[asterisk-users] Blind Transfer is not working in incoming calls

[Max Alex]

Hi Everybody,
i have installed asterisk 1.4.19 on my box,
I have setup agi script which is used while incoming and outgoing calls.
It will find the users for incoming and calls to them which is registered in
asterisk,
I have a setup *# for blind transfer to call any outbound or inbound
numbers.
when i am calling any outbound call and the calls are connected with my sip
peer, then i am pressing *# for blind transfer, it will ask me to enter the
transfer number and it is working,
But when an incoming call to my sip user and they are connected the *# is
not worked even the transfer prompt is also played, and dtmf is also set
properly.

But i am not getting why the incoming call is not transfer to any other
number?
Please help for this issue!


-- 
Thanks,
Max Alex
Voip Developer
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[asterisk-users] Voicemail has issues with DTMF

[Max Alex]

Hi everybody,
I have linksys phone at my location,
i am using asterisk version 1.4.19,
I have a issue regarding dtmf mode, i have set the Asterisk DTMF mode to
Auto in order to eliminate Asterisk effect on the DTMF transmission. Both
Inband and AVT from Linksys worked with PSTN IVR.
But, We have the issue why Asterisk Voicemail doesn't work with Linksys set
to Inband and Asterisk set to Auto.
And what is the reply of asterisk while the dtmf configuration like this?
Anyone please help me for this issue, i have searched many pages but i
haven't found the exact solution or reason for this?


-- 
Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] frequent channel reset problem

[Ming-Ching Tiew]

--- On Fri, 8/22/08, Ming-Ching Tiew <[EMAIL PROTECTED]> wrote:

> From: Ming-Ching Tiew <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] frequent channel reset problem
> To: asterisk-users@lists.digium.com
> Date: Friday, August 22, 2008, 11:00 PM
> Sorry to have posted reply to your email directly.
> Repost to maillist.
> 
> resetinterval already set to never to start with.
> 
> Thanks
> 
> 
> --- On Fri, 8/22/08, Ming-Ching Tiew
> <[EMAIL PROTECTED]> wrote:
> 
> > From: Ming-Ching Tiew <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Fw: [asterisk-dev]
> frequent channel reset problem
> > To: "Steve Totaro"
> <[EMAIL PROTECTED]>
> > Date: Friday, August 22, 2008, 10:58 PM
> > --- On Fri, 8/22/08, Steve Totaro
> > <[EMAIL PROTECTED]> wrote:
> > 
> > > From: Steve Totaro
> > <[EMAIL PROTECTED]>
> > 
> > > 
> > > Try setting resetinterval to never.
> > > 
> > > resetinterval: sets the time in seconds between
> > restart of
> > > unused
> > > channels, defaults to
> > > 3600 minimum 60 seconds. Some PBXs don't like
> > channel
> > > restarts. so set
> > > the interval to a
> > > very long interval e.g. 1 or
> 'never'
> > to
> > > disable *entirely*.
> > > 
> > 
> > Already did that, this is my zapata.conf :-
> > 
> > [channels]
> > switchtype=euroisdn
> > pridialplan=national
> > resetinterval=never
> > 
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > cancallforward=yes
> > echocancel=yes
> > 
> > rxgain=0
> > txgain=0
> > 
> > 
> > echocancel=yes
> > echocancelwhenbridged=yes
> > 
> > ;echocancel=no
> > ;echocancelwhenbridged=no
> > 
> > ;echotraining=no
> > 
> > ;faxdetect=both
> > ;faxdetect=incoming
> > ;faxdetect=outgoing
> > ;faxdetect=yes
> > faxdetect=no
> > 
> > group=1
> > ;signalling=pri_cpe
> > signalling=pri_net
> > context=incoming
> > channel => 1-15,17-31
> > 
> > jbenable=yes
> > jbforce=yes
> > jbmaxsize=200
> > 
> > 
> > ;jbimpl=adaptive
> > ;jbimpl=fixed
> > ;jbresyncthreshold = 4000
> > 
> > group=2
> > signalling=pri_cpe
> > context=outgoing
> > channel => 32-46,48-62
> > 
> > ;jbenable=yes
> > ;jbforce=yes
> > ;jbmaxsize=200
> > 
> > 
> > And my zaptel.conf :-
> > span=1,0,0,ccs,hdb3,crc4
> > span=2,1,0,ccs,hdb3,crc4
> > 
> > #Span 1 Channel Definition 
> > bchan=1-15,17-31
> > #hardhdlc=16
> > dchan=16
> > 
> > 
> > #Span 2 Channel Definition
> > bchan=32-46,48-62
> > #hardhdlc=47
> > dchan=47
> > 
> > loadzone = uk
> > defaultzone = uk
> 

When I re-read my post, there is a danger that people thought that 
resetinterval fixes my problem. That's not what I meant.

What I meant to say was that, the resetinterval was all the time 
set as 'never' to begin with and the frequent channel reset problem 
was observed with 'resetinterval' set as 'never'. 

What I really need is someone who knows how to read the pri debug 
at http://www.geocities.com/mctiew/asterisk/bd2.txt, and perhaps 
can tell me what wrong with the said PRI line giving this problem 
***ONLY*** to asterisk but not giving any problem to other PBXes
while the same asterisk settings I have used was working perfectly 
on another PRI from the same network provider.

Perhaps someone could tell me how to increase the "tolerance" level
of asterisk towards PRI lines with lower quality.

Regards.







  

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Re: [asterisk-users] Problem with modem data calls and xorcom astribanks

[Greg Woods]

Darren Sessions wrote:
> Not sure what you've heard before, but I have successfully used a 
> modem at 9600 baud

Well, OK, "it won't work" was a little strong. Faxes work because they 
too are at slow speed. But for me at  least, 9600 baud is pretty much 
useless. Instead, I just patch the modem directly into the wall plate 
(bypassing the asterisk box) on the rare occasions that I need to dial 
up so that I can get the 56k. Works in my small home setup, won't work 
for everyone. I suspect that neither patching around asterisk nor 
settling for 9600 is going to be acceptable for the motel manager.

--Greg



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Re: [asterisk-users] set callerid with plus sign

[ronald]

Hi Thanks for all your reply.

Just figured out that ISUP does not decode plus sign very well.

regards
nhadie

Eric "ManxPower" Wieling wrote:
> + is not a valid Caller*ID character.  Asterisk allows you to use + in 
> Caller*ID, but many carriers will reject the call if you do that.
> 
> Benny Amorsen wrote:
>> ronald <[EMAIL PROTECTED]> writes:
>>
>>> Is it possible to assign a plus sign on the callerid(num) ?
>> Yes.
>>
>>> currently this is what i do CALLERID(num)=+6523450017
>>>
>>> but telco is denying calls, coz they said they are seeing "bs523450017" 
>>> instead of +6523450017.
>> Which techology? SIP? PRI? POTS? ...?
>>
>>
>> /Benny
>>
>>
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> 

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Thursday 21 August 2008 10:08:53 J.M. wrote:
> I am running Asterisk 1.4.21.2 with Realtime.  I have a phone setup in the
> database and when I connect that phone to Asterisk there are suddenly an
> endless number of "SELECT * FROM sip WHERE name = '1001' AND host =
> 'dynamic'" queries being run.  The only way to stop the flood of queries
> coming from Asterisk to restart the Asterisk process.  Even disconnecting
> the phone doesn't stop Asterisk from running the queries.
>
> Has anyone seen this before?  Why would Asterisk do that and does anyone
> know the fix?

Asterisk does that because realtime data is not cached by default, so for each
access of the peer in question, Asterisk needs to reload the data on the peer
from the database.  If you'd like, turn on rtcachefriends in sip.conf, which
will cache the peer for the duration of the registration interval (or whatever
you have rtexpire set to).  Also, to get correct behavior on reload, you'll
need to have rtupdate turned on.  Some of the behavior isn't quite right in
1.4.21.2, even, but it should be fixed once 1.4.22 is released.

BTW, I would otherwise have responded sooner, but I am on vacation this week,
and I am not responding to email as quickly as I would usually.

-- 
Tilghman

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[asterisk-users] ztd-ethmf

[Bill Michaelson]

I expected to find th module ztd-ethmf[.c...] in support of the redfone 
TDMoE product in my zaptel distro (I have 1.4.11).  But it's not there.  
I am awaiting a response to a trouble ticket from redfone.  Can anyone 
give me a jumpstart?  I can't seem to google this up.


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Re: [asterisk-users] frequent channel reset problem

[Ming-Ching Tiew]

Sorry to have posted reply to your email directly.
Repost to maillist.

resetinterval already set to never to start with.

Thanks


--- On Fri, 8/22/08, Ming-Ching Tiew <[EMAIL PROTECTED]> wrote:

> From: Ming-Ching Tiew <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Fw: [asterisk-dev] frequent channel reset 
> problem
> To: "Steve Totaro" <[EMAIL PROTECTED]>
> Date: Friday, August 22, 2008, 10:58 PM
> --- On Fri, 8/22/08, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
> 
> > From: Steve Totaro
> <[EMAIL PROTECTED]>
> 
> > 
> > Try setting resetinterval to never.
> > 
> > resetinterval: sets the time in seconds between
> restart of
> > unused
> > channels, defaults to
> > 3600 minimum 60 seconds. Some PBXs don't like
> channel
> > restarts. so set
> > the interval to a
> > very long interval e.g. 1 or 'never'
> to
> > disable *entirely*.
> > 
> 
> Already did that, this is my zapata.conf :-
> 
> [channels]
> switchtype=euroisdn
> pridialplan=national
> resetinterval=never
> 
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> echocancel=yes
> 
> rxgain=0
> txgain=0
> 
> 
> echocancel=yes
> echocancelwhenbridged=yes
> 
> ;echocancel=no
> ;echocancelwhenbridged=no
> 
> ;echotraining=no
> 
> ;faxdetect=both
> ;faxdetect=incoming
> ;faxdetect=outgoing
> ;faxdetect=yes
> faxdetect=no
> 
> group=1
> ;signalling=pri_cpe
> signalling=pri_net
> context=incoming
> channel => 1-15,17-31
> 
> jbenable=yes
> jbforce=yes
> jbmaxsize=200
> 
> 
> ;jbimpl=adaptive
> ;jbimpl=fixed
> ;jbresyncthreshold = 4000
> 
> group=2
> signalling=pri_cpe
> context=outgoing
> channel => 32-46,48-62
> 
> ;jbenable=yes
> ;jbforce=yes
> ;jbmaxsize=200
> 
> 
> And my zaptel.conf :-
> span=1,0,0,ccs,hdb3,crc4
> span=2,1,0,ccs,hdb3,crc4
> 
> #Span 1 Channel Definition 
> bchan=1-15,17-31
> #hardhdlc=16
> dchan=16
> 
> 
> #Span 2 Channel Definition
> bchan=32-46,48-62
> #hardhdlc=47
> dchan=47
> 
> loadzone = uk
> defaultzone = uk


