Re: [asterisk-users] Two PRI setup questions

[Tilghman Lesher]

On Thursday 01 November 2007 19:31:39 Lutgring, Sam wrote:
> 2)  Is there a way to see the idle status of a B channel?  When AT&T tells
> me they don't see the B channels coming up, is there a way that I can see
> this in Asterisk???

Ask AT&T to turn off "B channel maintenance protocol" on the PRI.  Asterisk
does not support this mode.

-- 
Tilghman

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Re: [asterisk-users] zaptel.conf missing

[Rilawich Ango]

just copy from SRC/zaptel.conf.sample

On 11/2/07, kitti jaisong <[EMAIL PROTECTED]> wrote:
> Hi all
> I have installed zaptel on debian and missing file zaptel.conf in
> /etc/zaptel.conf .my system don't have card TDM
> please advice what the missing.
>
> thanks,
> kitti
>
>
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[asterisk-users] zaptel.conf missing

[kitti jaisong]

Hi all
I have installed zaptel on debian and missing file zaptel.conf in 
/etc/zaptel.conf .my system don't have card TDM
please advice what the missing.

thanks,
kitti


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[asterisk-users] __sip_xmit problem

[Rilawich Ango]

Hi,
  I got the following warning from CLI when I try to execute the Dial
command.  It makes the call failed.  Anyone can tell me what does it
mean and how to solve?

-- Executing [EMAIL PROTECTED]:61] Dial("SIP/4009-1f178ba0",
"SIP/[EMAIL PROTECTED]|35|L(7200:12)") in new stack
-- Limit Data for this call:
   > timelimit  = 7200
   > play_warning   = 12
   > play_to_caller = yes
   > play_to_callee = no
   > warning_freq   = 0
   > start_sound= (null)
   > warning_sound  = timeleft
   > end_sound  = (null)
[Nov  2 11:54:00] WARNING[8218]: chan_sip.c:1775 __sip_xmit: sip_xmit
of 0x1f188330 (len 793) to 10.0.0.22:0 returned -1: Invalid argument
-- Called [EMAIL PROTECTED]
[Nov  2 11:54:01] WARNING[8206]: chan_sip.c:1775 __sip_xmit: sip_xmit
of 0x1f188330 (len 793) to 10.0.0.22:0 returned -1: Invalid argument
[Nov  2 11:54:02] WARNING[8206]: chan_sip.c:1775 __sip_xmit: sip_xmit
of 0x1f188330 (len 793) to 10.0.0.22:0 returned -1: Invalid argument

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Re: [asterisk-users] OpenSER for Asterisk Load balance

[ram]

On 11/2/07, Edgar Guadamuz <[EMAIL PROTECTED]> wrote:
>
> Hi, guys
>
> I´ve just seen thta OpenSER can be coupled with Asterisk for load
> balance, with the dispatcher module, something like this:
>
> dispatcher.cfg file
>
> # group  sip addresses of your * units
> 1 sip:10.1.2.3:5060
> 1 sip:10.1.2.4:5060
> 1 sip:10.1.2.5:5060
>
> the basic openser.cfg should be like:
>
> loadmodule("dispatcher.so")
>
> ...
>
> if ( method=="INVITE" ) {
> # dst_select( "GROUP", "HASH METHOD")
> dst_select("1","4");
> sl_send_reply("100","Trying");
> forward(uri:host, uri:port);
> exit();
> }
>
> That´s OK, but what about failover, I mean, if a Asterisk box crashes,
> the dispatcher module will continue sending requests for that IP and
> in this case something like heartbeat had to be implemented to take
> the failed IP, but it would be more efficient if we could have OpenSER
> monitoring the Asterisk servers health, anyone knows how?


Hi

the recomendations are DNS SRV records or
do in round robin basic.

Like failover routes from OpenSER

http://gearsofvoip.com/index.htm

ram
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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[ram]

>
>
> What about:
>
> 1) Message waiting notifications? Especially in a distributed system
> with multiple Asterisk servers?


Openser with Asterisk real time integration can do this job for you

2) Different codecs for different SIP users/accounts? DTMF modes? I
> know SER doesn't deal with the media at all, but if you let SER handle
> registrations and authentication, then I'd rather not keep track of
> codecs/DTMF on asterisk as well.


yes OpenSER does not handle media, but you can use dispatcher module to use
asterisk as a media relay or Media proxy or rtpproxy can do the job

ram
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Re: [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?

[Philipp Kempgen]

Arpit Mehta wrote:

> I wanted to know a simple way in which I could read some file from a console
> (say by using system command) and based on that either return true or false
> back to dialplan. Is there any built in command in Asterisk for that ?

AGI()

In the AGI script do something like
print "SET VARIABLE my_return_var 123"

In the dialplan:
Verbose(1,My AGI script returned ${my_return_var});


Regards,
  Philipp Kempgen

-- 
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Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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[asterisk-users] Get value from linux terminal to dialplan in Asterisk ?

[Arpit Mehta]

Hello Asterisk Users,

I wanted to know a simple way in which I could read some file from a console
(say by using system command) and based on that either return true or false
back to dialplan. Is there any built in command in Asterisk for that ?

What are the options do I have ? Are there any sample code to do so ?

Thanks a lot
Regards

-- 
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
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Re: [asterisk-users] Two PRI setup questions

[Erik Anderson]

Yep - as Doug mentioned, give esf framing and national switchtype a try.

I have a PRI from AT&T in one of my offices, and use this setup.

-erik

On 11/1/07, Lutgring, Sam <[EMAIL PROTECTED]> wrote:
>
>
>
> I am in the process of implementing a new ISDN pri and have a couple of
> questions.  This is a full 24 channels (23 B and 1 D) delivered over a T1
> interface.  The interface looks good and is not showing any errors.  Any
> help that you can provide would be greatly appreciated.
>
> 1)  What switchtype should be configured in the zapata.conf file when AT&T
> is using CUSTOM?  My understanding is that this equates to the dms100 in
> Asterisk, is this right?  The D channel is coming up just fine, but AT&T
> tells me that they cannot see the B channels.  When I try to make a call I
> get a slow busy and the debug shows an ISDN cause code of 34, no circuit
> available.
>
> 2)  Is there a way to see the idle status of a B channel?  When AT&T tells
> me they don't see the B channels coming up, is there a way that I can see
> this in Asterisk???
>
> Thanks in advance.
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-- 
Erik Anderson
http://andersonfam.org

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Re: [asterisk-users] Intercom with Snom phones

[Doug]

At 15:20 11/1/2007, Zaheer Master wrote:
 >I am currently waiting on a SIP Trunk to be set up for my company. In the
 >mean time, can I setup my phones on asterisk server so I can try the
 >intercom function? I'd just like to be able to talk on the local network to
 >test the sound quality.
 >
 >I've added this to my extensions.conf:
 >[101]
 >  type=friend
 >  context=phones
 >  host=dynamic
 >  secret=1234
 >
 >*For both [101] and [102]
 >
 >And this to my sip.conf:
 >[101]
 >  callerid="Zaheer Master" <5168224466>
 >  type=friend
 >  nat=yes
 >  secret=1234
 >  context=default
 >  host=dynamic
 >  qualify=yes
 >
 >Am I missing something else? Thanks in advance for the help!
 >
 >Regards,
 >Zaheer

 


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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[Luki]

> Some caveats (which may be different for OpenSER, so someone else can
> chime in):

What about:

1) Message waiting notifications? Especially in a distributed system
with multiple Asterisk servers?

