identify which box(es) are involved with the dropped
calls.
> >
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [Asterisk-Users] AAH out bound routing problem

[Rajeev Natarajan]

if you are using AAH, please post extensions.conf, 
extensions_additional.conf - also send us more info on your phones.


thanks
rajeev


ram wrote:

Hi
 
all of them thanks for the quick reply
 
i was tried adding 9 as well as 00

but i get number invalid if i put any of the digits
 
what kind of config files need to post here to resolve the problem
 
please assists
 
ram


 
On 1/27/06, *Michael Collins* <[EMAIL PROTECTED] 
> wrote:


Ram,

 


On my AAH the stock dial plan requires a 9 first.  For kicks, try
dialing 919197543700 and see what you get.

 


-MC

 




*From:* [EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]
 ] *On Behalf Of *ram
*Sent:* Friday, January 27, 2006 6:14 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] AAH out bound routing problem

 


Hi all

 


I have installed AAH 2.2 in my P4 PC

 


following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp

 


and made as per the guide says

 


and downloaded SJ Phone, and registered user

 


and when i try to dial the 19197543700
 

 


i get message that, all circuits are busy now, please try your call
later

 


and when i see in the console i get this mesage

 


any help

 


Called easycall/19197543700
-- Got SIP response 488 "Not acceptable here" back from (PeerIP)
-- SIP/easycall-838e is circuit-busy

 


ram


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[Asterisk-Users] Re: Polycom 501 horrible echo

[Noah Miller]

Hi - 

>>> I'm running 1.6.2.0041 according to my phone.
>>> 
>>> Which firmware "worked" for you?
>> 
>> It was the old firmware from when we first got the phones actually.
>> 1.4.x I think. Then I read that they fixed the CID issue and decided
>> we needed an upgrade. I tried it out on my phone, but didn't really
>> notice the problem until we had upgraded the rest. Oh well...
>> 
> 
> Also, these are IP 500 SIP.

We've been using Polycom phones since firmware version 1.3.0, and I've used
every version of the firmware since then in production on IP300s, IP500s,
IP501s, IP600s, IP601s and IP4000s.  I've never had this issue on any of
them.  I don't mean to downplay the issue, but it may be possible that you
did, in fact, get a bad batch of phones.  When I've ordered these phones in
quantity before, I've gotten many phones with consecutive serial/mac
addresses, so they were probably manufactured in a bunch.  Maybe a bad batch
of mics got installed on a group of phones?

One thing I was pondering: you are not, by chance, using the same sip.cfg
between version 1.4.1 and version 1.6.2 are you?  The file has changed
significantly between these versions, and certain acoustic settings that
worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and
sip.cfg were merged in the 1.5.x release).


- Noah

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RE: [Asterisk-Users] AAH out bound routing problem

[Michael Collins]

Start with extensions.conf and also the
debug lines from the Asterisk console.

-MC

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Friday, January 27, 2006
9:54 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AAH
out bound routing problem



 



Hi





 





all of them thanks for the quick reply





 





i was tried adding 9 as well as 00





but i get number invalid if i put any of the digits





 





what kind of config files need to post here to resolve the problem





 





please assists





 





ram

 





On 1/27/06, Michael
Collins <[EMAIL PROTECTED]>
wrote: 



Ram,

 

On my AAH the stock dial plan requires a 9 first.  For
kicks, try dialing 919197543700 and see what you get. 

 

-MC

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
] On Behalf Of ram
Sent: Friday, January 27, 2006
6:14 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion 
Subject: [Asterisk-Users] AAH out
bound routing problem



 



Hi all





 





I have
installed AAH 2.2 in my P4 PC





 





following
AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp





 





and made
as per the guide says





 





and
downloaded SJ Phone, and registered user





 





and when
i try to dial the 19197543700
 





 





i get
message that, all circuits are busy now, please try your call later





 





and when
i see in the console i get this mesage





 





any help





 





Called
easycall/19197543700
    -- Got SIP response 488 "Not acceptable here" back
from (PeerIP)
    -- SIP/easycall-838e is circuit-busy 





 





ram






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Re: [Asterisk-Users] AAH out bound routing problem

[ram]

Hi
 
all of them thanks for the quick reply
 
i was tried adding 9 as well as 00
but i get number invalid if i put any of the digits
 
what kind of config files need to post here to resolve the problem
 
please assists
 
ram 
On 1/27/06, Michael Collins <[EMAIL PROTECTED]> wrote:


Ram,
 
On my AAH the stock dial plan requires a 9 first.  For kicks, try dialing 919197543700 and see what you get.

 
-MC
 




From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of ramSent: Friday, January 27, 2006 6:14 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] AAH out bound routing problem
 

Hi all

 

I have installed AAH 2.2 in my P4 PC

 

following AAH handbook PDF and 
http://mundy.org/blog/index.php?p=62#amp

 

and made as per the guide says

 

and downloaded SJ Phone, and registered user

 

and when i try to dial the 19197543700 

 

i get message that, all circuits are busy now, please try your call later

 

and when i see in the console i get this mesage

 

any help

 

Called easycall/19197543700    -- Got SIP response 488 "Not acceptable here" back from (PeerIP)    -- SIP/easycall-838e is circuit-busy


 

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[Asterisk-Users] Re: Good provider of Polycom Phones (mostly for access to latest/greatest firmware)

[Noah Miller]

Hi Gavin - 

> I've ordered a few IP501s from PC Connection, basically since we have an
> account with them. I like the phones for what they do, and now would like
> establish a relationship with a reseller that can give us maintenance and
> access to the most current firmware.
> 
> What are some good resellers out there?

I love PC Connection for most of my ordering, but I've actually never used
them for Polycom hardware, even though we have a lot of it.

My Polycom supplier has been http://www.tritechcoa.com

They are extremely responsive, and have the best prices around on Polycom.
When I recently asked for the latest firmware, I got a response back in
about 2 minutes.  Now I actually have firmware newer than what is listed on
the Polycom website.  I haven't tested it out yet, though.

- Noah







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