[asterisk-users] R: GXP2000 and asterisk 1.0.9

[Giordano Grandis]

1. The phone has not the DND active, i checked it several times
2. Outbound calls always success, the problem is when the phone receive a call, 
it repsnds with busy signalling.
3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade asterisk.

Thanks for all

-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di C F
Inviato: mercoledì 13 febbraio 2008 21.09
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

Just check DND if it's on on the phone or not.
What is the CLI output when you try making a phone call?
Why don't you try it with a later version of astrisk and a Phone?

On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>
>
> Hi all gusy,
> i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few
> go in "busy" state, if you call it get the busy tone but the phone can male
> any type of call.
> This is my sip.conf
>
> [502]
> language = it
> username = 502
> secret = 
> host = dynamic
> type = friend
> context = local
> canreinvite = yes
> dtmfmode = info
> callgroup = 1
> pickupgroup = 1
> callerid = 502 <502>
>
> Under Grandstream's support suggest, I set "Use randmom port" to yes and
> "Nat traversal (STUN)" to "No, but send keep alive" but without success.
> This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6
>
> Anyone can help me ?
>
> Thanks in advance
>
> Giordano
>
>
> No virus found in this outgoing message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008
> 15.20
>
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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 
15.20
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 
20.00
 

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[asterisk-users] R: GXP2000 and asterisk 1.0.9

[Giordano Grandis]

Thanks Henry,
anyway the phone is always registered when i get the busy tone

  * Name   : 502
  Secret   : 
  MD5Secret: 
  Context  : local
  Language : it
  FromUser :
  FromDomain   :
  Callgroup: 1 (2)
  Pickupgroup  : 1 (2)
  Mailbox  :
  LastMsgsSent : -1
  Dynamic  : Yes
  Expire   : 703 seconds
  Expiry   : 900
  Insecure : No
  Nat  : No
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  DTMFmode : info
  LastMsg  : 0
  ToHost   :
  Addr->IP : 192.168.13.171 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Username : 502
  Codecs   : 0x8010f (g723|gsm|ulaw|alaw|g729|h263)
  Codec Order  : (alaw|ulaw|gsm|g729|g723)
  Status   : OK (22 ms)
  Useragent: Grandstream GXP2000 1.1.5.15
  Full Contact : sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone

Any idea?

Thanks again to all


-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Henry Devito
Inviato: mercoledì 13 febbraio 2008 22.01
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

Is your phone actually registered to the switch.  go to the CLI and do a 
'sip show peers'  see if extension 502 is registered.  Making an outbound 
call does not prove that the phone is registered.


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

Sent: Wednesday, February 13, 2008 2:09 PM
Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9


> Just check DND if it's on on the phone or not.
> What is the CLI output when you try making a phone call?
> Why don't you try it with a later version of astrisk and a Phone?
>
> On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>>
>>
>> Hi all gusy,
>> i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a 
>> few
>> go in "busy" state, if you call it get the busy tone but the phone can 
>> male
>> any type of call.
>> This is my sip.conf
>>
>> [502]
>> language = it
>> username = 502
>> secret = 
>> host = dynamic
>> type = friend
>> context = local
>> canreinvite = yes
>> dtmfmode = info
>> callgroup = 1
>> pickupgroup = 1
>> callerid = 502 <502>
>>
>> Under Grandstream's support suggest, I set "Use randmom port" to yes and
>> "Nat traversal (STUN)" to "No, but send keep alive" but without success.
>> This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6
>>
>> Anyone can help me ?
>>
>> Thanks in advance
>>
>> Giordano
>>
>>
>> No virus found in this outgoing message.
>>  Checked by AVG Free Edition.
>>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 
>> 12/02/2008
>> 15.20
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 
15.20
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 
20.00
 

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Re: [asterisk-users] Analog DID

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

darren wrote:
>  An analog DID trunk is a line (typically part of a group) that has a group 
> of 
> numbers assigned to it at the telco side.  They work in a variety of ways 
> depending on the telco.  One example is the trunks as Telus provides them.  
> The 
> end user provides dialtone back to the telco.  When a call comes in on a DID 
> the 
> telco picks up the first available line (remember, the customer is providing 
> dial tone.) and dials the last 4 digits of the dialed number.  They are often 
> replaced by PRIs but in some locations a PRI is not affordable and these 
> provide 
> the same DID functionality for a small fraction of the price.

We've done installs on the same here.  Basically you set up FXS
connections and the telco picks them up as if they were a telephone, the
extensions get dialled and it kinda just works.  Only really useful
where you can't get a BRI here in New Zealand.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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1V/DGRiH61DV2IWqVZU5MXU=
=v6U7
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Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of

[Phil Knighton]

In the UK, to make us match the rest of Europe, it's also possible to
access the Emergency Services on 112.
 
Again, although a few calls were made around the right time, none of
them were 999 or 112.  The I've examined the master.csv for 30 mins
before the Police said the call was made, and can't see any possible way
the number was dialled.
 
We do have a "special" entry in our call plan to deal with emergency
calls, which was taken from voip-info, that intercepts an internal
extension of "999", and sends it out via a ZAP channel.  We even have it
setup so if it fails to find a free ZAP channel, it will end the call on
channel 1 and then dial.  Again though, this is only for a calls
specifically made to "999".
 
I'm baffled...



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Langstaff
Sent: 13 February 2008 15:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort
of


It might be possible to get to the emergency service via 112 or a local
number as well.
 
Do you have *any* calls placed at about the time of the "999" calls?
 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Knighton
Sent: 13 February 2008 14:12



Hello
 
This is a fun one for the list...
 
Twice now, the Police have contacted us to say they have had a
silent call then hangup from our landline number to the 999 service.  As
a matter of course, they follow up these calls in case someone is in
distress.  Nobody here was in distress - well, no more than normal!  The
Police aren't hugely happy when we tell them it must be a mistake.
 
Thing is, I have checked both our master log, and our dialled
calls log - and nobody dialled 999!  Each phone has an account code
applied from sip.conf, and we log all numbers dialled.  Nobody dialled
out.
 
There are no phones connected in anyway other than via Asterisk,
fax number is dealt with by a virtual machine, alarm system is on a
different number...
 
Any ideas before the rossers come and take me away?
 
Phil
  

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Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of

[Phil Knighton]

Hi Tilghman

As far as I can see from both master.csv and the account log, no number
was dialled beginning 999 (or 112 - both numbers connect to the
Emergency Services, and the Police couldn't tell me which had been
called).

Unfortunately, my Telco (British Telecomsigh) can't tell me exactly
when it was called, which line it was on or anything else!  All I have
is the information the Police gave me.

Is it possible for Asterisk (for whatever reason) to pass a call that
isn't logged in master.csv?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 13 February 2008 15:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort
of

On Wednesday 13 February 2008 08:12:25 Phil Knighton wrote:
> Thing is, I have checked both our master log, and our dialled calls 
> log
> - and nobody dialled 999!  Each phone has an account code applied from

> sip.conf, and we log all numbers dialled.  Nobody dialled out.

Have you checked all numbers that might have a PREFIX of 999?  Here in
the States, occasionally a prankster will tell someone annoying her to
call her on her cell phone at 911-5924 or something like that, and of
course, the system only sees the "911" portion, not the additional 4
digits, which connects them to the emergency number on this side of the
Pond.

--
Tilghman

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Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of

[Phil Knighton]

Can I just say I'm grateful for all the replies - this list is
invaluable.
 
Thanks for the suggestion Razza, I've been back again to the logs and no
call was placed that contained the string "999" or "112" at the right
time!
 
Glad it made you smile, said it was a fun one for the list.  Looking
like this one is going to go down as a mystery... until the Police call
again!  It has only happened twice in the 16 months we've been using
Asterisk, I'm just keen to find a solution.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Razza
Sent: 13 February 2008 21:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort
of


When I first set up asterisk, I had similar, fortunately not with the
old bill!
It basically boiled down to not enough delay between seizing the line
and dialing the digits, for example the following would have dialled the
coppers 01299 912345, as 012 would have been missed.
I'm guessing this isn't whats happening to you, if all your other calls
are uworking fine, but did bring back some memories and made me smile
:o)
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Re: [asterisk-users] multiple host in 1 context on sip.conf

[Johansson Olle E]

Hi Mark!

13 feb 2008 kl. 23.42 skrev Mark Quitoriano:

> Is it possilble for a single context to have multiple host=  
> something like this
First "context" is something we use to describe a segment of the  
dialplan. I would call this "section".


>
>
> [carrier]
> host=
> host=
> host=
> type=peer
> disallow=all
> allow=g729
> allow=ulaw
> canreinvite=no
> insecure=yes
> qualify=yes

No. You can only add one.

Normally I would add host=sip.mydomain.com and have multiple DNS  
entries or use SRV records to do failover and such,
provided you use this for outbound calls.

Since you call this peer "carrier" I assume you want to handle inbound  
calls. Today, you will have to define three different
peers, but remember that you can use templates.

[carrier](!)
type=peer
disallow=all
allow=g729
allow=ulaw
canreinvite=no
insecure=yes
qualify=yes

[carrier-01](carrier)
host=

[carrier-02](carrier)
host=

[carrier-03](carrier)
host=

You will now have three peers named carrier-01-03 but no peer named  
"carrier" in your sip driver when you run "sip show peers".

Regards,
/Olle


---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/




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Re: [asterisk-users] Grandstream GXP2000 Loses Connectivity

[Andreas van dem Helge]

I've had the opposite problem. Press mute while the call is still
ringing and it will say "MUTE" on the display but the microphone is
not muted. It was very embarrassing to discover this bug.

On Wed, Feb 13, 2008 at 2:03 AM, Thomas Kenyon
<[EMAIL PROTECTED]> wrote:
> Lutgring, Sam wrote:
>  I take it you've also not had the problem where the handset microphone
>  stops working. (This is apparently already fixed and will be available
>  in the next beta firmware release (1.1.6.x), when they've fixed some
>  more problems that have been very difficult to track down.)
>
>  I'd like to go back to 1.1.5.15 if nothing more than for the improved
>  audio quality and the on-hook dialling.
>

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Re: [asterisk-users] UK -999 dialing issue

[Phil Knighton]

Hi Amit
 
OK, the majority of our calls go out via zaptel fxo and pstn lines.
When these are all busy, calls are routed via a VOIP provider here in
the UK.  All activity is recorded in our logs, and I can find no trace
of either 999 or 112 (if since been reminded that in the UK, you can now
also use 112 which is consistent with continental Europe).
 
I can't find a call placed at the relevant time that had these numbers,
even as mid-part of a string.
 
Below is the part which deals with our external calls.  As you can see,
calls are routed out via zap, or VOIP (that's the gradwell bit).  If
someone prefixes a call with "9" it forces it our via VOIP and if
someone dials "999" it is intercepted and sent via the zap channels. If
no zap channel is free, a call on channel 1 is ended and the number
re-dialled.  This makes sure that emergency calls can always be placed
on a landline.

Any ideas would be appreciated!

Phil
 
[softoption-zap]

exten => _0[123456789].,1,NoOp(${EXTEN})
exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)

; The below section will allow for 3 digit BT numbers to be called, by
prefixing them with 9
; For example: 154 is BT Business Faults - dial 9154

exten => _9[123456789]XX,1,NoOp(${EXTEN})
exten => _9[123456789]XX,2,Dial(Zap/g0/${EXTEN:1},,j)

; The below section will allow for 999 Emergency calls to be made. This
will FORCE these calls
; over our BT lines, which will provide CallerID and location
information to the Emergency Operator
; If there are no BT lines free, it will force a call to end and then
dial

exten => 999,1,NoOp(999)
exten => 999,2,Dial(Zap/g0/999,,j)
exten => 999,3,Hangup()
exten => 999,102,SoftHangup(Zap/1-1)
exten => 999,103,Wait(1)
exten => 999,104,Goto(1)

[softoption-gradwell]
exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten => _0[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten => _[1-9]X,1,Dial(IAX2/Gradwell/441353${EXTEN},,)




From: amit salunkhe [mailto:[EMAIL PROTECTED] 
Sent: 14 February 2008 07:44
To: Phil Knighton
Subject: UK -999 dialing issue


HI Phil
Can u send me ur out call context config. Also tell me what
ur using with Asterisk to make out call SIp-Voip or Pstn line with Fxo
card?
also check with this command in ur Asterisk console. sip
show peers so u  can get anybody from out side place such call inbehalf
of u. check who & how many user regsiter with ur Asterisk. if ur using
FXO card then also there is chance to check this.
   also use Mysql for CDR table tocheck who try to call at time. so
u got any hint for this
 
Regards
Amit

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Re: [asterisk-users] PCI32 and PCI-X compatibility

[Marco]

Thanks Michael,
that's a *huge* thing you're telling me, in the wiki details for the 
PCI-X bus I've read about retrocompatibility, but I just wanted to be 
100% sure. I can go on and order my server, now!