2) Different codecs for different SIP users/accounts? DTMF modes? I
know SER doesn't deal with the media at all, but if you let SER handle
registrations and authentication, then I'd rather not keep track of
codecs/DTMF on asterisk as well.

Those two have been bugging me most.

Luki

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Re: [asterisk-users] Two PRI setup questions

[Doug Lytle]

Lutgring, Sam wrote:
>
> 1)  What switchtype should be configured in the zapata.conf file when 
> AT&T is using CUSTOM?  My understanding is that this equates to the 
> dms100 in Asterisk, is this right?  The D channel is coming up just 
> fine, but AT&T tells me that they cannot see the B channels.  When I 
> try to make a call I get a slow busy and the debug shows an ISDN cause 
> code of 34, no circuit available.

Most PRI setups in the US that I've delt with are esf.  My setup below:

#
# Zaptel Configuration File

span=1,1,0,esf,b8zs
defaultzone=us
loadzone=us
bchan=1-23
dchan=24


Zapata Configuration file
switchtype=national
context=pri
signalling=pri_cpe
group=1
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=-1.0
txgain=-4.0
busydetect=no
callprogress=no
pridialplan=unknown
usercallerid=yes
callerid=asreceived
channel => 1-23

> 2)  Is there a way to see the idle status of a B channel?  When AT&T 
> tells me they don't see the B channels coming up, is there a way that 
> I can see this in Asterisk???

You'll see the b channels restart when the first come up:

[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/1 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/2 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/3 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/4 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/5 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/6 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/7 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/8 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/9 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/10 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/11 
successfully restarted on span 1
[Oct 23 04:23:53] VERBOSE[17121] logger.c: -- B-channel 0/12 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/13 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/14 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/15 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/16 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/17 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/18 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/19 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/20 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/21 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/22 
successfully restarted on span 1
[Oct 23 04:23:54] VERBOSE[17121] logger.c: -- B-channel 0/23 
successfully restarted on span 1



-- 
Ben Franklin quote:

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



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[asterisk-users] Two PRI setup questions

[Lutgring, Sam]

I am in the process of implementing a new ISDN pri and have a couple of 
questions.  This is a full 24 channels (23 B and 1 D) delivered over a T1 
interface.  The interface looks good and is not showing any errors.  Any help 
that you can provide would be greatly appreciated.
 
1)  What switchtype should be configured in the zapata.conf file when AT&T is 
using CUSTOM?  My understanding is that this equates to the dms100 in Asterisk, 
is this right?  The D channel is coming up just fine, but AT&T tells me that 
they cannot see the B channels.  When I try to make a call I get a slow busy 
and the debug shows an ISDN cause code of 34, no circuit available.
 
2)  Is there a way to see the idle status of a B channel?  When AT&T tells me 
they don't see the B channels coming up, is there a way that I can see this in 
Asterisk???
 
Thanks in advance.
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Re: [asterisk-users] ring group containing external 10-digit numbers

[Don Pobanz]

Ryan Stille wrote: 
> I have a ring group setup that I'd like to ring a bunch of local 
> extensions, plus a few outside lines.  I want recipients to 
> confirm the call by pressing 1 before they are connected.  
> But when I add in an external number to this ring group (such 
> as 5551212#), none of my internal extensions ring any more.
> 
> Any ideas?

You did not indicate how you were connected to the outside world but if you are 
connected through fxo channels,  the line would be answered by the phone 
company very quickly. Once 1 phone is answered, I believe all the others would 
stop ringing. Could this be the issue? 

Don Pobanz
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[asterisk-users] OpenSER for Asterisk Load balance

[Edgar Guadamuz]

Hi, guys

I´ve just seen thta OpenSER can be coupled with Asterisk for load
balance, with the dispatcher module, something like this:

dispatcher.cfg file

# group  sip addresses of your * units
1 sip:10.1.2.3:5060
1 sip:10.1.2.4:5060
1 sip:10.1.2.5:5060

the basic openser.cfg should be like:

loadmodule("dispatcher.so")

...

if ( method=="INVITE" ) {
# dst_select( "GROUP", "HASH METHOD")
 dst_select("1","4");
 sl_send_reply("100","Trying");
 forward(uri:host, uri:port);
 exit();
}

That´s OK, but what about failover, I mean, if a Asterisk box crashes,
the dispatcher module will continue sending requests for that IP and
in this case something like heartbeat had to be implemented to take
the failed IP, but it would be more efficient if we could have OpenSER
monitoring the Asterisk servers health, anyone knows how?

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[asterisk-users] Polycom Park Button

[Kelly Opal]

Hi
I have a Polycom 501 phone. I set the park feature to 1 in sip.cfg and
the button shows up just fine. However when you press it it does
nothing. I have the t and T in the dial string. Is there some trick to
getting it to work with asterisk 1.4.

Thanks

Kelly


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[asterisk-users] ring group containing external 10-digit numbers

[Ryan Stille]

I have a ring group setup that I'd like to ring a bunch of local 
extensions, plus a few outside lines.  I want recipients to confirm the 
call by pressing 1 before they are connected.  But when I add in an 
external number to this ring group (such as 5551212#), none of my 
internal extensions ring any more.

Any ideas?

Thanks,
-Ryan


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Re: [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered

[Philipp Kempgen]

Costa Dinoteli wrote:

> We created a dial plan which performs and outbound dial out and
> deliveres a message to a receipient
> What call method/option in extensions or anywhere allow us to
> determine the length of the message.
> IE, what if a 3 minute message is attempted to be delivered and
> 45seconds into the call drops or is terminated by the end user.
> I need to know that the entire message was/was not delivered.
> 
> What I'm looking for is a variable that can be
> used in the exten file to determine call length of the
> completed call.

Maybe ${ANSWEREDTIME} is what you're looking for.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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[asterisk-users] Solved: Outgoing PRI CID?