  

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Re: [asterisk-users] Fw: [asterisk-dev] frequent channel reset problem

[Steve Totaro]

>> From: Ming-Ching Tiew <[EMAIL PROTECTED]>
>> Subject: [asterisk-dev] frequent channel reset problem
>> To: [EMAIL PROTECTED]
>> Date: Friday, August 22, 2008, 3:04 PM
>> Hi,
>>
>> I am stucked with a nasty PRI problem for 2 weeks now and
>> will appreciate if I could get some help from here. The
>> problem is that my zaptel.conf and zapata.conf have been
>> working with one PRI line but it is not working with
>> another
>> PRI. I could easily blame it to the PRI line quality
>> itself, but the fact is that the PRI line has been working
>> fine with at least 2 PBXes for ages without experiencing
>> channel
>> reset or call drop problem. However when connected to
>> asterisk using wct4xxp driver, it will get call drop
>> randomly
>> every now and then.
>>
>> From the asterisk console output, when calls are dropped,
>> there is associated alarm detected.
>>
>> I would like to know the following:
>>
>> (1) After detecting an alarm, does the ISDN spec. specifies
>> that all channels must be resetted or this is just an
>> implementation choice which asterisk has made ?
>>
>> Should asterisk reset the line after detecting the
>> alarm
>> and caused all calls being drop?
>>
>> (2) I have a pri monitor ( ISDN tester) that sniffs the PRI
>> line, I found that whenever the dropping calls problem
>> happens,  the monitor print out "RDI Begin ...
>> SAMBE
>>,RESTART, RESTART ACK... RDI end".
>>
>> Does anyone know what is RDI other than what stated in the
>> monitor's manual as "Remote Defect Detect"
>> and
>> what causes it.
>>
>> Attached is the  PRI intense debug file.
>> ( http://www.geocities.com/mctiew/asterisk/bd2.txt )
>>
>> Any pointer or help given is greatly appreciated !
>>
>>

Try setting resetinterval to never.

resetinterval: sets the time in seconds between restart of unused
channels, defaults to
3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set
the interval to a
very long interval e.g. 1 or 'never' to disable *entirely*.

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf

Thanks,
Steve Totaro

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[asterisk-users] Fw: [asterisk-dev] frequent channel reset problem

[Ming-Ching Tiew]

Sorry to have posted to the wrong maillist. Repost here.

Regards.

--- On Fri, 8/22/08, Ming-Ching Tiew <[EMAIL PROTECTED]> wrote:

> From: Ming-Ching Tiew <[EMAIL PROTECTED]>
> Subject: [asterisk-dev] frequent channel reset problem
> To: [EMAIL PROTECTED]
> Date: Friday, August 22, 2008, 3:04 PM
> Hi,
>  
> I am stucked with a nasty PRI problem for 2 weeks now and
> will appreciate if I could get some help from here. The
> problem is that my zaptel.conf and zapata.conf have been
> working with one PRI line but it is not working with
> another
> PRI. I could easily blame it to the PRI line quality
> itself, but the fact is that the PRI line has been working
> fine with at least 2 PBXes for ages without experiencing
> channel
> reset or call drop problem. However when connected to 
> asterisk using wct4xxp driver, it will get call drop
> randomly
> every now and then. 
> 
> From the asterisk console output, when calls are dropped, 
> there is associated alarm detected.
>  
> I would like to know the following:
>   
> (1) After detecting an alarm, does the ISDN spec. specifies
> that all channels must be resetted or this is just an
> implementation choice which asterisk has made ?
>  
> Should asterisk reset the line after detecting the
> alarm
> and caused all calls being drop? 
>   
> (2) I have a pri monitor ( ISDN tester) that sniffs the PRI
> line, I found that whenever the dropping calls problem
> happens,  the monitor print out "RDI Begin ... 
> SAMBE
>,RESTART, RESTART ACK... RDI end".
>
> Does anyone know what is RDI other than what stated in the
> monitor's manual as "Remote Defect Detect"
> and
> what causes it.
>  
> Attached is the  PRI intense debug file.
> ( http://www.geocities.com/mctiew/asterisk/bd2.txt )
>   
> Any pointer or help given is greatly appreciated !
> 
> 



  

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Re: [asterisk-users] Inefficient Codec Translation

[Brent Davidson]

Steve Totaro wrote:

On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin <[EMAIL PROTECTED]> wrote:
  

We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.

Here is the scenario

[DID Vendor] ---> [Asterisk ]
> External GW [G729]
 |

|---> External GW [iLBC]

Our DID vendor and asterisk box supports both ilbc & g729. However,
our external gateway termination supports either ilbc or g729 (and not
both) and depending on users location, we terminate it on either
gateway.

Since DID and asterisk box supports both the codecs, we assumed that
asterisk will appropriately select codecs depending on where we
terminate the call so that no codec translation happens. However, this
seems to be an incorrect assumption and we see that different codecs
get selected on two legs which leads to quality drop and extra CPU
cycles.

May be we are doing something wrong. Pls suggest what we are doing
wrong. Below is asterisk configuration.

[did]
type=friend
host=xxx
canreinvite=yes
disallow=all
allow=g729
allow=ilbc

[gw1]
type=friend
host=xxx
canreinvite=yes
disallow=all
allow=g729

[gw2]
type=friend
host=xxx
canreinvite=yes
disallow=all
allow=ilbc

Thanks
Jim




Why don't you allow=g729 only on all entries.  Maybe I have misread
your email but I interpret what you wrote to mean that all endpoints
support g729
  


I may be wrong but I understood the situation as the DID supplier 
supports either g.729 or ilibc, but the user has 2 locations that calls 
are routed to.  One location supports iLibc only, the other supports 
g.729 only.  What they seem to be trying to accomplish is to get the DID 
<-> Asterisk leg to use the same codec as the Asterisk <-> Remote 
Location leg.  I think the problem is going to be that the call has to 
be established to the Asterisk box before a destination can be 
selected.  The DID and Asterisk Box are going to negotiate the first 
available common codec before doing anything else, including setting a 
destination.  Since you can't change a codec once a call has been 
established you're always going to end up with calls to one of the 2 
remote locations being transcoded.


The only solution I could think of would be if there was some way to 
identify which incoming calls were going to be routed to which location 
and set the codec accordingly.  To do that, you'd either have to have 2 
different DID's or some other massively more complicated mechanism.


Forcing a reinvite (Is that even possible?) would be the only other 
long-shot I could think of.


Good luck,
Brent

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[asterisk-users] AEL (was: Re: Changing callerID in a context)

[Philipp Kempgen]

Andy Dixon schrieb:

> whats AEL?

Asterisk Extension Language. extensions.ael

Grüße,
Philipp Kempgen
-- 
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Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] About the CALLIDNUMBER of the fxs

[Philipp Kempgen]

larry schrieb:

>Here  is a question about the fxs of the zaptel card which is set a

Didn't you post almost the exact same question yesterday?
(twice already)

Grüße,
Philipp Kempgen
-- 
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Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] queue timeout

[Fred Posner]

On Aug 22, 2008, at 2:26 PM, Giedrius Augys wrote:

Hello,

  I want to ask, how to "detect" queue timeout? If queue members are  
busy or not answering to the call, and after queue timeout caller  
would hear : "Sorry all operators are busy, please leave a record":

 This example:

[ivr]
exten => start,1,Ringing
exten => start,n,Wait(2)
exten => start,n,Answer
exten => start,n,Playback(ivr/welcome)
exten => start,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/ 
operator-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})

exten => start,n,Set(MONITOR_FILENAME=${RECORD_FILENAME})
exten => start,n,Queue(ivr|tT|||30)


exten => start,n,voicemail(ACCOUNT)


Fred Posner
www.teamforrest.com








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Re: [asterisk-users] queue timeout

[Jared Smith]

On Fri, 2008-08-22 at 21:26 +0300, Giedrius Augys wrote:
> I want to ask, how to "detect" queue timeout? If queue members are
> busy or not answering to the call, and after queue timeout caller
> would hear : "Sorry all operators are busy, please leave a record":

The Queue() application sets a channel variable named QUEUESTATUS, and
by reading that channel variable, you'll be able to evaluate why the
call left the queue (due to a timeout or other reason).

See the end of the description for the queue application in the online
help by typing "core show application queue" at the Asterisk
command-line interface.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] em wink

[Jared Smith]

On Fri, 2008-08-22 at 14:40 -0400, Jerry Geis wrote:
> Are there parameters for em wink?

A quick glance at the sample zapata.conf that comes with Asterisk shows
prewink, wink, and rxwink timing parameters.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] queue timeout

[Mark Michelson]

Giedrius Augys wrote:
> Hello,
> 
>   I want to ask, how to "detect" queue timeout? If queue members are 
> busy or not answering to the call, and after queue timeout caller would 
> hear : "Sorry all operators are busy, please leave a record":
>  This example:
> 
> [ivr]
> exten => start,1,Ringing
> exten => start,n,Wait(2)
> exten => start,n,Answer
> exten => start,n,Playback(ivr/welcome)
> exten => 
> start,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/operator-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
> exten => start,n,Set(MONITOR_FILENAME=${RECORD_FILENAME})
> exten => start,n,Queue(ivr|tT|||30)
> 
> exten => t,1,Goto(ivr,recording,1)
> 
> exten => recording,1,Playback(ivr/leave-the-message)
> exten => 
> recording,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/irasas-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
> exten => recording,n,Record(${RECORD_FILENAME}:wav||60)
> 
> 
> So if operators are busy or not answering, and after 30 sec, I want to 
> run recording. But if operators answered the call, I want just hangup 
> call. And I don't no how to do that. Maybe use Dial command instead 
> using Queue ? Or is another way
> 
> Thanks for your help.
> 
> -- 
> Pagarbiai / Best Regards,
> Giedrius Augys
> 

You can check the value of QUEUESTATUS after the call to Queue(). If the call 
timed out, then it will be set to TIMEOUT.

Mark Michelson

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[asterisk-users] em wink

[Jerry Geis]

Are there parameters for em wink?

1) timing parameters

2) dial delay or pre dial.