Thanks again

Marco

ps. This proves also the complete unaccuracy of the information provided 
by the local Digium distributor - italian people, be aware!


Michael Spiceland ha scritto

Marco,

You should not have any issues using a PCI card in a PCI-X slot, as
long as the card is a 3.3V PCI card.  The cards that you mention above
are 3.3v compatible and you should be able to use them.

All of Digium's product line is available for 3.3v slots.  Most are
universal and can be used in 3.3v or 5v slots.  The only exceptions
are the dual and quad span T1/E1 digital cards.  For those cards,
there are 3.3v variants (TE410P and TE210P) and 5v variants (TE405P
and TE410P).



Oops, I meant that the 5v variants are the TE405P and *TE205P*.

3.3v -> TE410P and TE210P
5v   -> TE405P and TE205P

Michael

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Re: [asterisk-users] UK -999 dialing issue

[Gordon Henderson]

On Thu, 14 Feb 2008, Phil Knighton wrote:

> [softoption-zap]
>
> exten => _0[123456789].,1,NoOp(${EXTEN})
> exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)

OMG!!!

You're selecting 2 different output channels depending on the number 
dialled!!!

(UK or international)...

That's ... LCR!!!

In  ... Dialplan!!!

And according to a recent thread, that's like ... impossible, not 
recommended, really really hard, with databases and external hardware 
required, etc. (!!!)

(sorry)

Gordon
(dialplan junkie)

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Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP

[Steve Langstaff]

The "481 Call Leg/Transaction Does Not Exist" response to the
NOTIFY makes me think that you might need to configure the
phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
from the phone when it is booted?


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jaap Winius
> Sent: 13 February 2008 18:46
> 
> Hi list,
> 
> Before purchasing a number of Siemens DECT SIP phones, the Gigaset
> S675 IP, I read that the problems with MWI had been fixed 
> with the latest firmware version (see 
> http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). 
> Now I'm not so sure that's the case.
> 
> After setting up a network mailbox for one of these phones, 
> as well as an Asterisk voicemail account (ext. 1000) in 
> voicemail.conf's default context, I added the following line 
> to my phone's context in sip.conf:
> 
>mailbox=1000
> 
> However, soon after executing a 'sip reload' on the console, 
> the following error message will appear every three minutes:
> 
>[Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response:
> Remote host can't match request NOTIFY to call
>'[EMAIL PROTECTED]'. Giving up.
> 
> The IP address belongs to my server. At the same time, I used 
> tcpdump to see what else might be going on. I found the following:
> 
>19:18:22.540113 IP bitis.umrk.to.sip > 
> gigaset.umrk.to.sip: SIP, length: 545
>[EMAIL PROTECTED]
>.)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
>Via: SIP/2.0
>19:18:22.571452 IP gigaset.umrk.to.sip > 
> bitis.umrk.to.sip: SIP, length: 308
>E..P...f...
>.Via:
> 
> The latest comment on the voip-info.org page above outlines 
> the same problem. Can anyone here confirm that this is indeed 
> a Siemens problem, or might it be an Asterisk problem after all?
> 
> I'm running Asterisk v1.4.14 on a Debian etch server.

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[asterisk-users] Ser, asterisk and ip2ipgw

[Riccardo Cupardo]

Hi,

i use a ser, as proxy sip server(authentication), then a cisco router
as sip2h323 gw(authorization and accounting). i want to start asterisk
as sip statefull b2bua server, any suggestion to howto or documentation
to asterisk integration and b2b use?

ty in advance.

-- 
Riccardo Cupardo






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Re: [asterisk-users] UK -999 dialing issue

[Steve Langstaff]

> [softoption-zap]
> 
> exten => _0[123456789].,1,NoOp(${EXTEN})
> exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)

Just an aside -

1) For clarity, could you use 'Z' here instead of '[123456789]'?

2) It does not look like you would be able to dial numbers that
start with 0[123456789] and then have subsequent zeros
(e.g. 01xx xxx ) - is that your intent? 


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[asterisk-users] Error checking asterisk method - suggestions?

[Johan Sandgren]

Hi there dear users and dear developers of Asterisk!


I've got a maybe strange idea, let's see if you laugh or think it's reasonable J

I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering 
machine).
Each analog line can be reached through a phonenumber, since they are each 
connected to my telephone provider. Yes expensive I know J


The challenge:

I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, 
and 3) my operator is alive and working, and I have an Idea which I wonder if 
will/could work.

My first idea was to ask the zap-driver if it could detect if the line was ok, 
but no function existed to do that, what I could find. Anyone knows about one?

My second idea, was to try calling simply, to know if things were ok. But, I 
couldn't just call any number, I had to know the number was in use, and not 
disturbing anyone.
So, I called myself, or I called another of my phonelines.

So,
I'd like to use the asterisk manager interface in java to originate a call from 
one ZAP-channel, calling out to my telephone provider,
And then they will direct the call back to my, but into another ZAP-channel 
(since I'm calling that channel's number).

So: I'm making ZAP/1 calling out to no 323121321 -> telephone company, Ok: 
323121321 belongs to this guy -> redirecting me to my ZAP/2 channel, which 
answers the call.

Then I have a connection, and ZAP/2 will answer and do some DTMF.
My first ZAP/1 is run through my java program, and I'd like to listen for 
certain DTMF-tones, to know I have a working circuit.

The goal for all of this, is to verify things are working, so my provider is 
not down, or one of my ZAP-lines are dead.

So far, I've tried calling and got some half-success, but not sure what is 
going on doing all the right way.
For ex: why am I calling with Zap1, to Zap 3, and then Zap 7 is answering? 3 
channels used for one outgoing and one incoming call? Something must be very 
wrong J
Please educate me, dear experts.

Input?


Sincerely,
Johan Sandgren
www.svep.se, [EMAIL PROTECTED]
Frosty Sweden but with some sunshine today !! J

My code and settings below, for information.

=JAVA CODE (extract)

OriginateAction originateAction = new OriginateAction();
ManagerResponse originateResponse = null;
originateAction.setChannel("ZAP/1");
originateAction.setContext("Outgoing");
originateAction.setExten("201"); // maps to 
ZAP/7 through external phonecompany PBX
originateAction.setPriority(new Integer(1));
originateAction.setTimeout(new Long(15*1000));  
 // xml-milliseconds
originateAction.setAsync(false);

== extensions.conf (extract)

[Incoming]
exten => s,1,Set(DYNAMIC_FEATURES=hangup)
exten => s,2,Agi(agi://localhost/answer.agi)

[Outgoing]
exten => _X.,1,Set(DYNAMIC_FEATURES=hangupfeature)
exten => _X.,n,Dial(Zap/3/${EXTEN}

==  Asterisk response:

AGI Debugging Enabled
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'stt' logged on from 127.0.0.1
   > Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] Set("Zap/1-1", 
"DYNAMIC_FEATURES=hangupfeature") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("Zap/1-1", "Zap/3/201") in new stack
-- Called 3/201
-- Starting simple switch on 'Zap/7-1'
-- Zap/3-1 answered Zap/1-1
[Feb 14 13:02:33] WARNING[26260]: chan_zap.c:6499 ss_thread: CallerID returned 
with error on channel 'Zap/7-1'
-- Executing [EMAIL PROTECTED]:1] Set("Zap/7-1", "DYNAMIC_FEATURES=hangup") 
in new stack
-- Executing [EMAIL PROTECTED]:2] AGI("Zap/7-1", 
"agi://localhost/answer.agi") in new stack
AGI Tx >> agi_network: yes
AGI Tx >> agi_network_script: answer.agi
AGI Tx >> agi_request: agi://localhost/answer.agi
AGI Tx >> agi_channel: Zap/7-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1202990552.2
AGI Tx >> agi_callerid: unknown
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: Incoming
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << ANSWER
AGI Tx >> 200 result=0
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=6
AGI Rx << WAIT FOR DIGIT 1
  == Manager 'testmanager' logged off from 127.0.0.1
AGI Tx >> 200 result=0
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=6
AGI Rx << WAIT FOR DIGIT 1
AGI Tx >> 200 result=0
  == Spawn extension (Incoming, s, 2) exited non-zero on 'Zap/7-1'
-- Hungup 'Zap/7-1'

= Asterisk log

__
Johan Sandgren
Svep Design Center AB (www.svep.se)
St. Lars väg 42A, SE-222 70 Lund
Phone: 046-19 27 22

___

Re: [asterisk-users] Error checking asterisk method - suggestions?

[Tzafrir Cohen]

On Thu, Feb 14, 2008 at 01:17:45PM +0100, Johan Sandgren wrote:
> Hi there dear users and dear developers of Asterisk!
> 
> 
> I've got a maybe strange idea, let's see if you laugh or think it's 
> reasonable J
> 
> I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering 
> machine).
> Each analog line can be reached through a phonenumber, since they are each 
> connected to my telephone provider. Yes expensive I know J
> 
> 
> The challenge:
> 
> I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, 
> and 3) my operator is alive and working, and I have an Idea which I wonder if 
> will/could work.
> 
> My first idea was to ask the zap-driver if it could detect if the line 
> was ok, but no function existed to do that, what I could find. Anyone 
> knows about one?

What do you mean by "line is OK"?

> 
> My second idea, was to try calling simply, to know if things were ok. 
> But, I couldn't just call any number, I had to know the number was in 
> use, and not disturbing anyone.
> So, I called myself, or I called another of my phonelines.

And you assume noone else calls in at the time?

> 
> So,
> I'd like to use the asterisk manager interface in java to originate a 
> call from one ZAP-channel, calling out to my telephone provider,
> And then they will direct the call back to my, but into another 
> ZAP-channel (since I'm calling that channel's number).

For a basic test that the line works, try TestClient and TestServer .
Originate a call from testclient (and set there the number. And set all
incoming calls temporarily to go to TestServer (did I mention the
assumption that noone calls in?)

One relatively cheap method of "temporary" is through setting a global
variable to a non-standard value. This means that the non-default value
will never last after a reload. And you can set the global even through
'core set global VARNAME VALUE' in the CLI.

Check the resulting reports in /var/lib/asterisk/testresults . Make sure
all of them were generated, and that none "FAIL"-ed.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9

[Lutgring, Sam]

Try switching your DTMF mode on both asterisk and the phone to RFC2833.  I have 
not seen the issue that you are describing, but I had some very strange issues 
like call hang-ups when I was using INFO.  After switching the issues were gone 
and I have had no further troubles.

Hope this helps you.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Thursday, February 14, 2008 3:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9

Thanks Henry,
anyway the phone is always registered when i get the busy tone

  * Name   : 502
  Secret   : 
  MD5Secret: 
  Context  : local
  Language : it
  FromUser :
  FromDomain   :
  Callgroup: 1 (2)
  Pickupgroup  : 1 (2)
  Mailbox  :
  LastMsgsSent : -1
  Dynamic  : Yes
  Expire   : 703 seconds
  Expiry   : 900
  Insecure : No
  Nat  : No
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  DTMFmode : info
  LastMsg  : 0
  ToHost   :
  Addr->IP : 192.168.13.171 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Username : 502
  Codecs   : 0x8010f (g723|gsm|ulaw|alaw|g729|h263)
  Codec Order  : (alaw|ulaw|gsm|g729|g723)
  Status   : OK (22 ms)
  Useragent: Grandstream GXP2000 1.1.5.15
  Full Contact : sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone

Any idea?

Thanks again to all


-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Henry Devito
Inviato: mercoledì 13 febbraio 2008 22.01
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

Is your phone actually registered to the switch.  go to the CLI and do a 
'sip show peers'  see if extension 502 is registered.  Making an outbound 
call does not prove that the phone is registered.