[Turbo Fredriksson]

Quoting "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>:

> Remember Caller*ID Number is either country code + area code + number or 
> area code + number.  You never put a 1 or 0 at the beginning of the 
> number.  CallerID Number also can not have spaces, dashes, or other crud.

Darn! This last part (area code + number) solved it! My Telco 
lied to me (in retrospect, I'm not surprised - the're known to lie :).

Thanx for putting me on the right track...

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Re: [asterisk-users] Outgoing PRI CID?

[Turbo Fredriksson]

Quoting "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>:

> ;exten => s,n,Set(CALLERID(num)=${CALLERID(num)})
>
> This is the same as saying Set(FRED=${FRED}) 

I know. I've been playing with setting the CALLERID(num), and this was
my last, desperate attempt.

> exten => _X.,1,Dial(IAX2/graham/${EXTEN},30,r)
>
> DO NOT EVER USE THE "r" OPTION TO DIAL UNLESS YOU KNOW WHAT YOU ARE 
> DOING

Ok, thanx.

> Remember Caller*ID Number is either country code + area code + number or 
> area code + number.  You never put a 1 or 0 at the beginning of the 
> number.  CallerID Number also can not have spaces, dashes, or other crud.

My provider is sending ME the last tree digits of the phone number, and
they want three numbers from me. The problem is that I don't know how
to send these three digits...

Here I'm trying to fake the CALLERID(num) etc to something, just to test
(I'm all alone at the office now - the other wimps are home in bed :).

- s n i p -
[macro-dial]
exten => s,1,NoOp(Trying extension/number: ${ARG1} from ${CALLERID(all)})

exten => s,n,SetCallerPres(allowed)
exten => s,n,Set(CALLERID(num)=520)
exten => s,n,Set(CALLERID(ANI)=${CALLERID(num)})

exten => s,n,NoOp(CALLINGPRES 1: ${CALLINGPRES})
exten => s,n,Set(CALLINGPRESS=${CALLERID(num)})
exten => s,n,NoOp(CALLINGPRES 2: ${CALLINGPRES}) 

exten => s,n,Dial(${GLOBAL(PRI)}/${ARG1},${RINGTIME})   ; exten => 
s,n,Dial(SIP/[EMAIL PROTECTED],${RINGTIME},r)
exten => s,n,Playback(connection-failed)
exten => s,n,Congestion()
- s n i p -

This results in (doctored):
- s n i p -
-- Executing [@default:1] Macro("IAX2/graham-1", 
"dial||30|r") in new stack
-- Executing [EMAIL PROTECTED]:1] NoOp("IAX2/graham-1", "Trying 
extension/number:  from "Turbo Fredriksson" <528>") in new stack
-- Executing [EMAIL PROTECTED]:2] SetCallerPres("IAX2/graham-1", "allowed") 
in new stack
-- Executing [EMAIL PROTECTED]:3] Set("IAX2/graham-1", "CALLERID(num)=520") 
in new stack
-- Executing [EMAIL PROTECTED]:4] Set("IAX2/graham-1", "CALLERID(ANI)=520") 
in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp("IAX2/graham-1", "CALLINGPRES 1: 3") 
in new stack
-- Executing [EMAIL PROTECTED]:6] Set("IAX2/graham-1", "CALLINGPRESS=520") 
in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp("IAX2/graham-1", "CALLINGPRES 2: 3") 
in new stack
-- Executing [EMAIL PROTECTED]:8] Dial("IAX2/graham-1", 
"Zap/g1/|120|r") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/
-- Zap/1-1 is proceeding passing it to IAX2/graham-1
-- Hungup 'Zap/1-1'
  == Spawn extension (macro-dial, s, 8) exited non-zero on 'IAX2/graham-1' in 
macro 'dial'
  == Spawn extension (macro-dial, s, 8) exited non-zero on 'IAX2/graham-1'
-- Hungup 'IAX2/graham-1'
- s n i p -

Note that '528' is the real extension, '520' is the fake one. It doesn't
matter - I still get a call from '500'.
And trying to set the 'CALLINGPRES' didn't help (read-only).

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[asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered

[Costa Dinoteli]

Hi,

We created a dial plan which performs and outbound dial out and
deliveres a message to a receipient
What call method/option in extensions or anywhere allow us to
determine the length of the message.
IE, what if a 3 minute message is attempted to be delivered and
45seconds into the call drops or is terminated by the end user.
I need to know that the entire message was/was not delivered.

What I'm looking for is a variable that can be
used in the exten file to determine call length of the
completed call.

Thanks in advance,
-C

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Re: [asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)

[Eric "ManxPower" Wieling]

The fact that you are sending "528" as your Caller*ID might be a problem 
for your carrier.

Turbo Fredriksson wrote:
> Quoting mail-lists <[EMAIL PROTECTED]>:
> 
>> Turbo Fredriksson wrote:
>>> We have now got our new PRI line (10 channels, 100 numbers) connected
>>> and everything is working except the outgoing caller ID. Whatever
>>> SIP phone I'm using, the CID that's shown is the very first number...
> 
> Enabling PRI debugging shows the following:
> 
> - s n i p -
> 1 > [6c 05 21 81 35 32 38]
> 1 > Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> 1 >   Presentation: Presentation permitted, user 
> number passed network screening (1)  '528' ]
> - s n i p -
> 
> I _assume_ (maybe wrongly!) that this '0' is the ${CALLINGPRES}
> variable. If I print this just before the Dial(), this is indeed
> zero...
> 
> But it seems to be read-only... Any other way to send the extension
> down to the other end of the PRI line?
> 
> 
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[asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)

[Turbo Fredriksson]

Quoting mail-lists <[EMAIL PROTECTED]>:

> Turbo Fredriksson wrote:
>> We have now got our new PRI line (10 channels, 100 numbers) connected
>> and everything is working except the outgoing caller ID. Whatever
>> SIP phone I'm using, the CID that's shown is the very first number...

Enabling PRI debugging shows the following:

- s n i p -
1 > [6c 05 21 81 35 32 38]
1 > Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
1 >   Presentation: Presentation permitted, user number 
passed network screening (1)  '528' ]
- s n i p -

I _assume_ (maybe wrongly!) that this '0' is the ${CALLINGPRES}
variable. If I print this just before the Dial(), this is indeed
zero...

But it seems to be read-only... Any other way to send the extension
down to the other end of the PRI line?


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Re: [asterisk-users] Outgoing PRI CID?

[Eric "ManxPower" Wieling]

It is up to your carrier to permit you to send different Caller*ID. 
Many carriers that let you send your own Caller*ID number (you can't 
send Caller*ID Name) but if you send an invalid number, they will drop 
the call or override the Caller*ID number with the primary number of 
your PRI.