Thanks

Jerry

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[asterisk-users] queue timeout

[Giedrius Augys]

Hello,

  I want to ask, how to "detect" queue timeout? If queue members are busy or
not answering to the call, and after queue timeout caller would hear :
"Sorry all operators are busy, please leave a record":
 This example:

[ivr]
exten => start,1,Ringing
exten => start,n,Wait(2)
exten => start,n,Answer
exten => start,n,Playback(ivr/welcome)
exten =>
start,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/operator-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten => start,n,Set(MONITOR_FILENAME=${RECORD_FILENAME})
exten => start,n,Queue(ivr|tT|||30)

exten => t,1,Goto(ivr,recording,1)

exten => recording,1,Playback(ivr/leave-the-message)
exten =>
recording,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/irasas-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten => recording,n,Record(${RECORD_FILENAME}:wav||60)


So if operators are busy or not answering, and after 30 sec, I want to run
recording. But if operators answered the call, I want just hangup call. And
I don't no how to do that. Maybe use Dial command instead using Queue ? Or
is another way

Thanks for your help.

-- 
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] How to block incoming calls on PRI

[Dwayne Hubbard]

- "Jay R. Ashworth" <[EMAIL PROTECTED]> wrote:

> 1) can you do it "graceful"ly (both that and immediate are sometimes
> useful)?

Right now you can only disable an idle channel.


> 2) can you take down either an entire span, or a channel range on the
> command line?

This functionality will be added after the current development branches are 
merged into trunk.  I'm capping the feature creep on these branches so we can 
get this functionality tested and committed.


> (Oh yeah: can I backport it to 1.2?  :-)

You probably could but I don't know how cleanly the patches will apply.

-Dwayne.

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Re: [asterisk-users] [SOLVED] Linksys SPA3102-NA firmware upgrade on Linux

[Joseph]

On 08/20/08 14:46, Paul Hales wrote:
>Joseph wrote:
>> Does anybody know if the process of upgrading firmware on "Linksys 
>> SPA3102-NA" in Linux is the same as on Sipura 3K as described on 
>> voip-info.org
>> http://www.voip-info.org/wiki/view/Sipura
>>
>>   
>
>I'm pretty sure it works - I used it to upgrade a (god help me) SPA 9000 
>the other week.
>
>PaulH

No it DID NOT work for me - upgrading via http web-server as described in:
http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 
under: "Firmware" section

So I was forced myself to use an EXE file (what a pity).
However, I got in interesting post from a guru on Voxilla forum:

quote
Firmware can be loaded by placing a URL into Upgrade Rule (under Provisioning 
tab).

Example:
Place the following URL into the Upgrade Rule space: 
http://www.linuxstation.net/pub/voip/Linksys/Firmware/spa3102-5-1-7.bin

then save the changes. The ATA will reboot, then it will load the new firmware 
and reboot again.
-end quote--

I'll try it the next time, and I think www.voip-info.org should be appended 
with this information as good information is hard to find.

-- 
#Joseph
GPG KeyID: ED0E1FB7

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Re: [asterisk-users] Diamondware spatial conferencing

[Dean Collins]

This is a poor example but basically if you have 'stereo' I was thinking
an equation like this.


Number of speakers in a conference room = 'N' deviations 

Range = 80%  
(obviously you couldn't do 100% otherwise would be silent in outer
channels once you get over 10 participants.

(50/50 + / - the deviation N / range)


So in the example of a conference room with 3 participants deviation 26%

Caller 1 = 76% L + 24% R(-1 deviation of 26%)
Caller 2 = 50% L + 50% R(norm default 50/50)
Caller 3 = 24% L + 76% R(+1 deviation of 26%)





Example of a conference room with 7 participants deviation of 11% 

Caller 1 = 83% L + 17% R(-3 deviations of 33%)
Caller 2 = 72% L + 28% R(-2 deviations of 22%)
Caller 3 = 61% L + 39% R(-1 deviations of 11%)
Caller 4 = 50% L + 50% R(norm default 50/50)
Caller 5 = 39% L + 61% R(+1 deviations of 11%)
Caller 6 = 28% L + 72% R(+2 deviations of 22%)
Caller 7 = 17% L + 83% R(+3 deviations of 33%)




Cheers,

Dean



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Friday, 22 August 2008 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Diamondware spatial conferencing

On Fri, Aug 22, 2008 at 7:55 AM, Dean Collins <[EMAIL PROTECTED]>
wrote:
> It's probably covered under patents etc but has anyone tried to
'spatially
> separate' audio mixes for each participant in an Asterisk conference
call?

I've never tried it, but as soon as I heard about the idea a few
months ago, I thought it sounded like a great improvement to the
presence you feel in a conference.

r

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[asterisk-users] RES: DSS1 vs SS7

[Cordeiro, Marco]

Hello All, 

I have an Asterisk Box currently running 1.6.0, dahdi drivers and libss7
with a TE120P (01 E1) card for the past 02 months. 
I have it connected to a Cellular Operator switch (MSC), and it is working
perfectly. Traffic is still quite low, but increasing as we start to use it
for new applications everyday. 

I have made some stress call tests, using all available CICs at once, and
had no problem at all.

Congrats to the perfect development of the SS7 support to Mr. Fredrickson.

Hopefully soon we'll have MAP support as well. 

Marco


-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Matthew
Fredrickson
Enviada em: sexta-feira, 22 de agosto de 2008 11:49
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] DSS1 vs SS7

Kevin P. Fleming wrote:
> Alex Balashov wrote:
> 
>> Some carriers now do offer private SS7 instead of ISDN.  But there is 
>> absolutely no reason why you should be doing this with Asterisk. 
>> Asterisk-SS7 is quite tenuous at best.  Unless you have some specific 
>> reason to be using it, don't.
> 
> Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and
> it is being used in a quite a number of production deployments.

Thanks for the plug Kevin! :-)

Yeah, actually, if you guys want to know more there's an asterisk-ss7 
mailing list.  Asterisk-1.6.0 with libss7 is being used in many 
successful and high traffic installations around the world.

The current record (that I have been told of) is an installation doing 
over 100,000 calls per day.  So try to beat that ;-)

Matthew Fredrickson
Digium, Inc.

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Re: [asterisk-users] Diamondware spatial conferencing

[randulo]

On Fri, Aug 22, 2008 at 7:55 AM, Dean Collins <[EMAIL PROTECTED]> wrote:
> It's probably covered under patents etc but has anyone tried to 'spatially
> separate' audio mixes for each participant in an Asterisk conference call?

I've never tried it, but as soon as I heard about the idea a few
months ago, I thought it sounded like a great improvement to the
presence you feel in a conference.

r

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[asterisk-users] Friday's conference meeting - Astricon is in the air

[randulo]

We will be gathering at 9AM PDT, 12 Noon EDT, 4PM GMT (i think?) for
our weekly conference of asterisk and VoIP users. Any and all
discussion related to telephony is welcome. Please join us any Friday.
More info using the links below.

PSTN (724) 444-7444 and enter 22622# 1#
SIP [EMAIL PROTECTED] DTMF 22622# 1#
IRC.freenode.net #voip-users-conference

See http://bit.ly/voip for more info.

Note: we were supposed to hold the drawing for the free Astricon pass,
but Stelios was not able to do this today. Hopefully next week.

Randy

http://voipUsersConference.org

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Re: [asterisk-users] Problem with modem data calls and xorcom astribanks

[Tzafrir Cohen]

I'm working on this with Col Ferguson. I'll update the list when this is
resolved.

-- 
   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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[asterisk-users] Diamondware spatial conferencing

[Dean Collins]

Just read this on Alec Saunders fantastic blog
http://saunderslog.com/2008/08/21/diamondware-acquired/

 

Interesting concept, makes me wonder if it is possible in Asterisk to
'adjust' the left/right mix for audio conference participants?

Yeh I know there is only one channel in a telephone call but you get
what I mean.

 

It's probably covered under patents etc but has anyone tried to
'spatially separate' audio mixes for each participant in an Asterisk
conference call?

 

Regards,

Dean Collins
[EMAIL PROTECTED] 

+1-212-203-4357 (New York) 
+61-2-9016-5642 (Sydney)
http://www.Cognation.net  

 

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Re: [asterisk-users] DSS1 vs SS7

[Matthew Fredrickson]

Kevin P. Fleming wrote:
> Alex Balashov wrote:
> 
>> Some carriers now do offer private SS7 instead of ISDN.  But there is 
>> absolutely no reason why you should be doing this with Asterisk. 
>> Asterisk-SS7 is quite tenuous at best.  Unless you have some specific 
>> reason to be using it, don't.
> 
> Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and
> it is being used in a quite a number of production deployments.

Thanks for the plug Kevin! :-)

Yeah, actually, if you guys want to know more there's an asterisk-ss7 
mailing list.  Asterisk-1.6.0 with libss7 is being used in many 
successful and high traffic installations around the world.

The current record (that I have been told of) is an installation doing 
over 100,000 calls per day.  So try to beat that ;-)

Matthew Fredrickson
Digium, Inc.

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Re: [asterisk-users] Inefficient Codec Translation

[Steve Totaro]

On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin <[EMAIL PROTECTED]> wrote:
> We run asterisk to handle incoming DIDs and we have observed
> inefficient Codec Translation.
>
> Here is the scenario
>
> [DID Vendor] ---> [Asterisk ]
> > External GW [G729]
>  |
>
> |---> External GW [iLBC]
>
> Our DID vendor and asterisk box supports both ilbc & g729. However,
> our external gateway termination supports either ilbc or g729 (and not
> both) and depending on users location, we terminate it on either
> gateway.
>
> Since DID and asterisk box supports both the codecs, we assumed that
> asterisk will appropriately select codecs depending on where we
> terminate the call so that no codec translation happens. However, this
> seems to be an incorrect assumption and we see that different codecs
> get selected on two legs which leads to quality drop and extra CPU
> cycles.
>
> May be we are doing something wrong. Pls suggest what we are doing
> wrong. Below is asterisk configuration.
>
> [did]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=g729
> allow=ilbc
>
> [gw1]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=g729
>
> [gw2]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=ilbc
>
> Thanks
> Jim
>

To be more clear, when a call comes in on [did] the codec should match
on the order that they are listed and supported, so you should always
get g729 and transcode to ilbc on gw2.

Asterisk handles one leg at a time, so it does not look ahead to see
what the second leg of the call will be using for it's codec.