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

Sent: Wednesday, February 13, 2008 2:09 PM
Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9


> Just check DND if it's on on the phone or not.
> What is the CLI output when you try making a phone call?
> Why don't you try it with a later version of astrisk and a Phone?
>
> On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>>
>>
>> Hi all gusy,
>> i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a 
>> few
>> go in "busy" state, if you call it get the busy tone but the phone can 
>> male
>> any type of call.
>> This is my sip.conf
>>
>> [502]
>> language = it
>> username = 502
>> secret = 
>> host = dynamic
>> type = friend
>> context = local
>> canreinvite = yes
>> dtmfmode = info
>> callgroup = 1
>> pickupgroup = 1
>> callerid = 502 <502>
>>
>> Under Grandstream's support suggest, I set "Use randmom port" to yes and
>> "Nat traversal (STUN)" to "No, but send keep alive" but without success.
>> This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6
>>
>> Anyone can help me ?
>>
>> Thanks in advance
>>
>> Giordano
>>
>>
>> No virus found in this outgoing message.
>>  Checked by AVG Free Edition.
>>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 
>> 12/02/2008
>> 15.20
>>
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>>
>
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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 
15.20
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 
20.00
 

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Re: [asterisk-users] UK -999 dialing issue

[Thomas Kenyon]

Steve Langstaff wrote:
>> [softoption-zap]
>>
>> exten => _0[123456789].,1,NoOp(${EXTEN})
>> exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
>> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
>> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
>> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
> 
> Just an aside -
> 
> 1) For clarity, could you use 'Z' here instead of '[123456789]'?
> 
> 2) It does not look like you would be able to dial numbers that
> start with 0[123456789] and then have subsequent zeros
> (e.g. 01xx xxx ) - is that your intent? 
> 
Very good point, he probably wants.
exten => _90ZX.,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)

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Re: [asterisk-users] [SPAM] - Re: Error checking asterisk method - suggestions? - Email found in subject

[Johan Sandgren]

>> Hi there dear users and dear developers of Asterisk!
>>
>>
>> I've got a maybe strange idea, let's see if you laugh or think it's 
>> reasonable J
>>
>> I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering 
>> machine).
>> Each analog line can be reached through a phonenumber, since they are each 
>> connected to my telephone provider. Yes expensive I know J
>>
>>
>> The challenge:
>>
>> I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, 
>> and 3) my operator is alive and working, and I have an Idea which I wonder 
>> if will/could >work.
>>
>> My first idea was to ask the zap-driver if it could detect if the line
>> was ok, but no function existed to do that, what I could find. Anyone
>> knows about one?

>What do you mean by "line is OK"?

I mean, (being not so educated in the telephone technology), but to know that 
there is DC voltage connected to my ZAP-channel. (indicating status = OK)
According to Wiki, "A calling party wishing to speak to another telephone will 
pick up the handset, thus operating the switch hook, which puts the telephone 
into active state or off hook with a resistance short across the wires, causing 
current to flow."
So, I suppose to know there is current flowing, would say I'm connected, but 
probably not guarantee that I can make calls. So this test would not give me 
trustworthy results. Or what do you say Tzafrir Cohen? (or others :)



>>
>> My second idea, was to try calling simply, to know if things were ok.
>> But, I couldn't just call any number, I had to know the number was in
>> use, and not disturbing anyone.
>> So, I called myself, or I called another of my phonelines.

>And you assume noone else calls in at the time?

Yes, since I only test lines, that haven't had any incoming calls for some 
time. Sure, someone COULD be using the line right when I'm trying to use it. 
How would
I know it was busy through AMI? Is it possible?

>>
>> So,
>> I'd like to use the asterisk manager interface in java to originate a
>> call from one ZAP-channel, calling out to my telephone provider,
>> And then they will direct the call back to my, but into another
>> ZAP-channel (since I'm calling that channel's number).

>For a basic test that the line works, try TestClient and TestServer .
>Originate a call from testclient (and set there the number. And set all
>incoming calls temporarily to go to TestServer (did I mention the
>assumption that noone calls in?)

>One relatively cheap method of "temporary" is through setting a global
>variable to a non-standard value. This means that the non-default value
>will never last after a reload. And you can set the global even through
>'core set global VARNAME VALUE' in the CLI.

>Check the resulting reports in /var/lib/asterisk/testresults . Make sure
>all of them were generated, and that none "FAIL"-ed.



I will look into this testserver & testclient. Totally new stuff to me :)
Hope I get it working.
Thanks for the tip Tzafrir !
Nice of you!



--
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue

[Andrew Smith]

Hi Tim,

Imagine the scenario where we had 10x Asterisk servers, with calls
presenting sequentially starting from the first server, then server two,
etc.
 
If we took down the first server for maintenance with 'asterisk -rx stop
gracefully' we then will block all incoming calls to all servers as our
telco will simply relay the BUSY back to the caller. If there are a number
of calls on the first server that continue for another 20 minutes, then all
inbounds are blocked for that period of time.
 
We are finding at present we have to look at the calls on the server and
make a decision if we are busy to simply reboot the server and hence lose
calls. Not ideal but then we don't end up blocking our inbounds.

What I was hoping to do was find a way to cause the telco to present the
call to the next ISDN30 and therefore would allow us to cleanly take down an
Asterisk server for maintenance without causing this issue. In a sense to
put the ISDN30 into alarm mode while still continuing the active calls.
 
Do you know if this is at all possible, even if we considered patching
zaptel to add this functionality or does the telco rely on the entire PRI
being in alarm before it presents the call to the next ISDN30 ? This would
allow us to run maintenance on our servers during busy periods without
causing disruption, and would be an excellent feature.

Many thanks,
Andrew

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Nelson
Sent: 13 February 2008 18:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ISDN PRIs and taking a server down for
maintenance - blocking issue


Even if * is shutdown, zaptel is still running and your ISDN channels are
still technically up. Shutting down zaptel should close the channels and put
those circuits into alarm mode.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332

- Original Message -
From: "Andrew Smith" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago
Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance
- blocking issue


Hi there,
 
I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN
E1s.

Basically our telco is presenting calls in order of the ISDNs on our
servers.
 
SERVER1=1,2,3,4
SERVER2=5,6,7,8
 
We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in
alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2.

If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown
gracefully) any incoming calls receive a BUSY tone.

What I would like to know is if there is anyway to get around this and not
send a BUSY back to our callers and somehow allow our telco to present calls
immediately to SERVER2.

Anyone have any ideas or are we stuck with this behaviour until the calls
drop to 0 and Asterisk shuts down ?

Thanks,
Andrew
 
 
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Re: [asterisk-users] UK -999 dialing issue

[SIP]

Gordon Henderson wrote:
> On Thu, 14 Feb 2008, Phil Knighton wrote:
>
>   
>> [softoption-zap]
>>
>> exten => _0[123456789].,1,NoOp(${EXTEN})
>> exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
>> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
>> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
>> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
>> 
>
> OMG!!!
>
> You're selecting 2 different output channels depending on the number 
> dialled!!!
>
> (UK or international)...
>
> That's ... LCR!!!
>
> In  ... Dialplan!!!
>
> And according to a recent thread, that's like ... impossible, not 
> recommended, really really hard, with databases and external hardware 
> required, etc. (!!!)
>
> (sorry)
>
> Gordon
> (dialplan junkie)
>
>   
Not impossible. I think the explanation was that it was ugly. And... 
well... that is.  Now, imagine sorting through a list of 500,000 
possible dialing prefixes (something we have) instead of 3 or 4. Tell me 
that would be clean and pretty without a DB lookup.

Anyone can LCR 2 routes in a dialplan, but that's hardly an effective 
example of LCR.

N.

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Re: [asterisk-users] UK -999 dialing issue

[Benny Amorsen]

"Steve Langstaff" <[EMAIL PROTECTED]> writes:

>> [softoption-zap]
>> 
>> exten => _0[123456789].,1,NoOp(${EXTEN})
>> exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
>> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
>> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
>> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)

[..]
> 2) It does not look like you would be able to dial numbers that
> start with 0[123456789] and then have subsequent zeros
> (e.g. 01xx xxx ) - is that your intent? 

. does not repeat the previous pattern, it simply matches one or more
of anything. _0Z. will happily match 010.


/Benny



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Re: [asterisk-users] [Linux/Python 2.4.2] Forking Python doesn't work

[Vincent]

On Wed, 13 Feb 2008 22:26:16 -0500, Russell Bryant
<[EMAIL PROTECTED]> wrote:
>The arguments to System() are a bit different.  Put it in just like you would 
>type at the command line.
>
>System(/tmp/netcid.py 2000 Joe)

That did it :-) Thanks guys.

BTW, for those interested, I didn't have to double-fork:

==
#!/usr/bin/python

import socket,sys,time,os

sys.stdout = open(os.devnull, 'w')
if os.fork():
sys.exit(0)
else:
#Here, send broadcast
==


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Re: [asterisk-users] UK -999 dialing issue

[Tilghman Lesher]

On Thursday 14 February 2008 03:39:33 Phil Knighton wrote:
> OK, the majority of our calls go out via zaptel fxo and pstn lines.
> When these are all busy, calls are routed via a VOIP provider here in
> the UK.  All activity is recorded in our logs, and I can find no trace
> of either 999 or 112 (if since been reminded that in the UK, you can now
> also use 112 which is consistent with continental Europe).
>
> I can't find a call placed at the relevant time that had these numbers,
> even as mid-part of a string.

I had a recent run-in with the provider who provides my toll-free numbers, as
they had gotten a subpoena for the identity of the customer who ran some
toll-frees that were being used for fraudulent purposes.  It turns out that
they had two number transposed prior to getting the subpoena, so not only
did they have the wrong customer, they subpoenaed the wrong provider.

Consider that if the police will not provide you records of the call, they may
have already discovered that they queried the wrong provider.

-- 
Tilghman

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Re: [asterisk-users] UK -999 dialing issue

[Steve Langstaff]

> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Benny Amorsen
> Sent: 14 February 2008 13:57

> "Steve Langstaff" <[EMAIL PROTECTED]> writes:
> 
> >> [softoption-zap]
> >> 
> >> exten => _0[123456789].,1,NoOp(${EXTEN}) exten => 
> >> _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
> >> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
> >> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
> >> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
> 
> [..]
> > 2) It does not look like you would be able to dial numbers that
> > start with 0[123456789] and then have subsequent zeros
> > (e.g. 01xx xxx ) - is that your intent? 
> 
> . does not repeat the previous pattern, it simply matches one 
> or more of anything. _0Z. will happily match 010.

Oops! Yes, I see that now - my fault for confusing Asterisk pattern
matching with RFC3435 pattern matching. Sorry.

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Re: [asterisk-users] Asterisk Manager and Visual Basic

[Lee Jenkins]

Bill Andersen wrote:
> Has anyone tried to used VB6 to communicate with the Asterisk Manager?
> 
> If so, would you be willing to share some basic code showing your
> approach to getting connected and parsing results?
> 
> I've got a Telnet control that is allowing me to connect, authenticate
> and see the "flow" of status, etc., but I'm sure there is a better way
> to do this without using Telnet (maybe not?).  Any suggestions?
> 
> I want to write a presence monitor (a virtual "sidecar" if you will)
> 
> Bill
> 

As Razza said, you can just use the winsock control included with VB.  The 
protocol is very simple, basically just name/value pairs delimited by #13#10 
(CRLF) with an extra CRLF at the end to denote termination of the packet.

Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=123432|CALLERID(name)=Automated Call
Async: true
 <== extra CRLF here.

So, like this:

1. Send your properly formatted packet to AMI .

2. Read incoming response terminated by double #13#10.

3. Parse values as you are comfortable with.

I am in the process of writing a similar product for one of our customers. 
Well, a re-write to add features and make it cross platform.  Here's a 
screenshot running on Linux/GTK:
http://leebo.dreamhosters.com/images/guiApp.png

A couple of side notes from what I've learned myself and read on this mailing 
list or through the wiki:

1. Packet Volume
The volume of messages that you can get from the AMI is impressive.  I've 
tested 
on our Asterisk system which has only 2 pots lines and two sip trunks with 10 
desktop phones and the amount of messages can be staggering!

Use a proxy for AMI if you have any decent phone traffic.  AstManProxy is VERY 
propular.  I wrote one as well, but its still beta and I think there's another 
one out there somewhere.  Usually with these proxy servers you can filter out 
unwanted/extraneous events to reduce the amount of messages your app has to 
contend with.

2. Make good use of Observer/Mediator pattern to distribute events to different 
parts of your GUI.  Monolithic loops to write everything out on a timer's event 
or after a Sleep() for instance, is not a good way to go in my experience.

3. Check the source for manager interface for changes between Asterisk 1.2 and 
1.4 (and 1.6?) if you're using 1.2 or plan to.  I believe the latest version of 
AMI is 1.1 (someone can correct me here).  A few label names for some of the 
AMI 
packets have been changed and a couple events (like LINK event) have been 
changed drastically.

I originally wrote against the 1.2 Manager interface only to find that I had to 
refactor some code and write descendant classes to handle the slight 
differences 
between the two versions' events.  I could have saved myself some work had I 
thought to look for the changes.  I think this link is up to date:
http://svn.digium.com/view/asterisk/trunk/doc/manager_1_1.txt?revision=98152&view=markup

Happy coding.


-- 
Warm Regards,

Lee

"Everything I needed to learn in life, I learned selling encyclopedias door to 
door."

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Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9

[Henry Devito]

I had GXP-2000's running on 1.0 versions of asterisk even earlier.  So I 
know it does work.  I upgraded one of my customers GXP's to the latest 
firmware in it still works.  Can you post the output of the CLI with verbose 
set to 99 and the the output from the asterisk log file that has the call in 
it.  You can usually do a 'tail /var/log/asterisk/full -n 400' right after 
the call fails.