;exten => s,n,Set(CALLERID(num)=${CALLERID(num)})

This is the same as saying Set(FRED=${FRED})  It makes no sense to 
assign the value of a variable to that same variable.  I understand you 
have this commented out, but I wanted to mention it.

exten => _X.,1,Dial(IAX2/graham/${EXTEN},30,r)

DO NOT EVER USE THE "r" OPTION TO DIAL UNLESS YOU KNOW WHAT YOU ARE 
DOING  Asterisk will provide ringing sound to the caller by default. 
  "r" overrides audio messages, busy sounds, "the cellular caller you 
are trying to reach is not available" messages, etc with a ringing sound.

Remember Caller*ID Number is either country code + area code + number or 
area code + number.  You never put a 1 or 0 at the beginning of the 
number.  CallerID Number also can not have spaces, dashes, or other crud.




Turbo Fredriksson wrote:
> We have now got our new PRI line (10 channels, 100 numbers) connected
> and everything is working except the outgoing caller ID. Whatever
> SIP phone I'm using, the CID that's shown is the very first number...
> 
> - s n i p -
> [default]
> include => outgoing
> include => priin
> 
> [outgoing]
> exten => _NX.,1,Macro(dial,08${EXTEN},${RINGTIME})  ; Local 
> number (w/o areacode) - Stockholm
> exten => _0NX.,1,Macro(dial,${EXTEN},30,r)
> 
> [priin]
> exten => _X.,1,Dial(IAX2/graham/${EXTEN},30,r)
> 
> [macro-dial]
> exten => s,1,NoOp(Trying extension/number: ${ARG1} from ${CALLERID(num)})
> ;exten => s,n,Set(CALLERID(num)=${CALLERID(num)})
> exten => s,n,Dial(Zap/g1/${ARG1},${RINGTIME},r)
> exten => s,n,Playback(connection-failed)
> exten => s,n,Congestion()
> - s n i p -
> 
> 
> This * is only for PRI connection. The actual routing is done
> in an * installation running under a XEN domain...
> 
> Incoming works exactly as planed. So is the 'macro-dial'
> with the exception that the numer shown in the receiving
> end is '500' (the switchboard). This even if/when I call
> from '528'...
> 
> 
> Any ideas?
> 
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Re: [asterisk-users] DST

[BJ Weschke]

Turbo Fredriksson wrote:
> Quoting "Joe Acquisto" <[EMAIL PROTECTED]>:
>
>   
>> My thanks to all.  Problem resolved with the assistance.
>> 
>
> Would be nice if you posted HOW it was fixed to... I have this exact
> same problem at home, but the work phones displays time correctly...
>
>   
If you've got the files centrally managed, you can update the correct 
tags in sip.cfg to correct the situation.

These are the "correct" settings for regions affected by the new DST regs:

tcpIpApp.sntp.daylightSavings.enable="1" 
tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" 
tcpIpApp.sntp.daylightSavings.start.month="3" 
tcpIpApp.sntp.daylightSavings.start.date="8" 
tcpIpApp.sntp.daylightSavings.start.time="2" 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" 
tcpIpApp.sntp.daylightSavings.stop.month="11" 
tcpIpApp.sntp.daylightSavings.stop.date="1" 
tcpIpApp.sntp.daylightSavings.stop.time="2" 
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" 
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="0"


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Re: [asterisk-users] DST

[Joe Acquisto]

>>> On 11/1/2007 at 4:22 PM, Turbo Fredriksson <[EMAIL PROTECTED]> wrote:
> Quoting "Joe Acquisto" <[EMAIL PROTECTED]>:
> 
>> My thanks to all.  Problem resolved with the assistance.
> 
> Would be nice if you posted HOW it was fixed to... I have this exact
> same problem at home, but the work phones displays time correctly...
> 

Sorry, did not want to take up more list space.

To quote/snip/paste, from a very recent post (BJ Weschke) (and archives, 
polycom, etc) -:

***
If you've got the files centrally managed, you can update the correct 
tags in sip.cfg to correct the situation.

These are the "correct" settings for regions affected by the new DST regs:

tcpIpApp.sntp.daylightSavings.enable="1" 
tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" 
tcpIpApp.sntp.daylightSavings.start.month="3" 
tcpIpApp.sntp.daylightSavings.start.date="8" 
tcpIpApp.sntp.daylightSavings.start.time="2" 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" 
tcpIpApp.sntp.daylightSavings.stop.month="11" 
tcpIpApp.sntp.daylightSavings.stop.date="1" 
tcpIpApp.sntp.daylightSavings.stop.time="2" 
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" 
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="0"

***
joe a.


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Re: [asterisk-users] AEL2 and Callbacks

[Sean Bright]

Do a 'core show dialplan' and see what the AEL is generating.

On 11/1/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
> >- Original Message 
> >From: Richard Lyman <[EMAIL PROTECTED]>
> >To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> >Sent: Thursday, November 1, 2007 8:47:28 AM
> >Subject: Re: [asterisk-users] AEL2 and Callbacks
> >
> >Douglas Garstang wrote:
> >> I am originating a command via the AMI with this...
> >>
> >> Action: Login
> >> Username: xxx
> >> Secret: yyy
> >>
> >> ACTION: Originate
> >> Async: yes
> >> Timeout: 6
> >> Exten: callback
> >> Channel: Local/[EMAIL PROTECTED]
> >> Callerid: 849120
> >> Context: default
> >> ActionID: 849120
> >>
> >> My LegA context:
> >> ---
> >> context LegA {
> >>_X. => {
> >>Dial(SIP/[EMAIL PROTECTED]);
> >>}
> >>
> >> }
> >>
> >> And my default context:
> >> --
> >> context default {
> >>callback => {
> >>NoCDR();
> >>Wait(1);
> >>
> Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
> >>}
> >>
> >> }
> >>
> >> The A leg is established, and once Asterisk goes to dial the B leg...
> >>
> >>-- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2",
> "SIP/[EMAIL PROTECTED]") in new stack
> >>-- Called [EMAIL PROTECTED]
> >>-- SIP/Provider-09a8cff8 is making progress passing it to
> Local/[EMAIL PROTECTED],2
> >>
>  >   -- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
> >>  == Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed
> so falling back to exten 's'
> >>  == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed
> so falling back to context 'default'
> >> [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel '
> Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context
> 'default', >but no invalid handler
> >>
> >> Uhm, why? I have a default context with a callback extension. Of course
> I have no explicit priority 1 though... this is AEL2
> >> What's it complaining for?
> >>
> >> Doug.
> >>
> >>
> >>
> >originates have always had an issue where it falls back to an 's'
> >extension.  and since you do not have one, nor an 'i' for invalid
> >extension... it bombs out.
>
> Yes... but I DO have a default context and I DO have a callback extension.
> What's it whining about?
>
> Doug.
>
>
>
>
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Re: [asterisk-users] DST

[Martin Smith]

In case it helps, I've fixed that problem before by making sure my DHCP
server gives out the correct time AND offset fields, and updating the
Polycom firmware to a recent version :).