Thanks,
Steve T

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Re: [asterisk-users] Inefficient Codec Translation

[Steve Totaro]

On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin <[EMAIL PROTECTED]> wrote:
> We run asterisk to handle incoming DIDs and we have observed
> inefficient Codec Translation.
>
> Here is the scenario
>
> [DID Vendor] ---> [Asterisk ]
> > External GW [G729]
>  |
>
> |---> External GW [iLBC]
>
> Our DID vendor and asterisk box supports both ilbc & g729. However,
> our external gateway termination supports either ilbc or g729 (and not
> both) and depending on users location, we terminate it on either
> gateway.
>
> Since DID and asterisk box supports both the codecs, we assumed that
> asterisk will appropriately select codecs depending on where we
> terminate the call so that no codec translation happens. However, this
> seems to be an incorrect assumption and we see that different codecs
> get selected on two legs which leads to quality drop and extra CPU
> cycles.
>
> May be we are doing something wrong. Pls suggest what we are doing
> wrong. Below is asterisk configuration.
>
> [did]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=g729
> allow=ilbc
>
> [gw1]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=g729
>
> [gw2]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=ilbc
>
> Thanks
> Jim
>

Why don't you allow=g729 only on all entries.  Maybe I have misread
your email but I interpret what you wrote to mean that all endpoints
support g729

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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers

[Michael Graves]

On Fri, 22 Aug 2008 13:23:09 +0200, Johansson Olle E wrote:

>21 aug 2008 kl. 16.47 skrev <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>:
>
>> Yesterday I blogged a post about some ideas that I think will help
>> Asterisk appliances further penetrate SMB/SOHO sites in ways that are
>> not presently being addressed.
>
>I would prefer if you mailed the content too. After all this is a  
>mailing list. Clicking on the link
>just to see what the topic is is something that most readers won't do.  
>And it doesn't benefit the
>archives either. YOu can add a link, but just saying "Hey, I blogged  
>something interesting"
>without saying anything about the topic is not very helpful.
>
>Just some friendly advice if you really want a discussion. Of course,  
>I clicked, read and commented ;-)

My appologies. I'm mindful of not posting inappropriately to mailing
lists. Many thanks to those who read to, and the few who saw fit to
comment.

My premise is very simple. Any Asterisk Appliance in a small business
stands a good change of being core infrastructure. If it has hooks to
extend its reach easily into aspects of the business just beyond the
basic telephony/UC sphere then it may be dramatically more valuable to
the end user.

In my particular case I have some nice Polycom and Aastra desk phones.
I'd like to leverage the XHTML browsers in those phones to serve some
utility functions, like opening an electric door release, electric
gate, etc. I don't see why this sort of thing needs to be as difficult
as it is presently.

It seems that at the moment such matters are wholly DIY, or at best
left to a consultant. This takes them out of the sphere of possibility
of a large number of smaller installations, and therefore reduces the
potential utility of the PBX. That seems a waste.

The appliance approach is supposed to make things easier for end user
sites. I think that we should take a broad view of that, and not focus
solely on the telephony aspect. Consider the device a possible solution
to a variety of business needs. Of course, there are limits. I'd never
suggest a production Asterisk box be used as a file server beyond
provisioning phones.

Michael

Michael
--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves



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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[Steve Totaro]

Then after many hours of chasing ghosts, you decide, hmmm, let me
eliminate IAX2 as a possible cause, and boom!, everything works and
voice quality is perfect.

Then you are happy you got it fixed but mad you you wasted so much
time an the highly touted IAX2 that should "just work" but doesn't in
many cases.

Thanks,
Steve T

On Fri, Aug 22, 2008 at 10:22 AM, Darren Sessions <[EMAIL PROTECTED]> wrote:
> It's tough to say why a voice would start sounding like a robot. There are
> so many variables that could effect your Asterisk server.
> I always go for process of elimination when I have a problem similar to this
> with call quality.
> What I would do is install an end point on the same local network / subnet
> as your asterisk server (either a hard phone or a soft phone like X-Lite by
> Counterpath). Register the phone locally with your Asterisk server and make
> some calls or put an echo tester up.
> If things sound good, you know your Asterisk server is working just fine,
> and the problems lies somewhere on your network between the Asterisk server
> and whatever gateway / device. If it sounds awful, and the codecs match,
> then it's time to start troubleshooting the server.
>
> _
> Darren Sessions
> [EMAIL PROTECTED]
> http://www.darrensessions.com
> _
>
>
>
>
> On Aug 22, 2008, at 3:14 AM, bilal ghayyad wrote:
>
> Dear Darren;
>
> You might be right because one day it happened with me and the situation was
> same like this as following:
>
> The status that the ping result is very good for all partied (Asterisk
> machine, IP Phones on the Internet), and no problem in the processor
> utilization or RAM or hard disk space.
>
> Previously, we changed the DSL router and it worked fine !!
>
> But what can I do on the Asterisk level to overcome the problem?
>
> I already enabled the jitter on the IAX and SIP, but did not resolved. And I
> am using the G729 codec and sometimes I use GSM.
>
>
> Any advise for the robot voice with weak battery :) ?!
>
> Regards
> Bilal
>
> --- On Thu, 8/21/08, Darren Sessions <[EMAIL PROTECTED]> wrote:
>
> From: Darren Sessions <[EMAIL PROTECTED]>
>
> Subject: Re: [asterisk-users] Suddenly the voice become like robot
> (cutting), like sick man
>
> To: [EMAIL PROTECTED]
>
> Date: Thursday, August 21, 2008, 9:47 PM
>
> I doubt recompiling is going to help you unless you've
>
> got a very
>
> unstable system (hard drive going out or something), and
>
> then you've
>
> got bigger things to worry about then anyways.
>
> You should install (if you haven't already) the
>
> 'top' program. Top
>
> gives you a nice set of system statistics and a list of
>
> processes.
>
> If you're only having issues on the IP origination side
>
> of things, I
>
> would start checking your bandwidth and latency on your
>
> network.
>
> Is the originating end point on the Internet? or local?
>
>
> _
>
> Darren Sessions
>
> [EMAIL PROTECTED]
>
> http://www.darrensessions.com
>

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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers

[randulo]

On Fri, Aug 22, 2008 at 7:18 AM, Tim Panton <[EMAIL PROTECTED]> wrote:
> I often read this list offline (during my commute) and articles which
> reference a web page without at least summarizing the content
> are frustrating :-)

Not to argue, but to add that for what you describe I use Google
Reader. I load up on bligs (such as Michael's) and read it all offline
on trips. Of course Google Gears does not store images, but you have
whatever part of the text is available via RSS.

r

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Re: [asterisk-users] Problem with modem data calls and xorcom astribanks

[Darren Sessions]

Not sure what you've heard before, but I have successfully used a  
modem at 9600 baud (forced via AT commands) through a zaptel card on  
several occasions.



_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 22, 2008, at 8:14 AM, Greg Woods wrote:

I have been told before on this list that a modem through a zaptel  
card

will not work. And mine doesn't, at least not for data calls (it works
fine for fax). Apparently the modem requires the full bandwidth of the
POTS line, which you do not get through the zaptel card.

You might at least check to make sure that echo cancellation is turned
off. That can interfere with a data call.

--Greg



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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[Darren Sessions]

It's tough to say why a voice would start sounding like a robot. There  
are so many variables that could effect your Asterisk server.


I always go for process of elimination when I have a problem similar  
to this with call quality.


What I would do is install an end point on the same local network /  
subnet as your asterisk server (either a hard phone or a soft phone  
like X-Lite by Counterpath). Register the phone locally with your  
Asterisk server and make some calls or put an echo tester up.


If things sound good, you know your Asterisk server is working just  
fine, and the problems lies somewhere on your network between the  
Asterisk server and whatever gateway / device. If it sounds awful, and  
the codecs match, then it's time to start troubleshooting the server.



_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 22, 2008, at 3:14 AM, bilal ghayyad wrote:


Dear Darren;

You might be right because one day it happened with me and the  
situation was same like this as following:


The status that the ping result is very good for all partied  
(Asterisk machine, IP Phones on the Internet), and no problem in the  
processor utilization or RAM or hard disk space.


Previously, we changed the DSL router and it worked fine !!

But what can I do on the Asterisk level to overcome the problem?

I already enabled the jitter on the IAX and SIP, but did not  
resolved. And I am using the G729 codec and sometimes I use GSM.



Any advise for the robot voice with weak battery :) ?!

Regards
Bilal

--- On Thu, 8/21/08, Darren Sessions <[EMAIL PROTECTED]> wrote:


From: Darren Sessions <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Suddenly the voice become like robot  
(cutting), like sick man

To: [EMAIL PROTECTED]
Date: Thursday, August 21, 2008, 9:47 PM
I doubt recompiling is going to help you unless you've
got a very
unstable system (hard drive going out or something), and
then you've
got bigger things to worry about then anyways.

You should install (if you haven't already) the
'top' program. Top
gives you a nice set of system statistics and a list of
processes.

If you're only having issues on the IP origination side
of things, I
would start checking your bandwidth and latency on your
network.

Is the originating end point on the Internet? or local?


_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 21, 2008, at 4:55 PM, bilal ghayyad wrote:


Dear Darren;

I discovered that calling from the Asterisk to the IP

Phone

Extension (like calling from mobile to digium and then

enter the IP

Phone extension, or calling from fxs to the IP Phone

extension), it

goes very good without any problem.

But calling from the same IP Phone to another IP Phone

or to any

mobile (via fxo port) or to the fxs, it cause the

problem (voice

become very very bad, like robot with weak battery or

sick man).


Another way for the problem, if I called from another

Asterisk PBX

to our Asterisk PBX (that has the problem) and the

call was via IAX,

and I was need to reach to the IP Phone, then I hear

the voice like

robot with weak battery.

So, the problem appear if the call originator was IP

and not TDM.

What could be the reason for the problem? No one did

any change, I

am sure, it suddenly become like this.

Any help?
Regards
Bilal


--- On Thu, 8/21/08, Darren Sessions

<[EMAIL PROTECTED]> wrote:



From: Darren Sessions <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Suddenly the voice

become like robot

(cutting), like sick man
To: [EMAIL PROTECTED], "Asterisk Users

Mailing List - Non-

Commercial Discussion"



Date: Thursday, August 21, 2008, 6:13 PM
I'd run top on the server to see if the CPU

utilization

is going
through the roof. If you use AGI, make sure there
aren't any orphaned
processes consuming resources.

If all else fails on the software side of things,

I'd

restart the
server.


_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote:


Hi All;

My asterisk version is 1.4.19.2 and it

contains one

digium card of 2

fxs and 2 fxo ports, it was working great for

more

than one month

without any problem.

Suddenly, any call will be done, then voice

becoming

like robot (or

sick man), it slow and cutting.

I restarted the machine, but it is the same

!!!


I checked the RAM which is 1 GB and I found a

lot of

space.


Any advise what could be the problem?
Regards
Bilal








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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers

[Tim Panton]

On 22 Aug 2008, at 14:55, randulo wrote:

> On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E <[EMAIL PROTECTED]>  
> wrote:
>> Just some friendly advice if you really want a discussion. Of course,
>> I clicked, read and commented ;-)
>
> If this is a way we can get you to say something, Olle, I'm for it! :)
>
> This said, I think Michael was trying to work within the idea that
> those interested in this particular post might want to read it.
> Michael, if you (or anyone) want to post particular articles here, I'm
> sure everyone is ok with that. With the exception of spam, obviously.
> The problem too though is if there are images and links in the post,
> spam filters, etc. So in the long run, I'd say it's probably better
> just to do what he did :)

I'm more with Olle on this one.