I will be glad to help, just need a little more info to narrow down the 
issue.

Thanks
Henry


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

Sent: Thursday, February 14, 2008 2:15 AM
Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9


1. The phone has not the DND active, i checked it several times
2. Outbound calls always success, the problem is when the phone receive a 
call, it repsnds with busy signalling.
3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade 
asterisk.

Thanks for all

-Messaggio originale-
Da: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] Per conto di C F
Inviato: mercoledì 13 febbraio 2008 21.09
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

Just check DND if it's on on the phone or not.
What is the CLI output when you try making a phone call?
Why don't you try it with a later version of astrisk and a Phone?

On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>
>
> Hi all gusy,
> i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a 
> few
> go in "busy" state, if you call it get the busy tone but the phone can 
> male
> any type of call.
> This is my sip.conf
>
> [502]
> language = it
> username = 502
> secret = 
> host = dynamic
> type = friend
> context = local
> canreinvite = yes
> dtmfmode = info
> callgroup = 1
> pickupgroup = 1
> callerid = 502 <502>
>
> Under Grandstream's support suggest, I set "Use randmom port" to yes and
> "Nat traversal (STUN)" to "No, but send keep alive" but without success.
> This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6
>
> Anyone can help me ?
>
> Thanks in advance
>
> Giordano
>
>
> No virus found in this outgoing message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 
> 12/02/2008
> 15.20
>
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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 
15.20


No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 
20.00


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[asterisk-users] Monitor Asterisk

[Soumya Kat]

Thank you to all those who replied to my last query. For them and for the
suggestion, I can monitor asterisk using the asterisk -r -x ""
option. What I would like to know is that using asterisk -r -x way I can
only use the *CLI commands. Is there any other way in which I can monitor
asterisk?

Moreover it will be very helpful is someone can provide me the C file of the
ASTERISK-MIB. I will be using C language to develop my agent for the
monitoring of asterisk.

Thank you.
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[asterisk-users] Monitor Asterisk

[Soumya Kat]

Thank you to all those who replied to my last query. For them and for the
suggestion, I can monitor asterisk using the asterisk -r -x ""
option. What I would like to know is that using asterisk -r -x way I can
only use the *CLI commands. Is there any other way in which I can monitor
asterisk?

Moreover it will be very helpful is someone can provide me the C file of the
ASTERISK-MIB. I will be using C language to develop my agent for the
monitoring of asterisk.

Thank you.
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Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP

[Jaap Winius]

Quoting Steve Langstaff <[EMAIL PROTECTED]>:

> The "481 Call Leg/Transaction Does Not Exist" response to the
> NOTIFY makes me think that you might need to configure the
> phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
> from the phone when it is booted?

Yeah, sure. And there are some error messages mixed in too:

==

14:01:23.425955 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 473
...
SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
14:01:23.426075 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 509
[EMAIL PROTECTED]
...vSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.5
14:01:23.480238 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 634
E..k...
..F.SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
14:01:23.480375 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 432
[EMAIL PROTECTED]
...)SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.5:50
14:01:23.918830 arp who-has gigaset.umrk.to tell bitis.umrk.to
../.E
..
14:01:23.921726 arp reply gigaset.umrk.to is-at 00:01:e3:77:f8:67 (oui  
Unknown)
...w.g../.E
..
14:01:24.539636 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 476
E..
..2gSUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
14:01:24.539816 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 512
[EMAIL PROTECTED]
...ySIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.5
14:01:24.594442 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 634
E..i...
SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
14:01:24.594557 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 432
E...- [EMAIL PROTECTED]
...)SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.5:50

==

Before this was a series of REGISTER messages, and afterwards a series  
of OPTIONS messages. However, no errors there.

Also, this is without having set 'mailbox=1000' or '[EMAIL PROTECTED]' in
/etc/asterisk/sip.conf. And, now that I look at it again, the network  
mailbox settings for the Siemens phone won't have anything to do with  
these errors either, since it simply makes it possible to associate a  
button on each handset with an extension used to access a voicemail  
account.

Thanks,

Jaap

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[asterisk-users] X100P Burnouts

[Brent Davidson]

Thought I would post this experience to the list so it's archived for 
posterity...  My company is deploying Asterisk-based PBX's to all of our 
branch offices.  Each office has 2 analog Voice lines and a fax line.  
We didn't want to go to the expense of using TDM400's in the servers 
(which run asterisk and Hylafax) so we opted for 2 X100P cards in each 
box.  So far they have worked fine at all but one office.  The system at 
the office in question would work perfectly for an entire day once it 
was set up.  The next morning, however, one of the 2 phone lines would 
appear to be dead and the X100P would be in Red alarm.  Plugging a phone 
into the X100P's pass-through connection would show dial-tone on the 
line, and the phone worked perfectly.  It was as if the X100p lost it's 
ability to see the audio on the line and nothing would revive it.  Tried 
restarting the zaptel module, rebooting the server completely, complete 
power down, unplugging the phone line and even connecting up a phone 
line simulator and moving the card to another server. The card never 
works again.  This went on for three days.  Burned out an X100p every 
night.  I called the telco (Verizon) and they sent out a couple of guys 
to run tests on the line, but found nothing.  Their Demarc is properly 
grounded and has surge suppression modules attached, the cable that runs 
from their demarc to our punch-down block is in grounded metal conduit 
and does not run near any power source.  The cable that runs from the 
punch-down block to the wall jack also does not run anywhere near 
anything electrical, and everything is twisted pair all the way from the 
wall jack to the demarc.  To further eliminate the possibility of echo 
or other noise, I ran twisted pair carrying both voice lines from the 
wall jack to the server, approximately 3 feet in length.  Now here is 
where things get interesting...  That 3 foot cable run passes behind a 
21" monitor that was connected to the server.  When the line tests 
showed everything OK, I decided the monitor might be a long shot but I 
could understand how the degaussing coil coil could possibly induce a 
surge on the phone line if the monitor was somehow degaussing nightly, 
so I unplugged the monitor's power cable and left everything else as it 
was.  So far so good.  X100p #4 is still working this morning, so it 
looks like the problem is solved.  Hope this helps someone else later on.

Thanks,
Brent Davidson

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Re: [asterisk-users] Monitor Asterisk

[Matthew J. Roth]

Soumya Kat wrote:
> Thank you to all those who replied to my last query. For them and for 
> the suggestion, I can monitor asterisk using the asterisk -r -x 
> "" option. What I would like to know is that using asterisk 
> -r -x way I can only use the *CLI commands. Is there any other way in 
> which I can monitor asterisk?
Soumya,

Yes, "asterisk -rx" will only allow you to execute CLI commands.  It 
also tends to spew out a bunch of garbage that makes parsing difficult 
unless verbosity is always set to 0.

I recommend taking a look at the Asterisk Manager Interface (AMI) 
.  It's a cleaner 
interface that will allow you to read events and issue commands.  All of 
the CLI commands are available through the AMI, as well as an array of 
additional manager actions.

I recently wrote a program that maps the SIP call IDs of the two legs 
that make up a call using the 
"POE::Component::Client::Asterisk::Manager" Perl module 
.  
It provides a simple interface for filtering events, so if you're 
familiar with Perl I recommend taking a look at it.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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Re: [asterisk-users] Ser, asterisk and ip2ipgw

[Alex Balashov]

Riccardo Cupardo wrote:
> Hi,
> 
> i use a ser, as proxy sip server(authentication), then a cisco router as 
> sip2h323 gw(authorization and accounting). i want to start asterisk as 
> sip statefull b2bua server, any suggestion to howto or documentation to 
> asterisk integration and b2b use?

Well, Asterisk is a B2BUA.  And it keeps state.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] IAX load balancing

[Cavalera Claudio Luigi]

Hello,
I've seen that many solutions concerning asterisk dimensioning and load
balancing involve the use of sip proxy like openser.
Is there any recommended way to balance IAX load?

BRs,
Claudio


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[asterisk-users] translating iax2 register into sip register

[Cavalera Claudio Luigi]

Hello,
reading iax2 draft, I'm not sure if the protocol supports peer 2 peer
calls (e.g. like SIP).
If it doesn't, is Asterisk the only server side iax2 implementation?

I also would like to understand if it's possible for asterisk (by means
of some configuration rules) to translate a iax2 register into a sip
register in order to have user credentials verified by an external
entity. I don't like the idea to have all users provisioned within
asterisk in order to make them iax capable.

Regards,
Claudio


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Re: [asterisk-users] Asterisk Manager and Visual Basic

[Bill Andersen]

> I don't know if it would be of any use to you but we have some C# code
> that handles the basics of communicating the the Asterisk Manager
> Interface. It doesn't do anything fancy just sends single commands and
> checks the responses. We don't use it for monitoring.
> 
> Regards,
> 
> Greyman.

Thanks for the offer, I think I've got it figured out using winsock.

Thanks again.

Bill


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[asterisk-users] Pass arguments from extensions.conf

[Naveen Palani]

Hi,

I have been working with asterisk to make ivr calls (outbound and inbound). I 
have the functionality -

Read(variable|file_name)

used in my dialplan. Now i need to pass the variable to my ruby file to compare 
the data entered with the database (mysql).

How can i pass the arguments from my dialplan to the ruby file. Is there a way 
i can do it with the agi script?

Any one has any clues on it.

Regards,
Naveen.Palani



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[asterisk-users] SNMP monitoring

[Adrian Marsh]

Hi All,

 

I've been reading up on 1.4 snmp integration. When I try and compile
asterisk with a -with-netsnmp option it complains about net-snmp
installation being broken. However, the net-snmp-devel rpm is installed,
and snmpd on the machine runs fine.

 

Anyone have a guide for the pre-requisites needed ?

 

Cheers,

 

Adrian

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Re: [asterisk-users] UK -999 dialing issue

[Tilghman Lesher]

On Thursday 14 February 2008 07:55:08 SIP wrote:
> Gordon Henderson wrote:
> > On Thu, 14 Feb 2008, Phil Knighton wrote:
> >> [softoption-zap]
> >>
> >> exten => _0[123456789].,1,NoOp(${EXTEN})
> >> exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
> >> exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
> >> exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
> >> exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
> >
> > OMG!!!
> >
> > You're selecting 2 different output channels depending on the number
> > dialled!!!
> >
> > (UK or international)...
> >
> > That's ... LCR!!!
> >
> > In  ... Dialplan!!!
> >
> > And according to a recent thread, that's like ... impossible, not
> > recommended, really really hard, with databases and external hardware
> > required, etc. (!!!)
>
> Not impossible. I think the explanation was that it was ugly. And...
> well... that is.  Now, imagine sorting through a list of 500,000
> possible dialing prefixes (something we have) instead of 3 or 4. Tell me
> that would be clean and pretty without a DB lookup.
>
> Anyone can LCR 2 routes in a dialplan, but that's hardly an effective
> example of LCR.

Right, and as soon as you add func_odbc to the mix, it becomes easy to query
such a database  in the dialplan.

-- 
Tilghman

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Re: [asterisk-users] SIP over TCP

[Razza]

On 13/02/2008, Raj Jain <[EMAIL PROTECTED]> wrote:
>
> SIP over TCP is included in 1.6.
> http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co


Thanks all! :o)
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Re: [asterisk-users] X100P Burnouts

[Steve Edwards]

On Thu, 14 Feb 2008, Brent Davidson wrote:

> That 3 foot cable run passes behind a
> 21" monitor that was connected to the server.  When the line tests
> showed everything OK, I decided the monitor might be a long shot but I
> could understand how the degaussing coil coil could possibly induce a
> surge on the phone line if the monitor was somehow degaussing nightly,
> so I unplugged the monitor's power cable and left everything else as it
> was.  So far so good.  X100p #4 is still working this morning, so it
> looks like the problem is solved.  Hope this helps someone else later on.

Indicative, but not conclusive.

It would be interesting to disconnect the "pairs" from the x100p's and 
plug them into an oscilloscope or something similar and power-cycle, 
auto-tune, sleep-mode, and degauss the tube to see if your assumption is 
reasonable.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] Variable setting in AMI Originate

[Anthony Messina]

Working with asterisk 1.4; using the AMI Originate command, it is possible to 
do something like:

Variable: CDR(accountcode)123456

Or must the variable names be "var[n]" where n is a number?

I'd like to set the accountcode for a Local channel that originates a call.

Thanks.  -A

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports

[bilal ghayyad]

Hi;

Sorry, I forgot to post the zapata version, it is 1.4
but I do not know the release and I do not know how to
know the exact release.