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Turbo Fredriksson
> Sent: Thursday, November 01, 2007 4:22 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] DST
> 
> Quoting "Joe Acquisto" <[EMAIL PROTECTED]>:
> 
> > My thanks to all.  Problem resolved with the assistance.
> 
> Would be nice if you posted HOW it was fixed to... I have this exact
> same problem at home, but the work phones displays time correctly...
> 
> > joe a.
> >
>  On 11/1/2007 at 1:43 PM, "Joe Acquisto" 
> <[EMAIL PROTECTED]> wrote:
> >> My Polycom phones are displaying time, off by one hour.  
> Seems they are on 
> >> the old DST rules.  How do I fix this?
> >> 
> >> joe a.
> >> 
> >> 
> >> ___
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> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
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> -- 
> Uzi spy attack North Korea SEAL Team 6 counter-intelligence president
> Ortega fissionable $400 million in gold bullion Cuba jihad Soviet
> Delta Force radar
> [See http://www.echelonwatch.org/ for more about this]
> http://www.theregister.co.uk/2001/09/06/eu_releases_echelon_sp
> ying_report/
> http://www.aclu.org/safefree/nsaspying/23989res20060131.html#echelon
> 
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Re: [asterisk-users] DST

[Turbo Fredriksson]

Quoting "Joe Acquisto" <[EMAIL PROTECTED]>:

> My thanks to all.  Problem resolved with the assistance.

Would be nice if you posted HOW it was fixed to... I have this exact
same problem at home, but the work phones displays time correctly...

> joe a.
>
 On 11/1/2007 at 1:43 PM, "Joe Acquisto" <[EMAIL PROTECTED]> wrote:
>> My Polycom phones are displaying time, off by one hour.  Seems they are on 
>> the old DST rules.  How do I fix this?
>> 
>> joe a.
>> 
>> 
>> ___
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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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>
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-- 
Uzi spy attack North Korea SEAL Team 6 counter-intelligence president
Ortega fissionable $400 million in gold bullion Cuba jihad Soviet
Delta Force radar
[See http://www.echelonwatch.org/ for more about this]
http://www.theregister.co.uk/2001/09/06/eu_releases_echelon_spying_report/
http://www.aclu.org/safefree/nsaspying/23989res20060131.html#echelon

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[asterisk-users] Intercom with Snom phones

[Zaheer Master]

I am currently waiting on a SIP Trunk to be set up for my company. In the
mean time, can I setup my phones on asterisk server so I can try the
intercom function? I'd just like to be able to talk on the local network to
test the sound quality. 

I've added this to my extensions.conf:
[101]
  type=friend
  context=phones
  host=dynamic
  secret=1234

*For both [101] and [102]

And this to my sip.conf:
[101]
  callerid="Zaheer Master" <5168224466>
  type=friend
  nat=yes
  secret=1234
  context=default
  host=dynamic
  qualify=yes

Am I missing something else? Thanks in advance for the help!

Regards,
Zaheer


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Re: [asterisk-users] Outgoing PRI CID?

[Turbo Fredriksson]

Quoting "Anciso, Roy" <[EMAIL PROTECTED]>:

> I do this to tie extensions to a particular number:
>
> exten => _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD<2317231516>)
> exten => _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology<2317234264>)

Tried that but couldn't get it to work. I've tried all the CALLERID()
values, but none worked...

>> ;exten => s,n,Set(CALLERID(num)=${CALLERID(num)})
-- 
radar FSF iodine AK-47 spy PLO Khaddafi supercomputer colonel
plutonium class struggle assassination strategic KGB CIA
[See http://www.echelonwatch.org/ for more about this]
http://www.theregister.co.uk/2001/09/06/eu_releases_echelon_spying_report/
http://www.aclu.org/safefree/nsaspying/23989res20060131.html#echelon

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Re: [asterisk-users] DST

[Joe Acquisto]

My thanks to all.  Problem resolved with the assistance.

joe a.

>>> On 11/1/2007 at 1:43 PM, "Joe Acquisto" <[EMAIL PROTECTED]> wrote:
> My Polycom phones are displaying time, off by one hour.  Seems they are on 
> the old DST rules.  How do I fix this?
> 
> joe a.
> 
> 
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Re: [asterisk-users] Outgoing PRI CID?

[Turbo Fredriksson]

> "mail-lists" == mail-lists  <[EMAIL PROTECTED]> writes:

mail-lists> I don't know if the same is true for you but we had to
mail-lists> call our telco and have them set our callerid settings
mail-lists> to 'station level'. Not sure if your telco offers this
mail-lists> but they should.

Apparently this is already done... I'll look into the 'station level'
value though. Thanx.
-- 
genetic Waco, Texas NORAD Marxist spy cracking assassination nuclear
congress pits Ft. Meade DES kibo Soviet attack
[See http://www.echelonwatch.org/ for more about this]
http://www.theregister.co.uk/2001/09/06/eu_releases_echelon_spying_report/
http://www.aclu.org/safefree/nsaspying/23989res20060131.html#echelon

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[asterisk-users] Chanspy attaching to a caller ID entry?

[Martin Smith]

Hi folks,

We have a very rare problem with Asterisk 1.2 where Chanspy reports the
following:
 Oct 31 19:53:29 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010
to
 SIP/2015-11b21148
 Oct 31 19:53:42 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010
to
 SIP/2066-11d42b80
 Oct 31 19:53:46 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010
to
 SIP/2022-11c10170
 Oct 31 19:53:58 NOTICE[10982] rtp.c: Unknown RTP codec 126 received
 Oct 31 19:53:58 NOTICE[10982] rtp.c: Unknown RTP codec 126 received
 Oct 31 19:53:58 NOTICE[10982] rtp.c: Unknown RTP codec 126 received
 Oct 31 19:54:01 NOTICE[21189] app_chanspy.c: Attaching SIP/106-491611a8
to
 SIP/2033-11b6e300
 Oct 31 19:54:09 NOTICE[10273] app_chanspy.c: Attaching
SIP/2014-4853cb40
 to "University of Florida" <3523922908>

The last line results in a full on crash without a core dump. This is a
SIP phone going through an extension setting CID but before a Dial(),
and another SIP phone ChanSpy'ing them. We have ChanSpy set to only spy
on bridged channels as well.

Has anyone seen a bug like this? We often top 10,000 calls a month, and
this only happens maybe twice a year. We're switching to 1.4 soon, but
just curious :)

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 


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Re: [asterisk-users] Outgoing PRI CID?