I often read this list offline (during my commute) and articles which
reference a web page without at least summarizing the content
are frustrating :-)

Also the archives of this list are a valuable searchable resource
(again available offline as spotlight indexes them for me on the my  
mac).

So a short summary to accompany the link is great.



Tim.

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Re: [asterisk-users] Problem with modem data calls and xorcom astribanks

[Greg Woods]

I have been told before on this list that a modem through a zaptel card
will not work. And mine doesn't, at least not for data calls (it works
fine for fax). Apparently the modem requires the full bandwidth of the
POTS line, which you do not get through the zaptel card.

You might at least check to make sure that echo cancellation is turned
off. That can interfere with a data call.

--Greg



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Re: [asterisk-users] Inefficient Codec Translation

[Jim Boykin]

Requesting help.

Thaks


On Tue, Aug 19, 2008 at 4:40 PM, Jim Boykin <[EMAIL PROTECTED]> wrote:
> We run asterisk to handle incoming DIDs and we have observed
> inefficient Codec Translation.
>
> Here is the scenario
>
> [DID Vendor] ---> [Asterisk ]
> > External GW [G729]
>  |
>
> |---> External GW [iLBC]
>
> Our DID vendor and asterisk box supports both ilbc & g729. However,
> our external gateway termination supports either ilbc or g729 (and not
> both) and depending on users location, we terminate it on either
> gateway.
>
> Since DID and asterisk box supports both the codecs, we assumed that
> asterisk will appropriately select codecs depending on where we
> terminate the call so that no codec translation happens. However, this
> seems to be an incorrect assumption and we see that different codecs
> get selected on two legs which leads to quality drop and extra CPU
> cycles.
>
> May be we are doing something wrong. Pls suggest what we are doing
> wrong. Below is asterisk configuration.
>
> [did]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=g729
> allow=ilbc
>
> [gw1]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=g729
>
> [gw2]
> type=friend
> host=xxx
> canreinvite=yes
> disallow=all
> allow=ilbc
>
> Thanks
> Jim
>

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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers

[randulo]

On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E <[EMAIL PROTECTED]> wrote:
> Just some friendly advice if you really want a discussion. Of course,
> I clicked, read and commented ;-)

If this is a way we can get you to say something, Olle, I'm for it! :)

This said, I think Michael was trying to work within the idea that
those interested in this particular post might want to read it.
Michael, if you (or anyone) want to post particular articles here, I'm
sure everyone is ok with that. With the exception of spam, obviously.
The problem too though is if there are images and links in the post,
spam filters, etc. So in the long run, I'd say it's probably better
just to do what he did :)

Best,

r

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Re: [asterisk-users] interesting RDNIS question

[lenz]

I think you could minimize the incidence of the problem by having a PRI  
with like 100 numbers associated, with the CO doing  the routing stripping  
off the last two forwarded digits. You also have a premium service  
provider that forwards premium calls to one of those numbers (I think from  
what you say that you do not see the premium number as the originating  
ID). You rotate the accepted number daily/weekly, so it's very unlikely  
that someone  can simply guess it.
Just my euro .02,
l.

In data Fri, 22 Aug 2008 09:37:20 +0200, Sriram <[EMAIL PROTECTED]>  
ha scritto:

> Hi
>
> I am a premium voice service provider giving some services on IVR to a  
> Telco X . As my premises is some 10 kms away from that telco , i have  
> taken a PRI connection (30 DID with 1 hunting/pilot number) from telco  
> Y  When a customer of Telco X dials my short code @Rs.6/- per minute his  
> call is forwarded on the PRI connection of telco Y . All this works  
> fine..
>
> Now the problem arises during billing , many customers of Telco X /  
> Telco Z / Telco Y somehow get to know the pilot number of telco Y and  
> they directly dial in (it becomes a local call and not a premium rate)  
> the rsult being i dont get paid for those minutes and am giving the  
> service free virtually ...I tried to solve the problem as follows :
>
> 1. If i filter the calls using DNIS - no matter people call short code  
> or my pilot number - the DNIS would always be returned as the pilot  
> number
> 2. If i filter calls using ANI so that i allow  only customer of Telco X  
> , then eventhough i minimise the damage - but still am not sure if that  
> customer X has dialled short code or long code ?
>
> this question may sound off-topic but in asterisk is there a way out ?
>
> Rgds
> sriram



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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[Steve Totaro]

On Fri, Aug 22, 2008 at 5:14 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Dear Darren;
>
> You might be right because one day it happened with me and the situation was 
> same like this as following:
>
> The status that the ping result is very good for all partied (Asterisk 
> machine, IP Phones on the Internet), and no problem in the processor 
> utilization or RAM or hard disk space.
>
> Previously, we changed the DSL router and it worked fine !!
>
> But what can I do on the Asterisk level to overcome the problem?
>
> I already enabled the jitter on the IAX and SIP, but did not resolved. And I 
> am using the G729 codec and sometimes I use GSM.
>
>
> Any advise for the robot voice with weak battery :) ?!
>
> Regards
> Bilal
>

Try getting rid of IAX2 and use all SIP.  This has fixed the issue you
describe more than once in my experience.

There are obviously many other things that can cause this, but IAX2 is
the first to go when I am troubleshooting and there is no glaring
reason why.

You can usually setup OpenVPN for ease, or port forwarding to get
around NAT issues.

Give it a go, and let us know.

Thanks,
Steve Totaro

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Re: [asterisk-users] OT - Asterisk-Stats - Billsec instead of Duration

[Anthony Messina]

On Friday 22 August 2008 07:54:43 am Anthony Messina wrote:
> On Thursday 21 August 2008 08:26:47 am Olivier wrote:
> > Hi,
> >
> > To check telco billing, I'm usinfg Asterisk-Stats from
> > http://www.areski.net/asterisk-stat-v2/about.php .
> >
> > How can you tweak this application to display graphics and data that use
>
> i started working from that software to come up that was maybe simpler and
> css-based.  i'm still messing around with it, but you can look at
>
> https://messinet.com/svn/projects/asterisk-stat/trunk/

sorry, i forgot to mention that all you'd need to change is 
the 'formatDuration' function in include/config.inc.php

i currently have the billing duration contained in the  tag and it is 
displayed on hover over the duration.  but that is easily switched.

-- 
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] Changing callerID in a context

[Kevin P. Fleming]

Atis Lezdins wrote:

> [clid-mangle]
> exten => 70[01],1,Set(CALLERID(num)=581557)
> exten => 70[01],2,Return()
> exten => 10[01],1,Set(CALLERID(num)=581500)
> exten => 10[01],2,Return()
> ; and so on, just better reorganize your extensions so that this can
> match patterns better.
> 
> [dial-out]
> exten => _9.,1,GoSub(clid-mangle,${CALLERID(num)},1)
> exten => _9.,2,Dial(SIP/provider)

Actually, this can be even easier (although you didn't use the actual
CLID matching style):

[dial-out]
exten => _9./70[01],1,Set(CALLERID(num)=581557)
exten => _9./10[01],1,Set(CALLERID(num)=581500)
... more of the same 'priority 1' steps
exten => _9.,2,Dial(SIP/provider)

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] set callerid with plus sign

[Eric "ManxPower" Wieling]

+ is not a valid Caller*ID character.  Asterisk allows you to use + in 
Caller*ID, but many carriers will reject the call if you do that.

Benny Amorsen wrote:
> ronald <[EMAIL PROTECTED]> writes:
> 
>> Is it possible to assign a plus sign on the callerid(num) ?
> 
> Yes.
> 
>> currently this is what i do CALLERID(num)=+6523450017
>>
>> but telco is denying calls, coz they said they are seeing "bs523450017" 
>> instead of +6523450017.
> 
> Which techology? SIP? PRI? POTS? ...?
> 
> 
> /Benny
> 
> 
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-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
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Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)

[Michael Graves]

On Fri, 22 Aug 2008 01:13:25 -0700, Paul Chambers wrote:

>Another example of the North American frequency allocations being just a 
>little bit different from everywhere else in the world...  So does that 
>mean you've stopped using your S685 IP, Michael? ;)

Yes, between the power problems that I encountered (my own fault I
suspect) and certain minor quirks of the phones I've gone back to my
snom m3.

>Siemens USA does offer a few of the Gigaset DECT models over here (e.g. 
>the E450, S450 and S455 are not hard to find). But none of the -IP 
>models (e.g. no S450 IP). Perhaps they'd enjoy more success in the CE 
>business if they offered those :)

It's not at all claer what their thought process is about this. I know
that VOIP Supply has tried to get access to their product line, but
without success.

>If WiFi handsets were more affordable I wouldn't have this problem :)

True enough. I was surprised at how well Polycom's SpectraLink 8002
worked with an appropriate access point.

I still like the m3. Snom's commitment to continued firmware
development gives me some hope that it's one or two shortcommings will
eventually be overcome.

Michael

>Paul
>
>[EMAIL PROTECTED] wrote:
>> That's the purely technological answer, which is completely correct. 
>>
>> There's a business side to it as well. Siemens is simply not in the
>> consumer electronics business in North America. They make this decision
>> consciously. 
>>
>> Michael Graves
>> mgraves  mstvp.com
>> o(713) 861-4005
>> c(713) 201-1262
>> sip:[EMAIL PROTECTED]
>> skype mjgraves
>> FWD 54245
>>
>>   
>>>  Original Message 
>>> Subject: Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in
>>> particular)
>>> From: Drew Gibson <[EMAIL PROTECTED]>
>>> Date: Thu, August 21, 2008 12:08 pm
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> 
>>>
>>>
>>> Paul Chambers wrote:
>>> 
 For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP 
 range in the U.S. I'm particularly interested in the Gigaset S685 IP. 
 Since it's DECT 6.0, and there's an English (UK) version, I'm thinking 
 it should work just fine, after dealing with the walwart issue (and 
 maybe caller ID signalling).

 Anyone imported one from the UK and using it in the US? for how long? 
 impressions? anything not working?

 Have you purchased additional US-spec handsets and used them with the UK 
 basestation?