Regards
Bilal


--
> Hi All;
> 
> I am facing a problem that the telephon line in
Egypt
> does not work with the FXO port at the digium card
> (TDM22B), and I tried to play in loadzone and
> defaultzone without any success, when we call to the
> PBX it gives Busy signal sometimes, and othertimes
it
> rings without any response in Asterisk.
> 
> Is there any other configuration I have to do it to
> resolve this issue? Any advise about a
troubleshooting
> method to resolve it?

What version of zaptel? What do you have in
zaptel.conf?



  

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[asterisk-users] Emagen (a Telrad VM solution) -- any way to replace with *?

[Ken D'Ambrosio]

Hi, all.  I've got a PoS Emagen VM system tied in with our Telrad PBX.  I
hate 'em both, but I'm stuck with the Telrad for the time being.  That
being said, does anyone know of a way to replace the VM solution with
Asterisk?  I'd -love- to get an Asterisk box in the loop, here.

Thanks,

-Ken


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Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports

[bilal ghayyad]

Hi;

The PBX located in Egypt at Cairo city.

I am able to receive calls on the FXO ports at 3rd and
4th ports, but I am not able to place outgoing call
(it gives busy tone that coming from the service
provider, or it gives an voice message from the
service provider that the dialed number is wrong).

I am posting now the needed data that I was able to
collect it:

Asterisk version: Asterisk SVN-branch-1.4-r90231 

localhost*CLI> show globals
   ignorepat=9
   TRUNKMSD=1
   IAXINFO=guest
   CONSOLE=Console/dsp
   PSTNTRUNK=Zap/g1
   TRUNK=Zap/g2


zap show status
Description  Alarms
IRQbpviol CRC4
Wildcard TDM400P REV I Board 1   OK 0 
0  0


zap show cadences
r1: 125,125,2000,4000
r2: 250,250,500,1000,250,250,500,4000
r3: 125,125,125,125,125,4000
r4: 1000,500,2500,5000

zapata.conf:

[channels]
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=no
hidecallerid=no
callwaiting=yes
usecallingpres=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=no
relaxdtmf=yes
rxgain=15.0
txgain=15.0
group=1
channel => 3
callgroup=1
pickupgroup=1
immediate=yes
busydetect=yes
busycount=3
hanguponpolarityswitch=yes
callprogress=yes

context=Internal
signalling=fxo_ks
channel => 1,2

context=External
signalling=fxs_ks
channel => 3,4

zaptel.conf:

loadzone = uk
defaultzone=uk
fxoks=1,2
fxsks=3,4

Any advise?
Regards
Bilal


-
> Hi All;
> 
> I am facing a problem that the telephon line in
Egypt
> does not work with the FXO port at the digium card
> (TDM22B), and I tried to play in loadzone and
> defaultzone without any success, when we call to the
> PBX it gives Busy signal sometimes, and othertimes
it
> rings without any response in Asterisk.
> 
> Is there any other configuration I have to do it to
> resolve this issue? Any advise about a
troubleshooting
> method to resolve it?

What version of zaptel? What do you have in
zaptel.conf?



  

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know-it-all with Yahoo! Mobile.  Try it now.  
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Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue

[Matt]

Honestly.. this sounds like a telco issue.I understand what the other
person is saying about the PRI still being technically up... BUT... if the
channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to
the next available channel, which they clearly are not doing.

On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith <[EMAIL PROTECTED]> wrote:

>  Hi Tim,
> Imagine the scenario where we had 10x Asterisk servers, with calls
> presenting sequentially starting from the first server, then server two,
> etc.
>
> If we took down the first server for maintenance with 'asterisk -rx stop
> gracefully' we then will block all incoming calls to all servers as our
> telco will simply relay the BUSY back to the caller. If there are a number
> of calls on the first server that continue for another 20 minutes, then all
> inbounds are blocked for that period of time.
>
> We are finding at present we have to look at the calls on the server and
> make a decision if we are busy to simply reboot the server and hence lose
> calls. Not ideal but then we don't end up blocking our inbounds.
>
> What I was hoping to do was find a way to cause the telco to present the
> call to the next ISDN30 and therefore would allow us to cleanly take down an
> Asterisk server for maintenance without causing this issue. In a sense to
> put the ISDN30 into alarm mode while still continuing the active calls.
>
> Do you know if this is at all possible, even if we considered patching
> zaptel to add this functionality or does the telco rely on the entire PRI
> being in alarm before it presents the call to the next ISDN30 ? This would
> allow us to run maintenance on our servers during busy periods without
> causing disruption, and would be an excellent feature.
>
> Many thanks,
> Andrew
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Tim Nelson
> *Sent:* 13 February 2008 18:12
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Cc:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down for
> maintenance - blocking issue
>
> Even if * is shutdown, zaptel is still running and your ISDN channels are
> still technically up. Shutting down zaptel should close the channels and put
> those circuits into alarm mode.
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332
>
> - Original Message -
> From: "Andrew Smith" <[EMAIL PROTECTED]>
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago
> Subject: [asterisk-users] ISDN PRIs and taking a server down for
> maintenance - blocking issue
>
>  Hi there,
>
> I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN
> E1s.
>
> Basically our telco is presenting calls in order of the ISDNs on our
> servers.
>
> SERVER1=1,2,3,4
> SERVER2=5,6,7,8
>
> We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in
> alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2.
>
> If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown
> gracefully) any incoming calls receive a BUSY tone.
>
> What I would like to know is if there is anyway to get around this and not
> send a BUSY back to our callers and somehow allow our telco to present calls
> immediately to SERVER2.
>
> Anyone have any ideas or are we stuck with this behaviour until the calls
> drop to 0 and Asterisk shuts down ?
>
> Thanks,
> Andrew
>
>
>
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Re: [asterisk-users] X100P Burnouts

[Brent Davidson]

I considered doing just that, but since I didn't have my scope with me 
and it's an hour's drive away it didn't seem worth it at this point.  If 
we have trouble again I may take the scope down there and test it.


-Brent

Steve Edwards wrote:

On Thu, 14 Feb 2008, Brent Davidson wrote:

  

That 3 foot cable run passes behind a
21" monitor that was connected to the server.  When the line tests
showed everything OK, I decided the monitor might be a long shot but I
could understand how the degaussing coil coil could possibly induce a
surge on the phone line if the monitor was somehow degaussing nightly,
so I unplugged the monitor's power cable and left everything else as it
was.  So far so good.  X100p #4 is still working this morning, so it
looks like the problem is solved.  Hope this helps someone else later on.



Indicative, but not conclusive.

It would be interesting to disconnect the "pairs" from the x100p's and 
plug them into an oscilloscope or something similar and power-cycle, 
auto-tune, sleep-mode, and degauss the tube to see if your assumption is 
reasonable.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports

[Mojo with Horan & Company, LLC]

bilal ghayyad wrote:
> [channels]
> rxgain=15.0
> txgain=15.0
>   
Wow!  Is this necessary?  Is this something you took from a sample 
config somewhere, or numbers that you arrived at through trial and 
error?  They seem a bit high in my experience, *but* I've never been to 
Egypt before, and I sure wouldn't be surprised if this was necessary -- 
just wanted your confirmation ;)

Mojo

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Re: [asterisk-users] X100P Burnouts

[Jon Pounder]

Quoting Brent Davidson <[EMAIL PROTECTED]>:

> I considered doing just that, but since I didn't have my scope with me
> and it's an hour's drive away it didn't seem worth it at this point.
> If we have trouble again I may take the scope down there and test it.
>

unless the cable is in the same spot (relative to coil, ground, other  
metal), loaded with the same impedance characteristics as an x100p,  
etc etc you might not measure anything meaningful anyway. you'd have  
to wire the scope across it somehow in place without destroying the  
experiment with the scope leads themselves adding another end of line  
leg that is not the x100p.

I would say you are probably correct in the assumption, and everyone  
should just bear this sort of thing in mind. if in doubt its easy to  
plug in through a power bar with phone surge supressor close to the  
end of the run, or one of the many other devices that do the same thing.


> -Brent
>
> Steve Edwards wrote:
>> On Thu, 14 Feb 2008, Brent Davidson wrote:
>>
>>
>>> That 3 foot cable run passes behind a
>>> 21" monitor that was connected to the server.  When the line tests
>>> showed everything OK, I decided the monitor might be a long shot but I
>>> could understand how the degaussing coil coil could possibly induce a
>>> surge on the phone line if the monitor was somehow degaussing nightly,
>>> so I unplugged the monitor's power cable and left everything else as it
>>> was.  So far so good.  X100p #4 is still working this morning, so it
>>> looks like the problem is solved.  Hope this helps someone else later on.
>>>
>>
>> Indicative, but not conclusive.
>>
>> It would be interesting to disconnect the "pairs" from the x100p's   
>> and plug them into an oscilloscope or something similar and   
>> power-cycle, auto-tune, sleep-mode, and degauss the tube to see if   
>> your assumption is reasonable.
>>
>> Thanks in advance,
>> 
>> Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
>> Newline Fax: +1-760-731-3000
>>
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>>
>>



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Re: [asterisk-users] message: !! Got Busy in Connected State !?!

[Vieri]

--- Fons van der Beek <[EMAIL PROTECTED]>
wrote:

> What phone do you use?
> Linksys ?

SIP softphones and Alcatel analog phones behind ATA
gateways (Grandstream). However, I'm having a hard
time reproducing the problem. It doesn't happen often.



  

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Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue

[Matt]

That does sound like what is happening.. Telco knows channel 1-23 are not
busy (so far as they are concerned), however.. so far as you are concerned,
they are busy.. so telco sends the call down... but the equipment doesn't
take it.

I would *think* the Telco could keep trying channels down the hunt group,
but maybe not?  We have, in the past, seen this issue with our dial-up modem
banks.. especially if I would take one offline.   However, it is not a big
enough issue (i.e. we don't take things down that often) for me to look into
it fully.

On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly <[EMAIL PROTECTED]> wrote:

>  I think the problem is that the telco presents the call on a specific
> channel, then zaptel tells it that the channel is busy.
>
>
>
> We need to be able to tell the telco that each unused channel on a given
> span is unavailable, and it will determine that the others are in use and
> will present the call on a channel on another span.
>
>
>
> A rather ugly work-around (since Andrew seems to have lots of channels
> available, and one would assume that maintenance of this nature would occur
> during slow periods) would be to make calls to a DID in the same trunk group
> on all "idle" channels on the span shutting down then, when all channels on
> the span are "in use" and none of them are doing anything useful, take the
> span down hard so the telco will divert all calls to another span.
>
>   --Don
>
> Don Kelly
> PCF Corp
> Real Support for your Virtual Office TM
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>   --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Matt
> *Sent:* Thursday, February 14, 2008 2:28 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down
> formaintenance - blocking issue
>
>
>
> Honestly.. this sounds like a telco issue.I understand what the other
> person is saying about the PRI still being technically up... BUT... if the
> channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to
> the next available channel, which they clearly are not doing.
>
> On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith <[EMAIL PROTECTED]>
> wrote:
>
> Hi Tim,
>
> Imagine the scenario where we had 10x Asterisk servers, with calls
> presenting sequentially starting from the first server, then server two,
> etc.
>
>
>
> If we took down the first server for maintenance with 'asterisk -rx stop
> gracefully' we then will block all incoming calls to all servers as our
> telco will simply relay the BUSY back to the caller. If there are a number
> of calls on the first server that continue for another 20 minutes, then all
> inbounds are blocked for that period of time.
>
>
>
> We are finding at present we have to look at the calls on the server and
> make a decision if we are busy to simply reboot the server and hence lose
> calls. Not ideal but then we don't end up blocking our inbounds.
>
> What I was hoping to do was find a way to cause the telco to present the
> call to the next ISDN30 and therefore would allow us to cleanly take down an
> Asterisk server for maintenance without causing this issue. In a sense to
> put the ISDN30 into alarm mode while still continuing the active calls.
>
>
>
> Do you know if this is at all possible, even if we considered patching
> zaptel to add this functionality or does the telco rely on the entire PRI
> being in alarm before it presents the call to the next ISDN30 ? This would
> allow us to run maintenance on our servers during busy periods without
> causing disruption, and would be an excellent feature.
>
> Many thanks,
>
> Andrew
>
>
>  --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Tim Nelson
> *Sent:* 13 February 2008 18:12
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Cc:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down for
> maintenance - blocking issue
>
> Even if * is shutdown, zaptel is still running and your ISDN channels are
> still technically up. Shutting down zaptel should close the channels and put
> those circuits into alarm mode.
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332
>
> - Original Message -
> From: "Andrew Smith" <[EMAIL PROTECTED]>
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago
> Subject: [asterisk-users] ISDN PRIs and taking a server down for
> maintenance - blocking issue
>
> Hi there,
>
>
>
> I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN
> E1s.
>
>
> Basically our telco is presenting calls in order of the ISDNs on our
> servers.
>
>
>
> SERVER1=1,2,3,4
> SERVER2=5,6,7,8
>
>
>
> We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in
> alarm and the calls will then presented to PRIs 5,6,7,8 on SER

Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue

[Don Kelly]

I think the problem is that the telco presents the call on a specific
channel, then zaptel tells it that the channel is busy.