[Anciso, Roy]

I do this to tie extensions to a particular number:

exten => _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD<2317231516>)
exten => _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology<2317234264>)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Turbo
Fredriksson
Sent: Thursday, November 01, 2007 2:32 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Outgoing PRI CID?

We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...

- s n i p -
[default]
include => outgoing
include => priin

[outgoing]
exten => _NX.,1,Macro(dial,08${EXTEN},${RINGTIME})  ; Local
number (w/o areacode) - Stockholm
exten => _0NX.,1,Macro(dial,${EXTEN},30,r)

[priin]
exten => _X.,1,Dial(IAX2/graham/${EXTEN},30,r)

[macro-dial]
exten => s,1,NoOp(Trying extension/number: ${ARG1} from
${CALLERID(num)})
;exten => s,n,Set(CALLERID(num)=${CALLERID(num)})
exten => s,n,Dial(Zap/g1/${ARG1},${RINGTIME},r)
exten => s,n,Playback(connection-failed)
exten => s,n,Congestion()
- s n i p -


This * is only for PRI connection. The actual routing is done
in an * installation running under a XEN domain...

Incoming works exactly as planed. So is the 'macro-dial'
with the exception that the numer shown in the receiving
end is '500' (the switchboard). This even if/when I call
from '528'...


Any ideas?

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[asterisk-users] AsteriskNOW and TDM800P

[Rafael Canchola]

Hi all

I sold new TDM800P card with 8 FXO ports, someone know if can be use 
this card on AsteriskNOW or trixbox?

What can i do for use this card?

Thanks.



--
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
[EMAIL PROTECTED]
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP 523663899
d00d! cyberalph
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Re: [asterisk-users] Outgoing PRI CID?

[mail-lists]

Turbo Fredriksson wrote:
> We have now got our new PRI line (10 channels, 100 numbers) connected
> and everything is working except the outgoing caller ID. Whatever
> SIP phone I'm using, the CID that's shown is the very first number...

I don't know if the same is true for you but we had to call our telco 
and have them set our callerid settings to 'station level'. Not sure if 
your telco offers this but they should.

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[asterisk-users] Outgoing PRI CID?

[Turbo Fredriksson]

We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...

- s n i p -
[default]
include => outgoing
include => priin

[outgoing]
exten => _NX.,1,Macro(dial,08${EXTEN},${RINGTIME})  ; Local number 
(w/o areacode) - Stockholm
exten => _0NX.,1,Macro(dial,${EXTEN},30,r)

[priin]
exten => _X.,1,Dial(IAX2/graham/${EXTEN},30,r)

[macro-dial]
exten => s,1,NoOp(Trying extension/number: ${ARG1} from ${CALLERID(num)})
;exten => s,n,Set(CALLERID(num)=${CALLERID(num)})
exten => s,n,Dial(Zap/g1/${ARG1},${RINGTIME},r)
exten => s,n,Playback(connection-failed)
exten => s,n,Congestion()
- s n i p -


This * is only for PRI connection. The actual routing is done
in an * installation running under a XEN domain...

Incoming works exactly as planed. So is the 'macro-dial'
with the exception that the numer shown in the receiving
end is '500' (the switchboard). This even if/when I call
from '528'...


Any ideas?

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Re: [asterisk-users] PRI commands missing...

[Tzafrir Cohen]

On Thu, Nov 01, 2007 at 12:09:50PM -0500, Matthew Fredrickson wrote:
> Arpit Mehta wrote:
> > This happens when your pri line is down. This has happened a couple of times
> > to me and the pri commands come back when the pri line is up. My guess is
> > that your pri line is down.
> 
> No.  The pri commands should be there even if the span goes down.  The 
> reason why the pri commands wouldn't be there is that chan_zap didn't 
> load for some reason.  Try starting asterisk with the - options 
> to see what chan_zap says when it tries to load.

Actually you need much less v-s should give less noise for the case
Asterisk actually does load.

-- 
   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] PRI over T1 calls dropping, cause 100

[Michelle Dupuis]

Matt (& All),

Thanks for the info below. You obviously responded to my bug report too
(thanks again) - so I'll keep my dialog on the list to benefit other users
too.

>From your response below and on the bug site, I'm starting to get the
picture.  I know the firmware on the Nortel is old, so I'm guessing that
libpri is sending something that the Nortel does not know how to handle.

Is there a way to dumb down what libpri sends?  From everything I've read
PRI is an evolving standard - and older devices may struggle with newer
extensions/developments.  (This might be very handy for users trying to talk
to old pbx's.)

Is there a workaround?