 Thanks in advance,

 Paul
   
>>> The original DECT standard uses 1880-1900MHz, as implemented in Europe.
>>>
>>> The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0.
>>>
>>> The FCC might get angry at you for using regular DECT phones in the US.
>>>
>>> And your neighbours with iPhones (GSM) might also get angry...
>>>
>>> regards,
>>>
>>> Drew
>>>
>>>
>>> -- 
>>> Drew Gibson
>>>
>>> Systems Administrator
>>> OANDA Corporation
>>> www.oanda.com
>>> 
>
>
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>12:58 PM
>
>

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves



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Re: [asterisk-users] DSS1 vs SS7

[Kevin P. Fleming]

Alex Balashov wrote:

> Some carriers now do offer private SS7 instead of ISDN.  But there is 
> absolutely no reason why you should be doing this with Asterisk. 
> Asterisk-SS7 is quite tenuous at best.  Unless you have some specific 
> reason to be using it, don't.

Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and
it is being used in a quite a number of production deployments.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] OT - Asterisk-Stats - Billsec instead of Duration

[Anthony Messina]

On Thursday 21 August 2008 08:26:47 am Olivier wrote:
> Hi,
>
> To check telco billing, I'm usinfg Asterisk-Stats from
> http://www.areski.net/asterisk-stat-v2/about.php .
>
> How can you tweak this application to display graphics and data that use

i started working from that software to come up that was maybe simpler and 
css-based.  i'm still messing around with it, but you can look at

https://messinet.com/svn/projects/asterisk-stat/trunk/

-- 
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] set callerid with plus sign

[Benny Amorsen]

ronald <[EMAIL PROTECTED]> writes:

> Is it possible to assign a plus sign on the callerid(num) ?

Yes.

> currently this is what i do CALLERID(num)=+6523450017
>
> but telco is denying calls, coz they said they are seeing "bs523450017" 
> instead of +6523450017.

Which techology? SIP? PRI? POTS? ...?


/Benny


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Re: [asterisk-users] Changing callerID in a context

[Atis Lezdins]

On Thu, Aug 21, 2008 at 3:11 PM, Andy Dixon <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I am trying to alter the outbound callerID for extensions within a
> context I have created.
>
> I wrote the following:
>
> exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$
> {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560)
> exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$
> {REALCALLERIDNUM}" = "671"]]|Set|CALLERID(num)=581561)
> exten => _9.,4,ExecIf($[$["${REALCALLERIDNUM}" = "362"] | $["$
> {REALCALLERIDNUM}" = "672"]]|Set|CALLERID(num)=581562)
> exten => _9.,5,ExecIf($[$["${REALCALLERIDNUM}" = "363"] | $["$
> {REALCALLERIDNUM}" = "673"]]|Set|CALLERID(num)=581563)
> exten => _9.,6,ExecIf($[$["${REALCALLERIDNUM}" = "364"] | $["$
> {REALCALLERIDNUM}" = "674"]]|Set|CALLERID(num)=581564)
> exten => _9.,7,ExecIf($[$["${REALCALLERIDNUM}" = "365"] | $["$
> {REALCALLERIDNUM}" = "675"]]|Set|CALLERID(num)=581565)
> exten => _9.,8,ExecIf($[$["${REALCALLERIDNUM}" = "366"] | $["$
> {REALCALLERIDNUM}" = "676"]]|Set|CALLERID(num)=581566)
> exten => _9.,9,ExecIf($[$["${REALCALLERIDNUM}" = "367"] | $["$
> {REALCALLERIDNUM}" = "677"]]|Set|CALLERID(num)=581567)
> exten => _9.,10,ExecIf($[$["${REALCALLERIDNUM}" = "368"] | $["$
> {REALCALLERIDNUM}" = "678"]]|Set|CALLERID(num)=581568)
> exten => _9.,11,ExecIf($[$["${REALCALLERIDNUM}" = "369"] | $["$
> {REALCALLERIDNUM}" = "679"]]|Set|CALLERID(num)=581569)
> exten => _9.,12,ExecIf($[$["${REALCALLERIDNUM}" = "700"] | $["$
> {REALCALLERIDNUM}" = "701"]]|Set|CALLERID(num)=581557)
> exten => _9.,13,ExecIf($[$["${REALCALLERIDNUM}" = "100"] | $["$
> {REALCALLERIDNUM}" = "101"]]|Set|CALLERID(num)=581500)
>
>
> This *should* change the callerID for (for example) 700 and 701 to be
> 581557, and any extensions not listed above, it should leave them alone.
>
> If I call from extension 666, I get the correct outbound number (as it
> does exist), but the rules above are not being followed.
>
> I have tried to use Set(CALLERID(num)=581500) which works okay
> slightly further down.
>
> I am aiming for any numbers starting with a 9 to follow the rules
> above, and then to follow a further rule (eg if the number starts 901,
> or 907)
>
> I'm stuck.. If anyone could help, I would be eternally grateful..

Are you sure ${REALCALLERIDNUM} is set? Alternatively (to AEL) there's
a way how to simplify all this, by using Asterisk extension patterns:

[clid-mangle]
exten => 70[01],1,Set(CALLERID(num)=581557)
exten => 70[01],2,Return()
exten => 10[01],1,Set(CALLERID(num)=581500)
exten => 10[01],2,Return()
; and so on, just better reorganize your extensions so that this can
match patterns better.

[dial-out]
exten => _9.,1,GoSub(clid-mangle,${CALLERID(num)},1)
exten => _9.,2,Dial(SIP/provider)



Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Asterisk and Huawei SoftX3000

[Gustavo A Gonzalez]

 

 

 

Thanks for your answer and doing some test I have this SIP debug:

 

>From Huawei SIDE we have:

 

 

12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], proto:
UDP (17), length: 856) 189.8.113.170.5060 > 189.8.126.177.5060: SIP, length:
828

INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP
189.8.113.170:5060;branch=z9hG4bKba4h2m2070fhnc4q20k1.1

Call-ID: [EMAIL PROTECTED]

From: Anonymous;tag=c8959281

To: 

CSeq: 1 INVITE

Contact: 

Supported: 100rel

Privacy: user

User-Agent: Huawei SoftX3000 V300R006

Max-Forwards: 69

Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,ME
SSAGE,REFER

Content-Length: 226

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 228436 228436 IN IP4 189.8.113.170

s=Sip Call

c=IN IP4 189.8.113.170

t=0 0

m=audio 49606 RTP/AVP 8 0 97

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

 

 

and from Asterisk SIDE we have

 

 

 

12:54:30.364337 IP (tos 0x0, ttl  64, id 7173, offset 0, flags [none],
proto: UDP (17), length: 791) 192.168.0.12.5060 > 189.8.113.170.5060: SIP,
length: 763

SIP/2.0 200 OK

Via: SIP/2.0/UDP
189.8.113.170:5060;branch=z9hG4bKba4h2m2070fhnc4q20k1.1;received=189.8.113.1
70

From: Anonymous;tag=c8959281

To: ;tag=as2daca8e1

Call-ID: [EMAIL PROTECTED]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: 

Content-Type: application/sdp

Content-Length: 259

 

v=0

o=root 4569 4569 IN IP4 192.168.0.12

s=session

c=IN IP4 192.168.0.12

t=0 0

m=audio 11442 RTP/AVP 8 0 97

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

12:54:34.132505 IP (tos 0x0, ttl  64, id 7174, offset 0, flags [none],
proto: UDP (17), length: 487) 192.168.0.12.5060 > 189.8.113.170.5060: SIP,
length: 459

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1
70

From: ;tag=c963ca26

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: 

Content-Length: 0

 

 

12:54:34.132729 IP (tos 0x0, ttl  64, id 7175, offset 0, flags [none],
proto: UDP (17), length: 503) 192.168.0.12.5060 > 189.8.113.170.5060: SIP,
length: 475

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP
189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1
70

From: ;tag=c963ca26

To: ;tag=as3b751604

Call-ID: [EMAIL PROTECTED]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: 

Content-Length: 0

 

 

12:54:34.634264 IP (tos 0x0, ttl  64, id 7176, offset 0, flags [none],
proto: UDP (17), length: 487) 192.168.0.12.5060 > 189.8.113.170.5060: SIP,
length: 459

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1
70

From: ;tag=c963ca26

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: 

Content-Length: 0

 

 

12:54:35.635851 IP (tos 0x0, ttl  64, id 7177, offset 0, flags [none],
proto: UDP (17), length: 487) 192.168.0.12.5060 > 189.8.113.170.5060: SIP,
length: 459

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1
70

From: ;tag=c963ca26

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: 

Content-Length: 0 

 

 

Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
[EMAIL PROTECTED] 

 

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Re: [asterisk-users] How to block incoming calls on PRI

[Jay R. Ashworth]

On Thu, Aug 21, 2008 at 08:36:44PM -0500, Dwayne Hubbard wrote:
> I also want to reiterate that the libpri and Asterisk branches above
> are development branches, so be careful in a production environment.
>  This functionality will be available in Asterisk 1.6.2.  To disable a
> channel via the CLI type 'pri service disable channel ' and to
> enable the channel type 'pri service enable channel '.


That sounds cool.  Two questions:

1) can you do it "graceful"ly (both that and immediate are sometimes
useful)?

2) can you take down either an entire span, or a channel range on the
command line?

(Oh yeah: can I backport it to 1.2?  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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[asterisk-users] Asterisk, Xen and a TDM400P

[--[ UxBoD ]--]

Hi,

Has anybody managed to get this configuration work with PCI passthrough or 
should I look to buying a separate server ?

Regards,

-- 
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import"
// Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84
// Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84
// Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED]

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


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Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

[Benny Amorsen]

"Philippe Sultan" <[EMAIL PROTECTED]> writes:

> Well, if someone steals the md5secret (HA1) for a given username and
> realm, he can use it to authenticate to the SIP proxy or B2BUA that
> serves the target user.

This is unavoidable with password-based systems.

Either you transfer the password unencrypted on the network (or
e.g. hashed with MD5, but that just means that the hash is the actual
password), and then you can store the password as a hash on the
server.

Or you use a secure protocol, e.g. a nonce-based one, to prove that
the other end has the same password as you -- but then the server
needs to have the unhashed password available for comparison.

SIP tries to do both, but effectively it picks the second choice:
Trust the server, not the network.

To do better you need public key cryptography. Alas, noone has
invented a way to create a private key from a password, so that means
you don't get to pick your own private key. Still, I think that would
have been a vastly better choice for SIP and for anything else where
humans aren't expected to regularly type their password. Either way,
SIP can't do it.

You can also go the whole way with client certificates and SIP/TLS,
and then you can hire a few people to keep your PKI running and secure
-- and I'm not sure that Asterisk can do it yet.


/Benny


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Re: [asterisk-users] Changing callerID in a context

[Doug Lytle]

Andy Dixon wrote:
> On 21 Aug 2008, at 14:40, Philipp Kempgen wrote:
>   
>>> This *should* change the callerID for (for example) 700 and 701 to be
>>> 581557, and any extensions not listed above, it should leave them  
>>> alone.
>>>   

Andy,

If you're not bound and determined to do it this way, you can use the 
MySQL addon to do a sql lookup for outbound CallerID. 