 

We need to be able to tell the telco that each unused channel on a given
span is unavailable, and it will determine that the others are in use and
will present the call on a channel on another span.

 

A rather ugly work-around (since Andrew seems to have lots of channels
available, and one would assume that maintenance of this nature would occur
during slow periods) would be to make calls to a DID in the same trunk group
on all "idle" channels on the span shutting down then, when all channels on
the span are "in use" and none of them are doing anything useful, take the
span down hard so the telco will divert all calls to another span.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax



  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, February 14, 2008 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN PRIs and taking a server down
formaintenance - blocking issue

 

Honestly.. this sounds like a telco issue.I understand what the other
person is saying about the PRI still being technically up... BUT... if the
channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to
the next available channel, which they clearly are not doing.

On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith <[EMAIL PROTECTED]> wrote:

Hi Tim,

Imagine the scenario where we had 10x Asterisk servers, with calls
presenting sequentially starting from the first server, then server two,
etc.

 

If we took down the first server for maintenance with 'asterisk -rx stop
gracefully' we then will block all incoming calls to all servers as our
telco will simply relay the BUSY back to the caller. If there are a number
of calls on the first server that continue for another 20 minutes, then all
inbounds are blocked for that period of time.

 

We are finding at present we have to look at the calls on the server and
make a decision if we are busy to simply reboot the server and hence lose
calls. Not ideal but then we don't end up blocking our inbounds.

What I was hoping to do was find a way to cause the telco to present the
call to the next ISDN30 and therefore would allow us to cleanly take down an
Asterisk server for maintenance without causing this issue. In a sense to
put the ISDN30 into alarm mode while still continuing the active calls.

 

Do you know if this is at all possible, even if we considered patching
zaptel to add this functionality or does the telco rely on the entire PRI
being in alarm before it presents the call to the next ISDN30 ? This would
allow us to run maintenance on our servers during busy periods without
causing disruption, and would be an excellent feature.

Many thanks,

Andrew

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Nelson
Sent: 13 February 2008 18:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ISDN PRIs and taking a server down for
maintenance - blocking issue

Even if * is shutdown, zaptel is still running and your ISDN channels are
still technically up. Shutting down zaptel should close the channels and put
those circuits into alarm mode.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332

- Original Message -
From: "Andrew Smith" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago
Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance
- blocking issue

Hi there,

 

I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN
E1s.


Basically our telco is presenting calls in order of the ISDNs on our
servers.

 

SERVER1=1,2,3,4
SERVER2=5,6,7,8

 

We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in
alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2.

If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown
gracefully) any incoming calls receive a BUSY tone.

What I would like to know is if there is anyway to get around this and not
send a BUSY back to our callers and somehow allow our telco to present calls
immediately to SERVER2.

Anyone have any ideas or are we stuck with this behaviour until the calls
drop to 0 and Asterisk shuts down ?

Thanks,
Andrew

 

 


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[asterisk-users] ExtenSpy strange behavior on Asterisk 1.4.18

[Jose P. Espinal]

Hi list,

I have been experiencing a strange behavior with asterisk and i would 
like to know if someone else has face it.


This is my scenario,

3 extensions created on sip.conf: 121 | 123 | 123

Everything work just perfect except for the following issue:

I have this block on my extensions.conf

[record] ;---Extensiones individuales
exten => _7781[23]X,1,Authenticate(/etc/asterisk/eavepass|am)
  ; Authenticate
exten => 
_7781[23]X,n,extenspy(${EXTEN:3:[EMAIL PROTECTED]|bqr(${EXTEN:3:6}-))  ; 
extenspy([EMAIL PROTECTED]|bqr(fileprefix-)
exten => _7781[23]X,n,Hangup 
  ; Hangup



If 122 calls 123 , and 123 answers ( 122 =[called]> 123 ):
- Dialing 778*122* from 121 (for spying and recording) generates a 0 
byte file like 122-.1203025287.raw (on /var/spool/asterisk/monitor/)
- Dialing 778*123* from 121 (for spying and recording) generates a 
byte-significant file like 123-.1203025598.raw


If 123 calls 122 , and 122 answers ( 123 =[called]> 122 ):
- Dialing 778*122* from 121 (for spying and recording) generates a 
byte-significant file like 122-.1203025808.raw
- Dialing 778*123* from 121 (for spying and recording) generates a 0 
byte file like 123-.1203025923.raw


In the situation where the 0 byte file is generated, NO AUDIO is 
listened on 121; I have NO IDEA of what could be causing this, as 
there's no apprently refference on the documentation. Could this be a 
possible bug?



Regards,


# --
# Jose P. Espinal
# http://www.slackware-es.com
# --

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Re: [asterisk-users] SNMP monitoring

[Ricardo Carvalho]

I had the same problem some time ago...
You got to install also this packages:

net-snmp-devel
newt-devel
lm_sensors-devel
bzip2-devel

That should do it!

Regards,
Ricardo Carvalho.




On Thu, Feb 14, 2008 at 5:30 PM, Adrian Marsh <[EMAIL PROTECTED]>
wrote:

>  Hi All,
>
>
>
> I've been reading up on 1.4 snmp integration. When I try and compile
> asterisk with a –with-netsnmp option it complains about net-snmp
> installation being broken. However, the net-snmp-devel rpm is installed, and
> snmpd on the machine runs fine.
>
>
>
> Anyone have a guide for the pre-requisites needed ?
>
>
>
> Cheers,
>
>
>
> Adrian
>
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Re: [asterisk-users] UK -999 dialing issue

[Benny Amorsen]

"Steve Langstaff" <[EMAIL PROTECTED]> writes:

> Oops! Yes, I see that now - my fault for confusing Asterisk pattern
> matching with RFC3435 pattern matching. Sorry.

Unfortunately inventing a new regex syntax seems to be a favourite
pastime.

Perhaps it would be possible to allow exten => /00.*/,Dial... It might
cause problems with the ex-GF syntax. Another starting character could
mean RFC3435 pattern matching.


/Benny



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Re: [asterisk-users] Monitor Asterisk

[Benny Amorsen]

"Matthew J. Roth" <[EMAIL PROTECTED]> writes:

> Yes, "asterisk -rx" will only allow you to execute CLI commands.  It 
> also tends to spew out a bunch of garbage that makes parsing difficult 
> unless verbosity is always set to 0.

It would be very handy if it was possible to turn off messages that
aren't the direct result of a command in a particular instance of
asterisk -r. Perhaps asterisk -r -q?


/Benny



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Re: [asterisk-users] UK -999 dialing issue

[Jared Smith]

On Thu, 2008-02-14 at 22:32 +0100, Benny Amorsen wrote:
> Perhaps it would be possible to allow exten => /00.*/,Dial... It might
> cause problems with the ex-GF syntax. Another starting character could
> mean RFC3435 pattern matching.

I've been suggesting that for about four years now (long before I ever
started working for Digium), but the core Asterisk developers tell me it
will have a very negative impact on Asterisk performance.  I'd obviously
like to see it use a different character to inform the dialplan parser
that we're using a different pattern matching system, so that we can
limit the performance impact to just those extensions that require it.

But, for now, I've lived to learn to get along with the things that the
dialplan provides.  Don't forget that we have two different regex
operators what we can use inside of an Asterisk dialplan expression. :-)

exten => _X.,1,GotoIf($[${EXTEN} : /#+[2-7][0-9]{3}/]?happy)

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] gtalk and dtmf

[Adam KOSA]

Hi,

i've just finished setting up gtalk connection with asterisk.  it works 
nice, audio is full duplex.

i just have one question which i could not find an exact answer to.  Is 
gtalk able to send dtmf codes?  Because i'd like to listen to my 
voicemails while away from home.

I've been googling for half an hour, i found some sort of jingle 
protocol which i'm not sure what to use for but it might be the 
solution?  It seems to me that my problem is sending the dtmf tones, not 
receiving them, so this is really gtalk related.

I'm writing here because i read many of you have successfully integrated 
gtalk to asterisk and hoping somebody have a solution or at least some 
direction where i can move forward to.

thanks
Adam

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Re: [asterisk-users] restart asterisk daily

[Doug Bailey]

If you want to flush your disk cache to see how much memory is being eaten 
cache pages, try this:
 echo 3 >/proc/sys/vm/drop_caches

- "ast erisk" <[EMAIL PROTECTED]> wrote:
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Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue

[Lyle Giese]

If you take Asterisk down, the PRI should go down as the D channel is
down.  Then the telco should KNOW that there is trouble with the PRI and
those channels are in trouble busy and not availible.  If the telco
still tries to push a call to a channel on a PRI that is down, then the
telco is at fault.

Lyle

Matt wrote:
> That does sound like what is happening.. Telco knows channel 1-23 are
> not busy (so far as they are concerned), however.. so far as you are
> concerned, they are busy.. so telco sends the call down... but the
> equipment doesn't take it.
>
> I would *think* the Telco could keep trying channels down the hunt
> group, but maybe not?  We have, in the past, seen this issue with our
> dial-up modem banks.. especially if I would take one offline.  
> However, it is not a big enough issue (i.e. we don't take things down
> that often) for me to look into it fully.
>
> On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly <[EMAIL PROTECTED]
> > wrote:
>
> I think the problem is that the telco presents the call on a
> specific channel, then zaptel tells it that the channel is busy.
>
>  
>
> We need to be able to tell the telco that each unused channel on a
> given span is unavailable, and it will determine that the others
> are in use and will present the call on a channel on another span.
>
>  
>
> A rather ugly work-around (since Andrew seems to have lots of
> channels available, and one would assume that maintenance of this
> nature would occur during slow periods) would be to make calls to
> a DID in the same trunk group on all "idle" channels on the span
> shutting down then, when all channels on the span are "in use" and
> none of them are doing anything useful, take the span down hard so
> the telco will divert all calls to another span.
>
>   --Don
>
> Don Kelly
> PCF Corp
> Real Support for your Virtual Office TM
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
> 
>
> *From:* [EMAIL PROTECTED]
> 
> [mailto:[EMAIL PROTECTED]
> ] *On Behalf Of *Matt
> *Sent:* Thursday, February 14, 2008 2:28 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down
> formaintenance - blocking issue
>
>  
>
> Honestly.. this sounds like a telco issue.I understand what
> the other person is saying about the PRI still being technically
> up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco
> should be forwarding the call to the next available channel, which
> they clearly are not doing.
>
> On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith
> <[EMAIL PROTECTED] > wrote:
>
> Hi Tim,
>
> Imagine the scenario where we had 10x Asterisk servers, with calls
> presenting sequentially starting from the first server, then
> server two, etc.
>
>  
>
> If we took down the first server for maintenance with 'asterisk
> -rx stop gracefully' we then will block all incoming calls to all
> servers as our telco will simply relay the BUSY back to the
> caller. If there are a number of calls on the first server that
> continue for another 20 minutes, then all inbounds are blocked for
> that period of time.
>
>  
>
> We are finding at present we have to look at the calls on the
> server and make a decision if we are busy to simply reboot the
> server and hence lose calls. Not ideal but then we don't end up
> blocking our inbounds.
>
> What I was hoping to do was find a way to cause the telco to
> present the call to the next ISDN30 and therefore would allow us
> to cleanly take down an Asterisk server for maintenance without
> causing this issue. In a sense to put the ISDN30 into alarm mode
> while still continuing the active calls.
>
>  
>
> Do you know if this is at all possible, even if we considered
> patching zaptel to add this functionality or does the telco rely
> on the entire PRI being in alarm before it presents the call to
> the next ISDN30 ? This would allow us to run maintenance on our
> servers during busy periods without causing disruption, and would
> be an excellent feature.
>
> Many thanks,
>
> Andrew
>
>  
>
> 
>
> *From:* [EMAIL PROTECTED]
> 
> [mailto:[EMAIL PROTECTED]
> ] *On Behalf Of
> *Tim Nelson
> *Sent:* 13 February 2008 18:12
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Cc:* asterisk-users@lists.digium.com
> 
> *Subject:* Re: [asterisk-users] ISDN PRIs

Re: [asterisk-users] SNMP monitoring

[Darrick Hartman (lists)]

Ricardo Carvalho wrote:
> I had the same problem some time ago...
> You got to install also this packages:
> 
> net-snmp-devel
> newt-devel
> lm_sensors-devel
> bzip2-devel
> 
> That should do it!