Thanks,
Michelle

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matthew Fredrickson
> Sent: Thursday, November 01, 2007 1:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] PRI over T1 calls dropping, cause 100
> 
> Michelle Dupuis wrote:
> > I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a 
> > Meridian Option 61C.  Calls either way drop with error 
> "Channel 0/23, span 1 got
> > hangup, cause 100".   Can anyone offer insight into the cause and
> > solution/workaround?  (I tried upgrading to Ast 1.4.13, and 
> upgrading 
> > matching zaptel & libpri, put the problem is identical).
> >
> > For testing, I tried a call from the Meridian to an extension which 
> > just plays the monkey sounds.  The call sets up, but I never hear 
> > monkeys (just a busy tone) as the call disconnects immediately.
> >
> > I have three attachments below:
> > 1. The CLI output showing a dropped call (inbound from 
> Meridian1) 2. A 
> > PRI DEBUG of the same call, showing "!! Invalid Protocol 
> Profile field 
> > 0x11" in the trace.
> > 3. A PRI INTENSE DEBUG SPAN 1 of the same call.  (I tried 
> upgrading to 
> > Ast
> > 1.4.13 prior to this call trace - same result)
> >
> > From what I can see in the trace, a Q.932 Interpretation 
> component is 
> > not handled.
> > Sowhat do I do?
> 
> The Q.932 part of that and invalid protocol profile field 
> shouldn't be your problem.  It looks like the switch is 
> rejecting it for this reason:
> 
> < Protocol Discriminator: Q.931 (8)  len=8 < Call Ref: len= 1 
> (reference 23/0x17) (Originator) < Message type: RELEASE (77) 
> < [08 02 81 e4]
> < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 
> 0: 0   Location:
> Private network serving the local user (1)
> <  Ext: 1  Cause: Invalid information element contents
> (100), class = Protocol Error (6) ]
> 
> 
> Invalid information element contents.  You need to figure out 
> what the switch doesn't seem to be liking.  Looks like it 
> doesn't like the way one of the IEs is coded.
> 
> Matthew Fredrickson
> 
> >
> > Thanks,
> > MD
> >
> > =1===
> >
> > !! Invalid Protocol Profile field 0x11
> > -- Accepting call from '2004000' to '111' on channel 
> 0/23, span 1
> > -- Executing NoOp("Zap/23-1", "Incoming call from 
> Meridian1") in 
> > new stack
> > -- Executing NoOp("Zap/23-1", "  From number: 2004000| 
> revised to 
> > 2004") in new stack
> > -- Executing Set("Zap/23-1", "CALLERID(num)=2004") in new stack
> > -- Executing NoOp("Zap/23-1", "  From name: TELEPHONE ROOM") in 
> > new stack
> > -- Executing Macro("Zap/23-1", "dialfromMeridian|111") 
> in new stack
> > -- Executing NoOp("Zap/23-1", "Incoming call to number 
> 111| which 
> > maps to device: ") in new stack
> > -- Executing GotoIf("Zap/23-1", "1?NoMatch") in new stack
> > -- Goto (macro-dialfromMeridian,s,5)
> > -- Executing NoOp("Zap/23-1", "Invalid extension in macro
> > dialfromMeridian") in new stack
> > -- Executing Hangup("Zap/23-1", "") in new stack
> >   == Spawn extension (macro-dialfromMeridian, s, 6) exited 
> non-zero on 
> > 'Zap/23-1' in macro 'dialfromMeridian'
> >   == Spawn extension (macro-dialfromMeridian, s, 6) exited 
> non-zero on 
> > 'Zap/23-1'
> > -- Hungup 'Zap/23-1'
> > !! Invalid Protocol Profile field 0x11
> > -- Accepting call from '2004000' to '123' on channel 
> 0/23, span 1
> > -- Executing Answer("Zap/23-1", "") in new stack
> > -- Executing Playback("Zap/23-1", "tt-monkeys") in new stack
> > -- Playing 'tt-monkeys' (language 'en')
> > -- Channel 0/23, span 1 got hangup, cause 100
> >   == Spawn extension (entryocgexternal, 123, 2) exited non-zero on 
> > 'Zap/23-1'
> > -- Hungup 'Zap/23-1'
> >
> > =2===
> > *CLI> pri debug span 1
> > Enabled debugging on span 1
> > < Protocol Discriminator: Q.931 (8)  len=61 < Call Ref: len= 1 
> > (reference 23/0x17) (Originator) < Message type: SETUP (5) 
> < [04 03 80 
> > 90 a2] < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info 
> > transfer
> > capability: Speech (0)
> > <  Ext: 1  Trans mode/rate: 
> 64kbps, circu

Re: [asterisk-users] DST

[BJ Weschke]

Joe Acquisto wrote:
> My Polycom phones are displaying time, off by one hour.  Seems they are on 
> the old DST rules.  How do I fix this?
>
> joe a.
>
>   

 If you've got the files centrally managed, you can update the correct 
tags in sip.cfg to correct the situation.

 These are the "correct" settings for regions affected by the new DST regs:

 tcpIpApp.sntp.daylightSavings.enable="1" 
tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" 
tcpIpApp.sntp.daylightSavings.start.month="3" 
tcpIpApp.sntp.daylightSavings.start.date="8" 
tcpIpApp.sntp.daylightSavings.start.time="2" 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" 
tcpIpApp.sntp.daylightSavings.stop.month="11" 
tcpIpApp.sntp.daylightSavings.stop.date="1" 
tcpIpApp.sntp.daylightSavings.stop.time="2" 
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" 
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="0"


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Re: [asterisk-users] DST

[Dave Fullerton]

Joe Acquisto wrote:
> My Polycom phones are displaying time, off by one hour.  Seems they are on 
> the old DST rules.  How do I fix this?
> 
> joe a.
> 
> 

The archives are your friend:
http://lists.digium.com/pipermail/asterisk-users/2007-March/181696.html

-Dave

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Re: [asterisk-users] DST

[Philipp Kempgen]

Joe Acquisto wrote:
> My Polycom phones are displaying time, off by one hour.  Seems they are on 
> the old DST rules.  How do I fix this?

http://www.google.com/search?q=polycom+dst+rules
gives me some good results.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] DST

[Kevin P. Fleming]

Joe Acquisto wrote:
> My Polycom phones are displaying time, off by one hour.  Seems they are on 
> the old DST rules.  How do I fix this?

Update the firmware to a recent release, or copy the DST rules section
from the sip.cfg file from a recent release into your older version of
sip.cfg. If you aren't feeding the phones from a provisioning server,
then you may have to adjust it via the web interface on the phones
themselves.

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

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Re: [asterisk-users] AEL2 and Callbacks

[Douglas Garstang]

>- Original Message 
>From: Richard Lyman <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion 
>
>Sent: Thursday, November 1, 2007 8:47:28 AM
>Subject: Re: [asterisk-users] AEL2 and Callbacks
>

>Douglas Garstang wrote:
>> I am originating a command via the AMI with this...
>>
>> Action: Login
>> Username: xxx
>> Secret: yyy
>>
>> ACTION: Originate
>> Async: yes
>> Timeout: 6
>> Exten: callback
>> Channel: Local/[EMAIL PROTECTED]
>> Callerid: 849120
>> Context: default
>> ActionID: 849120
>>
>> My LegA context:
>> ---
>> context LegA {
>> _X. => {
>> Dial(SIP/[EMAIL PROTECTED]); 
>> }
>>
>> }
>>
>> And my default context:
>> --
>> context default {
>> callback => {
>> NoCDR();
>> Wait(1);
>>
 
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
>> }
>>
>> }
>>
>> The A leg is established, and once Asterisk goes to dial the B leg...
>>
>> -- Executing [EMAIL PROTECTED]:1]
 Dial("Local/[EMAIL PROTECTED],2", "SIP/[EMAIL PROTECTED]") in new stack
>> -- Called [EMAIL PROTECTED]
>> -- SIP/Provider-09a8cff8 is making progress passing it to
 Local/[EMAIL PROTECTED],2
>>
 >-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
>>   == Starting Local/[EMAIL PROTECTED],1 at default,callback,1
 failed so falling back to exten 's'
>>   == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still
 failed so falling back to context 'default'
>> [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
 'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in
 context 'default', >but no invalid handler
>>
>> Uhm, why? I have a default context with a callback extension. Of
 course I have no explicit priority 1 though... this is AEL2 
>> What's it complaining for?
>>
>> Doug.
>>
>>
>>   
>originates have always had an issue where it falls back to an 's' 
>extension.  and since you do not have one, nor an 'i' for invalid 
>extension... it bombs out.

Yes... but I DO have a default context and I DO have a callback extension. 
What's it whining about?

Doug.