It works great.  Just before the dial out, we do a query against the 
extension, if there is a match, we set the caller-id to whatever was 
returned on the query, if there isn't a match, we use the main number.

If you'd like more information, let me know,

Doug

-- 
 
Ben Franklin quote:

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


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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers

[Johansson Olle E]

21 aug 2008 kl. 16.47 skrev <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>:

> Yesterday I blogged a post about some ideas that I think will help
> Asterisk appliances further penetrate SMB/SOHO sites in ways that are
> not presently being addressed.

I would prefer if you mailed the content too. After all this is a  
mailing list. Clicking on the link
just to see what the topic is is something that most readers won't do.  
And it doesn't benefit the
archives either. YOu can add a link, but just saying "Hey, I blogged  
something interesting"
without saying anything about the topic is not very helpful.

Just some friendly advice if you really want a discussion. Of course,  
I clicked, read and commented ;-)

/O

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Re: [asterisk-users] Changing callerID in a context

[Andy Dixon]

On 21 Aug 2008, at 14:40, Philipp Kempgen wrote:

> Andy Dixon schrieb:
>
>> I am trying to alter the outbound callerID for extensions within a
>> context I have created.
>>
>> I wrote the following:
>>
>> exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$
>> {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560)
>> exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$
>> {REALCALLERIDNUM}" = "671"]]|Set|CALLERID(num)=581561)
>> exten => _9.,4,ExecIf($[$["${REALCALLERIDNUM}" = "362"] | $["$
>> {REALCALLERIDNUM}" = "672"]]|Set|CALLERID(num)=581562)
>> exten => _9.,5,ExecIf($[$["${REALCALLERIDNUM}" = "363"] | $["$
>> {REALCALLERIDNUM}" = "673"]]|Set|CALLERID(num)=581563)
>> exten => _9.,6,ExecIf($[$["${REALCALLERIDNUM}" = "364"] | $["$
>> {REALCALLERIDNUM}" = "674"]]|Set|CALLERID(num)=581564)
>> exten => _9.,7,ExecIf($[$["${REALCALLERIDNUM}" = "365"] | $["$
>> {REALCALLERIDNUM}" = "675"]]|Set|CALLERID(num)=581565)
>> exten => _9.,8,ExecIf($[$["${REALCALLERIDNUM}" = "366"] | $["$
>> {REALCALLERIDNUM}" = "676"]]|Set|CALLERID(num)=581566)
>> exten => _9.,9,ExecIf($[$["${REALCALLERIDNUM}" = "367"] | $["$
>> {REALCALLERIDNUM}" = "677"]]|Set|CALLERID(num)=581567)
>> exten => _9.,10,ExecIf($[$["${REALCALLERIDNUM}" = "368"] | $["$
>> {REALCALLERIDNUM}" = "678"]]|Set|CALLERID(num)=581568)
>> exten => _9.,11,ExecIf($[$["${REALCALLERIDNUM}" = "369"] | $["$
>> {REALCALLERIDNUM}" = "679"]]|Set|CALLERID(num)=581569)
>> exten => _9.,12,ExecIf($[$["${REALCALLERIDNUM}" = "700"] | $["$
>> {REALCALLERIDNUM}" = "701"]]|Set|CALLERID(num)=581557)
>> exten => _9.,13,ExecIf($[$["${REALCALLERIDNUM}" = "100"] | $["$
>> {REALCALLERIDNUM}" = "101"]]|Set|CALLERID(num)=581500)
>>
>>
>> This *should* change the callerID for (for example) 700 and 701 to be
>> 581557, and any extensions not listed above, it should leave them  
>> alone.
>>
>> If I call from extension 666, I get the correct outbound number (as  
>> it
>> does exist), but the rules above are not being followed.
>>
>> I have tried to use Set(CALLERID(num)=581500) which works okay
>> slightly further down.
>>
>> I am aiming for any numbers starting with a 9 to follow the rules
>> above, and then to follow a further rule (eg if the number starts  
>> 901,
>> or 907)
>>
>> I'm stuck.. If anyone could help, I would be eternally grateful..
>
> This would be much more readable in AEL.
> Or in an external script.
> But maybe all you really need is fromuser in sip.conf or similar.

Hi,

I'm not wanting the callerID to change internally - eg so extension  
700 will show as 700 internally, but as 581557 when it goes through  
our ZAP trunk, which I believe the fromuser would cause.

Sorry to sound dim, but whats AEL?

Thanks

Andy


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Re: [asterisk-users] set callerid with plus sign

[Luis Morales]

Check on dial plan rules, remember if you need dial to +number your
rule must be +|number this submit the number on your dialout plan
without +.

Regards,

Luis Morales

On Fri, Aug 22, 2008 at 3:40 AM, ronald <[EMAIL PROTECTED]> wrote:
> Hi Sir,
>
> I actually have a plus sign on my dial plan
>
> exten => _+.,1,Dial (
>
> that is ok, dialed number (telco refers to it as B-number) is correct.
>
> the prob is the originating number(they call this A-Number), i want to
> set it to +65 so that it shows it is an international call.
>
> so on my dial plan:
>
> exten => _+.,1,Set(CALLERID(num)=+65)
> exten => _+.,1,Dial(SIP/[EMAIL PROTECTED])
>
> what i don't get is why +65 is being seen as bs5.
>
> Regards,
> Nhadie
>
>
>
> Darren Sessions wrote:
>> Just change your dial command and add the plus sign there.
>>
>>
>> _
>>
>> Darren Sessions
>> [EMAIL PROTECTED] 
>> http://www.darrensessions.com
>> _
>>
>>
>>
>>
>>
>> On Aug 22, 2008, at 1:28 AM, ronald wrote:
>>
>>> Hi,
>>>
>>> Is it possible to assign a plus sign on the callerid(num) ?
>>>
>>> currently this is what i do CALLERID(num)=+6523450017
>>>
>>> but telco is denying calls, coz they said they are seeing "bs523450017"
>>> instead of +6523450017.
>>>
>>> i tried putting it inside double quotes CALLERID(num)="+6523450017"
>>> telco says the same thing.
>>>
>>> is this possible? thank you
>>>
>>> Regards,
>>> nhadie
>>>
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>>
>>
>> 
>>
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>



-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

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-

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Re: [asterisk-users] callfiles/manager api originate call fails

[Rizwan Hisham]

Actually both calls have to be originated to the outside world. Thats why im
using @TRUNK-OUT, when the first call is answered only then the call goes to
a context. Thats where the problem is, the first call does not originate so
i cant throw it to any context.

On Thu, Aug 21, 2008 at 8:47 PM, Anthony Francis <[EMAIL PROTECTED]>wrote:

>
>
> Rizwan Hisham wrote:
> > Hi all,
> > asterisk is giving me tough time. its been 3 days I am trying to
> > originate outgoing call using manager api/callfiles.
> >
> I would say remove the "@TRUNK-OUT" part and make sure that the context
> you send the call to knows about sending calls to the outside world.
>
> --
> Thank you and have any kind of day you want,
>
> Anthony Francis
>
>
>
>
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Re: [asterisk-users] Problem with modem data calls and xorcom astribanks

[Col Ferguson]

Hello Tzafrir,
Yes the trunk is an FXO port in the astribank

One astribank is 32 FXS ports, and one is 24 FXS and 8 FXO ports.

Just in case it makes a difference, the testing I am doing is with the modem
plugged in to the same astribank as the FXO ports.

Zap/69 is an FXO port and Zap/67 is an FXS port

Thanks,
Col

- Original Message -
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
To: 
Sent: Friday, August 22, 2008 6:56 PM
Subject: Re: [asterisk-users] Problem with modem data calls and xorcom
astribanks


> On Fri, Aug 22, 2008 at 03:12:41PM +1000, Col Ferguson wrote:
> > Hello all,
> > I have a system at a motel that is mostly analog phones with 2 32 port
> > astribanks.
>
> What exactly is the trunk? FXO ports in the astribank?
>
> >
> > I am having problems getting a modem data call to connect.
> > There are many travelling salesmen that require this functionality to
work
> > to dial direct into their company systems.
> >
> > I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and
> > Oslec echo can.
> >
> > I have now the simplest dialplan I can come up with and get a 4800
> > connection about 1 in 10 times. This should bypass any smarts that
freePBX
> > is adding in.
> >
> > The dialplan is
> > [outbound-allroutes-custom]
> > exten => 791,1,Dial(Zap/69/ww019830,300)
> > exten => 791,n,Hangup
> >
> > In Hyperterminal I do
> > atdt791
> >
> > The number dialled is for a large dialup ISP.
> > ww is needed to get a dialtone for the modem. Could this be causing the
> > problem ?
> >
> > The log file shows
> > [Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing...
> > [Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww019830
> > [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answered
Zap/67-1
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69,
nothingok: 0
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talking
to
> > 69/0
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talking
to
> > 67/0
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at
0
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67
> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69
> > [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1
and
> > Zap/69-1
> > [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67
> > [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67
> > [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69
> > [Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1'
> >
> > Does anyone know if there is some type of native echo canceller in the
> > astribanks that could be affecting this ? Or anything else I could try ?
> > Looking at /proc/oslec/info shows that oslec is not being used at the
time.
> >
> > If I have the modem connected directly into the phone line, and
completely
> > bypass the astribank, I get a 50666 connection every time.
> >
> > Any suggestions gratefully accepted.
> >
> > Thanks,
> > Col
> >
> >
> > ___
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> >
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
>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] Suddenly the voice become like robot (cutting), like sick man

[bilal ghayyad]

Dear Darren;

You might be right because one day it happened with me and the situation was 
same like this as following:

The status that the ping result is very good for all partied (Asterisk machine, 
IP Phones on the Internet), and no problem in the processor utilization or RAM 
or hard disk space.

Previously, we changed the DSL router and it worked fine !!

But what can I do on the Asterisk level to overcome the problem?

I already enabled the jitter on the IAX and SIP, but did not resolved. And I am 
using the G729 codec and sometimes I use GSM.


Any advise for the robot voice with weak battery :) ?!