Why would this depend on newt?  net-snmp and lm-sensor headers and 
libraries make sense.  newt doesn't make any sense as a dependency.

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue

[Brent Davidson]

Correct me if I'm wrong, but as I understand it your issue is that when 
you give Asterisk the "stop gracefully" command it waits until all 
active calls have finished before it takes the ISDN down but gives busy 
signals to new incoming calls on idle channels.  If this is the case 
then it would seem that Asterisk is actually answering the call on the 
incoming channel and playing a busy signal.  From reading a couple of 
threads on another list it appears this is the case (Google: Asterisk 
"busy out" PRI to find the discussion).  There also appears to be some 
interest in making a function do what you need in the future.


For the time being, however, a simple solution would be to create a 
temporary dial-plan that follows each outgoing hangup with a "dial" 
command to either a test number or some other service that will just 
keep playing audio down the line and not hangup.  (You'd probably need 
to set some variable to know which channels had been "busied") When you 
need to take down a server, load this dial plan and wait for all 
channels to call the "busy" number, then hang them all up and issue a 
"stop now".


It's a messy solution, but it's all I can think of without hacking 
code.  The only other way I'd know would be to hack the code for the 
dial or answer command and build another command that simply takes the 
channel off-hook and leaves it there.


Good luck,
Brent Davidson

Lyle Giese wrote:
If you take Asterisk down, the PRI should go down as the D channel is 
down.  Then the telco should KNOW that there is trouble with the PRI 
and those channels are in trouble busy and not availible.  If the 
telco still tries to push a call to a channel on a PRI that is down, 
then the telco is at fault.


Lyle

Matt wrote:
That does sound like what is happening.. Telco knows channel 1-23 are 
not busy (so far as they are concerned), however.. so far as you are 
concerned, they are busy.. so telco sends the call down... but the 
equipment doesn't take it.


I would *think* the Telco could keep trying channels down the hunt 
group, but maybe not?  We have, in the past, seen this issue with our 
dial-up modem banks.. especially if I would take one offline.   
However, it is not a big enough issue (i.e. we don't take things down 
that often) for me to look into it fully.


On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly <[EMAIL PROTECTED] 
> wrote:


I think the problem is that the telco presents the call on a
specific channel, then zaptel tells it that the channel is busy.

 


We need to be able to tell the telco that each unused channel on
a given span is unavailable, and it will determine that the
others are in use and will present the call on a channel on
another span.

 


A rather ugly work-around (since Andrew seems to have lots of
channels available, and one would assume that maintenance of this
nature would occur during slow periods) would be to make calls to
a DID in the same trunk group on all "idle" channels on the span
shutting down then, when all channels on the span are "in use"
and none of them are doing anything useful, take the span down
hard so the telco will divert all calls to another span.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax



*From:* [EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]
] *On Behalf Of *Matt
*Sent:* Thursday, February 14, 2008 2:28 PM

*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ISDN PRIs and taking a server
down formaintenance - blocking issue

 


Honestly.. this sounds like a telco issue.I understand what
the other person is saying about the PRI still being technically
up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco
should be forwarding the call to the next available channel,
which they clearly are not doing.

On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith
<[EMAIL PROTECTED] > wrote:

Hi Tim,

Imagine the scenario where we had 10x Asterisk servers, with
calls presenting sequentially starting from the first server,
then server two, etc.

 


If we took down the first server for maintenance with 'asterisk
-rx stop gracefully' we then will block all incoming calls to all
servers as our telco will simply relay the BUSY back to the
caller. If there are a number of calls on the first server that
continue for another 20 minutes, then all inbounds are blocked
for that period of time.

 


We are finding at present we have to look at the calls on the
server and make a decision if we are busy to simply reboot the
server and hence lose calls. Not ideal bu

Re: [asterisk-users] SNMP monitoring

[Ricardo Carvalho]

Maybe you'r right and newt isn't really necessary. I just read somewhere
that those dependencies were needed, I've installed them and it worked...
Try to only install the other ones and if res_snmp gets compiled without it,
great!

Regards,
Ricardo Carvalho.




On Fri, Feb 15, 2008 at 12:01 AM, Darrick Hartman (lists) <
[EMAIL PROTECTED]> wrote:

> Ricardo Carvalho wrote:
> > I had the same problem some time ago...
> > You got to install also this packages:
> >
> > net-snmp-devel
> > newt-devel
> > lm_sensors-devel
> > bzip2-devel
> >
> > That should do it!
>
> Why would this depend on newt?  net-snmp and lm-sensor headers and
> libraries make sense.  newt doesn't make any sense as a dependency.
>
> Darrick
> --
> Darrick Hartman
> DJH Solutions, LLC
> http://www.djhsolutions.com
>
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Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue

[Don Kelly]

Andrew wants to take the system down softly-there are active calls on some
channels. He doesn't want to accept additional calls on the idle channels.
He can't take the D channel down without disruption to the active calls.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax



  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Thursday, February 14, 2008 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN PRIs and taking a server down
formaintenance - blocking issue

 

If you take Asterisk down, the PRI should go down as the D channel is down.
Then the telco should KNOW that there is trouble with the PRI and those
channels are in trouble busy and not availible.  If the telco still tries to
push a call to a channel on a PRI that is down, then the telco is at fault.

Lyle

Matt wrote: 

That does sound like what is happening.. Telco knows channel 1-23 are not
busy (so far as they are concerned), however.. so far as you are concerned,
they are busy.. so telco sends the call down... but the equipment doesn't
take it.

I would *think* the Telco could keep trying channels down the hunt group,
but maybe not?  We have, in the past, seen this issue with our dial-up modem
banks.. especially if I would take one offline.   However, it is not a big
enough issue (i.e. we don't take things down that often) for me to look into
it fully.

On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly <[EMAIL PROTECTED]> wrote:

I think the problem is that the telco presents the call on a specific
channel, then zaptel tells it that the channel is busy.

 

We need to be able to tell the telco that each unused channel on a given
span is unavailable, and it will determine that the others are in use and
will present the call on a channel on another span.

 

A rather ugly work-around (since Andrew seems to have lots of channels
available, and one would assume that maintenance of this nature would occur
during slow periods) would be to make calls to a DID in the same trunk group
on all "idle" channels on the span shutting down then, when all channels on
the span are "in use" and none of them are doing anything useful, take the
span down hard so the telco will divert all calls to another span.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, February 14, 2008 2:28 PM 


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] ISDN PRIs and taking a server down
formaintenance - blocking issue

 

Honestly.. this sounds like a telco issue.I understand what the other
person is saying about the PRI still being technically up... BUT... if the
channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to
the next available channel, which they clearly are not doing.

On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith <[EMAIL PROTECTED]> wrote:

Hi Tim,

Imagine the scenario where we had 10x Asterisk servers, with calls
presenting sequentially starting from the first server, then server two,
etc.

 

If we took down the first server for maintenance with 'asterisk -rx stop
gracefully' we then will block all incoming calls to all servers as our
telco will simply relay the BUSY back to the caller. If there are a number
of calls on the first server that continue for another 20 minutes, then all
inbounds are blocked for that period of time.

 

We are finding at present we have to look at the calls on the server and
make a decision if we are busy to simply reboot the server and hence lose
calls. Not ideal but then we don't end up blocking our inbounds.

What I was hoping to do was find a way to cause the telco to present the
call to the next ISDN30 and therefore would allow us to cleanly take down an
Asterisk server for maintenance without causing this issue. In a sense to
put the ISDN30 into alarm mode while still continuing the active calls.

 

Do you know if this is at all possible, even if we considered patching
zaptel to add this functionality or does the telco rely on the entire PRI
being in alarm before it presents the call to the next ISDN30 ? This would
allow us to run maintenance on our servers during busy periods without
causing disruption, and would be an excellent feature.

Many thanks,

Andrew

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Nelson
Sent: 13 February 2008 18:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ISDN PRIs and taking a server down for
maintenance - blocking issue

Even if * is shutdown, zaptel is still running and your ISDN channels are
still technically up. Shutting down zaptel should close the channels and put
those circuits into alarm mode.

Tim Nel

Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9

[C F]

On Thu, Feb 14, 2008 at 10:12 AM, Henry Devito <[EMAIL PROTECTED]> wrote:
> I had GXP-2000's running on 1.0 versions of asterisk even earlier.  So I
>  know it does work.  I upgraded one of my customers GXP's to the latest

I'm not sure you are right, since I have had Polycoms that didn't
work, it's quite possible you should have GPXs that do work.

>  firmware in it still works.  Can you post the output of the CLI with verbose
>  set to 99 and the the output from the asterisk log file that has the call in
>  it.  You can usually do a 'tail /var/log/asterisk/full -n 400' right after
>  the call fails.
>
>  I will be glad to help, just need a little more info to narrow down the
>  issue.
>
>  Thanks
>  Henry
>
>
>  - Original Message -
>  From: "Giordano Grandis" <[EMAIL PROTECTED]>
>  To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>  
>  Sent: Thursday, February 14, 2008 2:15 AM
>  Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9
>
>
>  1. The phone has not the DND active, i checked it several times
>  2. Outbound calls always success, the problem is when the phone receive a
>  call, it repsnds with busy signalling.
>  3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade
>  asterisk.
>
>  Thanks for all
>
>  -Messaggio originale-
>  Da: [EMAIL PROTECTED]
>  [mailto:[EMAIL PROTECTED] Per conto di C F
>  Inviato: mercoledì 13 febbraio 2008 21.09
>  A: Asterisk Users Mailing List - Non-Commercial Discussion
>  Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9
>
>  Just check DND if it's on on the phone or not.
>  What is the CLI output when you try making a phone call?
>  Why don't you try it with a later version of astrisk and a Phone?
>
>  On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote:
>  >
>  >
>  > Hi all gusy,
>  > i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a
>  > few
>  > go in "busy" state, if you call it get the busy tone but the phone can
>  > male
>  > any type of call.
>  > This is my sip.conf
>  >
>  > [502]
>  > language = it
>  > username = 502
>  > secret = 
>  > host = dynamic
>  > type = friend
>  > context = local
>  > canreinvite = yes
>  > dtmfmode = info
>  > callgroup = 1
>  > pickupgroup = 1
>  > callerid = 502 <502>
>  >
>  > Under Grandstream's support suggest, I set "Use randmom port" to yes and
>  > "Nat traversal (STUN)" to "No, but send keep alive" but without success.
>  > This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6
>  >
>  > Anyone can help me ?
>  >
>  > Thanks in advance
>  >
>  > Giordano
>  >
>  >
>  > No virus found in this outgoing message.
>  >  Checked by AVG Free Edition.
>  >  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date:
>  > 12/02/2008
>  > 15.20
>  >
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>  No virus found in this incoming message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008
>  15.20
>
>
>  No virus found in this outgoing message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008
>  20.00
>
>
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[asterisk-users] DialPlan help with Analog Fax Machine

[Jim Duda]

I'm struggling to get my dialplan to work with a simple analog fax 
machine.

I have TDM400B zaptel card with an FXO and FXS port.  I have the FXO 
port connected to the POTS machine and the FAX machine connected to the 
FXS port.

The FAX machine itself works fine, I can FAX outgoing messages fine.  I 
can also dial the FAX extension from the internal context, the FAX 
machine answers and I hear the FAX tones.

I'm struggling to get the fax detection to work, causing a transfer to 
the FAX machine.  I think the fax transfer starts, but for some reason 
the dialplan "falls through" and the connection is dropped immediately.