Regards
Bilal

--- On Thu, 8/21/08, Darren Sessions <[EMAIL PROTECTED]> wrote:

> From: Darren Sessions <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), 
> like sick man
> To: [EMAIL PROTECTED]
> Date: Thursday, August 21, 2008, 9:47 PM
> I doubt recompiling is going to help you unless you've
> got a very  
> unstable system (hard drive going out or something), and
> then you've  
> got bigger things to worry about then anyways.
> 
> You should install (if you haven't already) the
> 'top' program. Top  
> gives you a nice set of system statistics and a list of
> processes.
> 
> If you're only having issues on the IP origination side
> of things, I  
> would start checking your bandwidth and latency on your
> network.
> 
> Is the originating end point on the Internet? or local?
> 
> 
> _
> 
> Darren Sessions
> [EMAIL PROTECTED]
> http://www.darrensessions.com
> _
> 
> 
> 
> 
> 
> On Aug 21, 2008, at 4:55 PM, bilal ghayyad wrote:
> 
> > Dear Darren;
> >
> > I discovered that calling from the Asterisk to the IP
> Phone  
> > Extension (like calling from mobile to digium and then
> enter the IP  
> > Phone extension, or calling from fxs to the IP Phone
> extension), it  
> > goes very good without any problem.
> >
> > But calling from the same IP Phone to another IP Phone
> or to any  
> > mobile (via fxo port) or to the fxs, it cause the
> problem (voice  
> > become very very bad, like robot with weak battery or
> sick man).
> >
> > Another way for the problem, if I called from another
> Asterisk PBX  
> > to our Asterisk PBX (that has the problem) and the
> call was via IAX,  
> > and I was need to reach to the IP Phone, then I hear
> the voice like  
> > robot with weak battery.
> >
> > So, the problem appear if the call originator was IP
> and not TDM.  
> > What could be the reason for the problem? No one did
> any change, I  
> > am sure, it suddenly become like this.
> >
> > Any help?
> > Regards
> > Bilal
> >
> >
> > --- On Thu, 8/21/08, Darren Sessions
> <[EMAIL PROTECTED]> wrote:
> >
> >> From: Darren Sessions <[EMAIL PROTECTED]>
> >> Subject: Re: [asterisk-users] Suddenly the voice
> become like robot  
> >> (cutting), like sick man
> >> To: [EMAIL PROTECTED], "Asterisk Users
> Mailing List - Non- 
> >> Commercial Discussion"
> 
> >> Date: Thursday, August 21, 2008, 6:13 PM
> >> I'd run top on the server to see if the CPU
> utilization
> >> is going
> >> through the roof. If you use AGI, make sure there
> >> aren't any orphaned
> >> processes consuming resources.
> >>
> >> If all else fails on the software side of things,
> I'd
> >> restart the
> >> server.
> >>
> >>
> >> _
> >>
> >> Darren Sessions
> >> [EMAIL PROTECTED]
> >> http://www.darrensessions.com
> >> _
> >>
> >>
> >>
> >>
> >>
> >> On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote:
> >>
> >>> Hi All;
> >>>
> >>> My asterisk version is 1.4.19.2 and it
> contains one
> >> digium card of 2
> >>> fxs and 2 fxo ports, it was working great for
> more
> >> than one month
> >>> without any problem.
> >>>
> >>> Suddenly, any call will be done, then voice
> becoming
> >> like robot (or
> >>> sick man), it slow and cutting.
> >>>
> >>> I restarted the machine, but it is the same
> !!!
> >>>
> >>> I checked the RAM which is 1 GB and I found a
> lot of
> >> space.
> >>>
> >>> Any advise what could be the problem?
> >>> Regards
> >>> Bilal
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> ___
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> >>>
> >>> asterisk-users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>
> >>
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >


  

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Re: [asterisk-users] Problem with modem data calls and xorcom astribanks

[Tzafrir Cohen]

On Fri, Aug 22, 2008 at 03:12:41PM +1000, Col Ferguson wrote:
> Hello all,
> I have a system at a motel that is mostly analog phones with 2 32 port
> astribanks.

What exactly is the trunk? FXO ports in the astribank?

> 
> I am having problems getting a modem data call to connect.
> There are many travelling salesmen that require this functionality to work
> to dial direct into their company systems.
> 
> I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and
> Oslec echo can.
> 
> I have now the simplest dialplan I can come up with and get a 4800
> connection about 1 in 10 times. This should bypass any smarts that freePBX
> is adding in.
> 
> The dialplan is
> [outbound-allroutes-custom]
> exten => 791,1,Dial(Zap/69/ww019830,300)
> exten => 791,n,Hangup
> 
> In Hyperterminal I do
> atdt791
> 
> The number dialled is for a large dialup ISP.
> ww is needed to get a dialtone for the modem. Could this be causing the
> problem ?
> 
> The log file shows
> [Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing...
> [Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww019830
> [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answered Zap/67-1
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69, nothingok: 0
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talking to
> 69/0
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talking to
> 67/0
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at 0
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67
> [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69
> [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1 and
> Zap/69-1
> [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67
> [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67
> [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69
> [Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1'
> 
> Does anyone know if there is some type of native echo canceller in the
> astribanks that could be affecting this ? Or anything else I could try ?
> Looking at /proc/oslec/info shows that oslec is not being used at the time.
> 
> If I have the modem connected directly into the phone line, and completely
> bypass the astribank, I get a 50666 connection every time.
> 
> Any suggestions gratefully accepted.
> 
> Thanks,
> Col
> 
> 
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-- 
   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] Siemens Gigaset IP in USA (S685 IP in particular)

[Paul Chambers]

Another example of the North American frequency allocations being just a 
little bit different from everywhere else in the world...  So does that 
mean you've stopped using your S685 IP, Michael? ;)

Siemens USA does offer a few of the Gigaset DECT models over here (e.g. 
the E450, S450 and S455 are not hard to find). But none of the -IP 
models (e.g. no S450 IP). Perhaps they'd enjoy more success in the CE 
business if they offered those :)

If WiFi handsets were more affordable I wouldn't have this problem :)

Paul

[EMAIL PROTECTED] wrote:
> That's the purely technological answer, which is completely correct. 
>
> There's a business side to it as well. Siemens is simply not in the
> consumer electronics business in North America. They make this decision
> consciously. 
>
> Michael Graves
> mgraves  mstvp.com
> o(713) 861-4005
> c(713) 201-1262
> sip:[EMAIL PROTECTED]
> skype mjgraves
> FWD 54245
>
>   
>>  Original Message 
>> Subject: Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in
>> particular)
>> From: Drew Gibson <[EMAIL PROTECTED]>
>> Date: Thu, August 21, 2008 12:08 pm
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>>
>>
>> Paul Chambers wrote:
>> 
>>> For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP 
>>> range in the U.S. I'm particularly interested in the Gigaset S685 IP. 
>>> Since it's DECT 6.0, and there's an English (UK) version, I'm thinking 
>>> it should work just fine, after dealing with the walwart issue (and 
>>> maybe caller ID signalling).
>>>
>>> Anyone imported one from the UK and using it in the US? for how long? 
>>> impressions? anything not working?
>>>
>>> Have you purchased additional US-spec handsets and used them with the UK 
>>> basestation?
>>>
>>> Thanks in advance,
>>>
>>> Paul
>>>   
>> The original DECT standard uses 1880-1900MHz, as implemented in Europe.
>>
>> The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0.
>>
>> The FCC might get angry at you for using regular DECT phones in the US.
>>
>> And your neighbours with iPhones (GSM) might also get angry...
>>
>> regards,
>>
>> Drew
>>
>>
>> -- 
>> Drew Gibson
>>
>> Systems Administrator
>> OANDA Corporation
>> www.oanda.com
>> 


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Re: [asterisk-users] set callerid with plus sign

[ronald]

Hi Sir,

I actually have a plus sign on my dial plan

exten => _+.,1,Dial (

that is ok, dialed number (telco refers to it as B-number) is correct.

the prob is the originating number(they call this A-Number), i want to 
set it to +65 so that it shows it is an international call.

so on my dial plan:

exten => _+.,1,Set(CALLERID(num)=+65)
exten => _+.,1,Dial(SIP/[EMAIL PROTECTED])

what i don't get is why +65 is being seen as bs5.

Regards,
Nhadie



Darren Sessions wrote:
> Just change your dial command and add the plus sign there.
> 
> 
> _
> 
> Darren Sessions
> [EMAIL PROTECTED] 
> http://www.darrensessions.com
> _
> 
> 
> 
> 
> 
> On Aug 22, 2008, at 1:28 AM, ronald wrote:
> 
>> Hi,
>>
>> Is it possible to assign a plus sign on the callerid(num) ?
>>
>> currently this is what i do CALLERID(num)=+6523450017
>>
>> but telco is denying calls, coz they said they are seeing "bs523450017"
>> instead of +6523450017.
>>
>> i tried putting it inside double quotes CALLERID(num)="+6523450017"
>> telco says the same thing.
>>
>> is this possible? thank you
>>
>> Regards,
>> nhadie
>>
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> 
> 
> 
> 
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[asterisk-users] interesting RDNIS question

[Sriram]

Hi 

I am a premium voice service provider giving some services on IVR to a Telco X 
. As my premises is some 10 kms away from that telco , i have taken a PRI 
connection (30 DID with 1 hunting/pilot number) from telco Y  When a customer 
of Telco X dials my short code @Rs.6/- per minute his call is forwarded on the 
PRI connection of telco Y . All this works fine..

Now the problem arises during billing , many customers of Telco X / Telco Z / 
Telco Y somehow get to know the pilot number of telco Y and they directly dial 
in (it becomes a local call and not a premium rate) the rsult being i dont get 
paid for those minutes and am giving the service free virtually ...I tried to 
solve the problem as follows :

1. If i filter the calls using DNIS - no matter people call short code or my 
pilot number - the DNIS would always be returned as the pilot number
2. If i filter calls using ANI so that i allow  only customer of Telco X , then 
eventhough i minimise the damage - but still am not sure if that customer X has 
dialled short code or long code ?

this question may sound off-topic but in asterisk is there a way out ?

Rgds
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Re: [asterisk-users] set callerid with plus sign

[Darren Sessions]

Just change your dial command and add the plus sign there.


_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 22, 2008, at 1:28 AM, ronald wrote:


Hi,

Is it possible to assign a plus sign on the callerid(num) ?

currently this is what i do CALLERID(num)=+6523450017

but telco is denying calls, coz they said they are seeing  
"bs523450017"

instead of +6523450017.

i tried putting it inside double quotes CALLERID(num)="+6523450017"
telco says the same thing.

is this possible? thank you

Regards,
nhadie

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[asterisk-users] set callerid with plus sign

[ronald]

Hi,

Is it possible to assign a plus sign on the callerid(num) ?

currently this is what i do CALLERID(num)=+6523450017

but telco is denying calls, coz they said they are seeing "bs523450017" 
instead of +6523450017.

i tried putting it inside double quotes CALLERID(num)="+6523450017" 
telco says the same thing.

is this possible? thank you

Regards,
nhadie

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