This should be so simple ...

version:

asterisk*CLI> core show version
Asterisk 1.6.0-beta2 built by jduda @ asterisk on a i686 running Linux 
on 2008-02-03 03:23:54 UTC

zapata.conf has:

; FAX machine connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
faxdetect=no
signalling=fxo_ks
context=internal
channel => 1

; PSTN connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
mwimonitor=yes
;mwilevel=512
mwimonitornotify=/usr/local/sbin/zapnotify.sh
faxdetect=incoming
signalling=fxs_ks
context=incoming
channel => 4

extensions.conf has:

[fax-hardware]
exten => s,1,StopPlaytones
exten => s,2,Dial(ZAP/1,40,tr)
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1)

[incoming]
exten => fax,1,Goto(fax-hardware,s,1)
exten => s,1,Goto(incoming-dial,s,1)
exten => ,1,Goto(incoming-dial,s,1)

[incoming-dial]
exten => s,1,Zapateller(nocallerid)
exten => s,2,SetMusicOnHold(icecast)
exten => 
s,3,GotoIf(${DB_EXISTS(blacklist/${CALLERID(number)})}?custom-blacklisted,s,1)
exten => s,4,Set(DB(CALLTRACE/lastcaller)=${CALLERID(number)})
exten => s,5,AGI(MisterHouse.agi,"CallerID")
exten => s,6,Answer
exten => s,7,Playtones(ring)
exten => 
s,8,Dial(${PHONES0}&${PHONES1}&${PHONES2}&${PHONES7}&${PHONES11},20,tr)
exten => s,9,Goto(s-${DIALSTATUS},1) ; if no fax, branch on dialstatus
exten => s-NOANSWER,1,Macro(voicemail,${PHONES0VM})
exten => s-NOANSWER,2,Hangup()
exten => s-BUSY,1,Macro(voicemail,${PHONES0VM})
exten => s-BUSY,2,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1) ; everything else is treated as no 
answer
exten => s,105,Goto(5)

exten => fax,1,Goto(fax-hardware,s,1)

When the FAX call comes in, I get this:

[Feb 14 20:01:03] NOTICE[1826]: chan_zap.c:7306 mwi_thread: Got event 18 
(Ring Begin)...  Passing along to ss_thread
 -- Starting simple switch on 'Zap/4-1'
[Feb 14 20:01:04] NOTICE[1826]: chan_zap.c:7066 ss_thread: Got event 2 
(Ring/Answered)...
 -- Executing [EMAIL PROTECTED]:1] Goto("Zap/4-1", "incoming-dial,s,1") in 
new stack
 -- Goto (incoming-dial,s,1)
 -- Executing [EMAIL PROTECTED]:1] Zapateller("Zap/4-1", 
"nocallerid") in new stack
 -- Executing [EMAIL PROTECTED]:2] SetMusicOnHold("Zap/4-1", 
"icecast") in new stack
 -- Executing [EMAIL PROTECTED]:3] GotoIf("Zap/4-1", 
"0?custom-blacklisted,s,1") in new stack
 -- Executing [EMAIL PROTECTED]:4] Set("Zap/4-1", 
"DB(CALLTRACE/lastcaller)=8884732963") in new stack
 -- Executing [EMAIL PROTECTED]:5] AGI("Zap/4-1", 
"MisterHouse.agi,"CallerID"") in new stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/MisterHouse.agi
   MisterHouse.agi,"CallerID": AGI Environment Dump:
   MisterHouse.agi,"CallerID":  -- accountcode =
   MisterHouse.agi,"CallerID":  -- arg_1 = CallerID
   MisterHouse.agi,"CallerID":  -- callerid = 8884732963
   MisterHouse.agi,"CallerID":  -- calleridname = UNAVAILABLE
   MisterHouse.agi,"CallerID":  -- callingani2 = 0
   MisterHouse.agi,"CallerID":  -- callingpres = 0
   MisterHouse.agi,"CallerID":  -- callingtns = 0
   MisterHouse.agi,"CallerID":  -- callington = 0
   MisterHouse.agi,"CallerID":  -- channel = Zap/4-1
   MisterHouse.agi,"CallerID":  -- context = incoming-dial
   MisterHouse.agi,"CallerID":  -- dnid = unknown
   MisterHouse.agi,"CallerID":  -- enhanced = 0.0
   MisterHouse.agi,"CallerID":  -- extension = s
   MisterHouse.agi,"CallerID":  -- language = en
   MisterHouse.agi,"CallerID":  -- priority = 5
   MisterHouse.agi,"CallerID":  -- rdnis = unknown
   MisterHouse.agi,"CallerID":  -- request = MisterHouse.agi
   MisterHouse.agi,"CallerID":  -- threadid = -1232077936
   MisterHouse.agi,"CallerID":  -- type = Zap
   MisterHouse.agi,"CallerID":  -- uniqueid = 1203037263.20
   MisterHouse.agi,"CallerID":  -- version = 1.6.0-beta2
   MisterHouse.agi,"CallerID": here CallerID
   MisterHouse.agi,"CallerID": CallerID: 8884732963 Line: Zap/4-1
 -- AGI Script MisterHouse.agi completed, returning 0
 -- Executing [EMAIL PROTECTED]:6] Answer("Zap/4-1", "") in new stack
 -- Executing [EMAIL PROTECTED]:7] PlayTones("Zap/4-1", "ring") in 
new stack
 -- Executing [EMAIL PROTECTED]:8] Dial("Zap/4-1", 
"SIP/100&SIP/101&SIP/102&SIP/107&SIP/111,20,tr") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
   == Using UDPTL TOS bits 184
   == Using

[asterisk-users] 57iCT BLF problem

[Paul Hales]

We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and
the BLF indicators no longer function. 

Has anyone had a similar issue? And a solution?

PaulH



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[asterisk-users] HPEC

[Al lists]

Just wondering how your experience is with HPEC,
Is it just for analog interfaces or we can use it on TE122 as well?
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[asterisk-users] patch which makes Asterisk-Addons 1.4.5 work when codec negotiation patch applied to asterisk

[Ganbold Tsagaankhuu]

Hi,

Since the original codec negotiation patch (
http://bugs.digium.com/view.php?id=4825 report) just closed yesterday,
and as well as my report (http://bugs.digium.com/view.php?id=11998), I had
nothing to do but send my patches to the list.
It might be good if my patches are placed at
http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch, but
don't know whom should I contact.
Anyway sending here.

thanks,

Ganbold


chan_h323.c.patch1
Description: Binary data


ooh323cDriver.c.patch
Description: Binary data
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Re: [asterisk-users] restart asterisk daily

[Al lists]

Always rely on free -m to see how much free memory you have not top.
in terms of memory leak, i have asterisk running on servers with uptime of
400 days (CentOs), if there was any leak, i'm guessing i would have crashed
server long time ago.

On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey <[EMAIL PROTECTED]> wrote:

> If you want to flush your disk cache to see how much memory is being eaten
> cache pages, try this:
>  echo 3 >/proc/sys/vm/drop_caches
>
> - "ast erisk" <[EMAIL PROTECTED]> wrote:
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Re: [asterisk-users] restart asterisk daily

[Erik Anderson]

On Thu, Feb 14, 2008 at 8:38 PM, Al lists <[EMAIL PROTECTED]> wrote:
> Always rely on free -m to see how much free memory you have not top.

You could install and use "htop" - it's a much more functional (and
informative) version of top.  It shows the difference between
shared/buffer/cache memory.

-erik

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Re: [asterisk-users] restart asterisk daily

[Anthony Francis]

Al lists wrote:
> Always rely on free -m to see how much free memory you have not top.
> in terms of memory leak, i have asterisk running on servers with 
> uptime of 400 days (CentOs), if there was any leak, i'm guessing i 
> would have crashed server long time ago.
>
> On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey <[EMAIL PROTECTED] 
> > wrote:
>
> If you want to flush your disk cache to see how much memory is
> being eaten cache pages, try this:
>  echo 3 >/proc/sys/vm/drop_caches
>
> - "ast erisk" <[EMAIL PROTECTED]
> > wrote:
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> 
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You will see asterisk behave its worst with multiple queues and heavy 
dialplan logic. I restart my boxes with queues everynight at midnight 
just to reset the queue stats displayed with show queue.

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Re: [asterisk-users] restart asterisk daily

[Tzafrir Cohen]

On Thu, Feb 14, 2008 at 09:32:04PM -0600, Erik Anderson wrote:
> On Thu, Feb 14, 2008 at 8:38 PM, Al lists <[EMAIL PROTECTED]> wrote:
> > Always rely on free -m to see how much free memory you have not top.
> 
> You could install and use "htop" - it's a much more functional (and
> informative) version of top.  It shows the difference between
> shared/buffer/cache memory.

It also consumes more CPU.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] restart asterisk daily

[Erik Anderson]

On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
>  It also consumes more CPU.

True, a fraction more.  If you have that little overhead on your
server, though, that this would cause a problem, you probably should
upgrade your hardware, IMHO.

-eriik

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Re: [asterisk-users] What is a "secure call"?

[Olivier]

2008/2/13, Johansson Olle E <[EMAIL PROTECTED]>:
>
>
> In SIP, there's a specification for how I as a domain owner can
> request all calls to my domain to use
> SIP/TLS by using DNS NAPTR and SRV records.


Which one ?
Does it also deal with SPIT ?

But how do I as a caller
> request a secure service?


I  think SPIT is a major concern (though I've not heard a single case of
SPIT abuse, yet).

How do we place a secure call with DIAL? Do we need SECUREDIAL?
>
> Any ideas and thoughts on the subject are welcome!
>
> Regards,
> /Olle
>
>
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Re: [asterisk-users] Realtime SIP peers - reloading cached info

[Olivier]

2008/2/13, Atis Lezdins <[EMAIL PROTECTED]>:
>
> On 2/13/08, Rob Hillis <[EMAIL PROTECTED]> wrote:
> >
> >  -BEGIN PGP SIGNED MESSAGE-
> >  Hash: SHA1
> >
>
> >  If it is being removed in 1.6, I'm a little concerned since there's no
> > mention of this when you show the application, nor on voip-info.org
> .  What
> > application/function is it being replaced by?
>
>
> There's an obsolete warning in 1.4.18, but i somehow remember that
> it's obsolete already since some 1.4.11
>
> It's func_realtime as i said before. usage shouldn't be much
> different, you can replace with:
>
> Set(REALTIME(sip_buddies,name,100,my_field)=foo);
>
> Also, seems that func_realtime will soon support SQL INSERT's and DELETE's
> :)


Do you mean it should be added between today's beta and future GA 1.6 ?

Regards,
>
> Atis
>
>
>
> >
> >  Atis Lezdins wrote:
> >  | On 2/13/08, Rob Hillis <[EMAIL PROTECTED]> wrote:
> >  |> -BEGIN PGP SIGNED MESSAGE-
> >  |> Hash: SHA1
> >  |>
> >  |> Atis Lezdins wrote:
> >  |> | By RealTimeUpdate do you mean func_realtime? It shouldn't care, as
> >  |> | cache is not implemented in realtime level, but higher (chan_sip).
> >  |> |
> >  |> | Are you sure you need "sip show XXX load". If you "sip prune" peer
> >  |> | data, it should be re-loaded on next access.
> >  |> |
> >  |> | What i was suggesting - to dig into chan_sip and create dialplan
> >  |> | application SipPrune(peer) that would prune the peer directly, by
> >  |> | using corresponding function - sip_prune_peer() in chan_sip.c -
> that
> >  |> | way you will gain some extra performance, as there's no
> manager/cli
> >  |> | overhead.
> >  |> |
> >  |> | However if you're uncomfortable with C, the app_system shouldn't
> cause
> >  |> | any troubles :)
> >  |>
> >  |> RealTimeUpdate is more likely to correspond to app_realtime rather
> than
> >  |> func_realtime.
> >  |
> >  | As to my knowledge - that is obsolete and being removed in 1.6,
> >  | func_realtime replaces it. That's why i wondered about name -  I just
> >  | never happened to use it :)
> >  |
> >  | Regards,
> >  | Atis
> >  |
> >  |
> >
> >  -BEGIN PGP SIGNATURE-
> >  Version: GnuPG v1.4.7 (GNU/Linux)
> >  Comment: Using GnuPG with Remi - http://enigmail.mozdev.org
> >
> > iD8DBQFHsnaM6uKn5cBSgGQRAo/TAKDCruPrn2nm2XV/PYbfSuBKA0j5OwCfQ/Ox
> >  QE3SYEmZ01QHUT4ITwmLnT0=
> >  =SKEW
> >  -END PGP SIGNATURE-
> >
> >
>
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>
>
> --
> Atis Lezdins
> VoIP Developer,
> IQ Labs Inc.
> [EMAIL PROTECTED]
> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Work phone: +1 800 7502835
>
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[asterisk-users] How to check if a local channel member of a queue?

[Rajkumar S]

Hi,

I am using asterisk-1.4.15

I have a queue with one agent added using AddQueueMember
(FAO|Local/[EMAIL PROTECTED]|0||Agent/602).

Once this command executes queue show FAO shows:

FAO  has 0 calls (max unlimited) in 'roundrobin' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 60s
   Members:
  Agent/602 (dynamic) (Not in use) has taken no calls yet

There is no mention of the fact that which channel is used by
Agent/602. I am adding agents to queue both from an agi using
AddQueueMember, via phone,  and manager command QueueAdd via web. In
both cases I needs to find out whether the given sip channel has
logged in to any queue previously, for proper validation. show agents
command in previous version gave such details.

How can I get the details given out by show agents in the new 1.4
scheme of things?

With much regards,

raj

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[asterisk-users] Asterisk DNS SIP issue

[Kevin Kiely]

The other day my asterisk local SIP clients got hung when my provider had a
DNS failure.  All registrations went dead (even the ones that were IP
addresses) and all sip peers went offline.  I know this was know problem at
one point is there any update on this when using a FQDN for one of the peer
addresses in sip.conf?
 
 
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