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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[ram]

Ser or openser with asterisk can be possible

ram


On 11/1/07, Antoine Megalla <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I have a client who requires an Asterisk system with
> 1500 SIP clients.
> All clients will have ATAs (mostly Grandstream), so I
> think a single
> Asterisk server will not be able to handle all 1500
> registrations, plus
> typical applications like Voicemail, call forwarding,
> etc.. and the billing
> needs for all the clients.
>
> I have searched all over, and it seems that the
> perfect solution is using
> SER/OpenSER as registration server for the SIP
> clients, and then use
> Asterisk (one or more servers in load balancing mode)
> for everything else.
>
> The problem is that I cannot find any configuration
> files for such a setup.
> I can do all the Asterisk configuration, dial plan,
> AGIs, apps, etc.. but
> for SER/OpenSER I cannot find anything.
>
> Can anyone please point me in the right direction,
> provide me with OpenSER
> configuration, or any pointers on the subject. I tried
> to read all the
> material on how to write configuration files for
> OpenSER, but it is
> incomprehensible to me, and it is much harder that
> when I learning Asterisk
> 3 years ago.
>
> Your help is much appreciated.
>
> Regards,
>
> Antoine Megalla.
>
>
>
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[asterisk-users] DST

[Joe Acquisto]

My Polycom phones are displaying time, off by one hour.  Seems they are on the 
old DST rules.  How do I fix this?

joe a.


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Re: [asterisk-users] issues with downloads.digium.com

[Matthew Fredrickson]

Carlos Chavez wrote:
> On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote:
>> On a slightly different matter: 
>> http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
>> 1.4.1 .
>>
> 
>   Yes, I noticed that too and was wondering if it is just because they
> have not updated the site or if there is a problem with the newest
> versions and they do not want people to download them.

It's because the site hasn't been updated.  I will look into this...

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] PRI commands missing...

[Matthew Fredrickson]

Arpit Mehta wrote:
> This happens when your pri line is down. This has happened a couple of times
> to me and the pri commands come back when the pri line is up. My guess is
> that your pri line is down.

No.  The pri commands should be there even if the span goes down.  The 
reason why the pri commands wouldn't be there is that chan_zap didn't 
load for some reason.  Try starting asterisk with the - options 
to see what chan_zap says when it tries to load.

Matthew Fredrickson

> 
> Hope it helps.
> 
> Arpit
> 
> 
> On 10/31/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
>> I have an Asterisk server running Elastix but patched to use Unicall.
>> Everything seems to be working fine and the TE220 card is up and running
>> with
>> port 1 configured as PRI and port 2 as MFC/R2.  We can already send and
>> receive calls on port two but we cannot on port one.  That is when we
>> noticed
>> that there are no PRI commands available on the Asterisk CLI.  We cannot
>> use
>> PRI DEBUG SPAN to determine why port 1 is not receiving or sending calls.
>>
>> Why would this commands be missing?
>>
>> --
>> Carlos Chavez
>> Director de Tecnología
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Tel: +52-55-91169161 Ext 2001
>>
>>
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>>
> 
> 
> 
> 
> 
> 
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-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] PRI over T1 calls dropping, cause 100

[Matthew Fredrickson]

Michelle Dupuis wrote:
> I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian
> Option 61C.  Calls either way drop with error "Channel 0/23, span 1 got
> hangup, cause 100".   Can anyone offer insight into the cause and
> solution/workaround?  (I tried upgrading to Ast 1.4.13, and upgrading
> matching zaptel & libpri, put the problem is identical).
>  
> For testing, I tried a call from the Meridian to an extension which just
> plays the monkey sounds.  The call sets up, but I never hear monkeys (just a
> busy tone) as the call disconnects immediately.
> 
> I have three attachments below:
> 1. The CLI output showing a dropped call (inbound from Meridian1)
> 2. A PRI DEBUG of the same call, showing "!! Invalid Protocol Profile field
> 0x11" in the trace.
> 3. A PRI INTENSE DEBUG SPAN 1 of the same call.  (I tried upgrading to Ast
> 1.4.13 prior to this call trace - same result)
> 
> From what I can see in the trace, a Q.932 Interpretation component is not
> handled.  
> Sowhat do I do?

The Q.932 part of that and invalid protocol profile field shouldn't be 
your problem.  It looks like the switch is rejecting it for this reason:

< Protocol Discriminator: Q.931 (8)  len=8
< Call Ref: len= 1 (reference 23/0x17) (Originator)
< Message type: RELEASE (77)
< [08 02 81 e4]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location:
Private network serving the local user (1)
<  Ext: 1  Cause: Invalid information element contents
(100), class = Protocol Error (6) ]


Invalid information element contents.  You need to figure out what the 
switch doesn't seem to be liking.  Looks like it doesn't like the way 
one of the IEs is coded.

Matthew Fredrickson

>  
> Thanks,
> MD
>  
> =1===
>  
> !! Invalid Protocol Profile field 0x11
> -- Accepting call from '2004000' to '111' on channel 0/23, span 1
> -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new
> stack
> -- Executing NoOp("Zap/23-1", "  From number: 2004000| revised to 2004")
> in new stack
> -- Executing Set("Zap/23-1", "CALLERID(num)=2004") in new stack
> -- Executing NoOp("Zap/23-1", "  From name: TELEPHONE ROOM") in new
> stack
> -- Executing Macro("Zap/23-1", "dialfromMeridian|111") in new stack
> -- Executing NoOp("Zap/23-1", "Incoming call to number 111| which maps
> to device: ") in new stack
> -- Executing GotoIf("Zap/23-1", "1?NoMatch") in new stack
> -- Goto (macro-dialfromMeridian,s,5)
> -- Executing NoOp("Zap/23-1", "Invalid extension in macro
> dialfromMeridian") in new stack
> -- Executing Hangup("Zap/23-1", "") in new stack
>   == Spawn extension (macro-dialfromMeridian, s, 6) exited non-zero on
> 'Zap/23-1' in macro 'dialfromMeridian'
>   == Spawn extension (macro-dialfromMeridian, s, 6) exited non-zero on
> 'Zap/23-1'
> -- Hungup 'Zap/23-1'
> !! Invalid Protocol Profile field 0x11
> -- Accepting call from '2004000' to '123' on channel 0/23, span 1
> -- Executing Answer("Zap/23-1", "") in new stack
> -- Executing Playback("Zap/23-1", "tt-monkeys") in new stack
> -- Playing 'tt-monkeys' (language 'en')
> -- Channel 0/23, span 1 got hangup, cause 100
>   == Spawn extension (entryocgexternal, 123, 2) exited non-zero on
> 'Zap/23-1'
> -- Hungup 'Zap/23-1'
>  
> =2===
> *CLI> pri debug span 1
> Enabled debugging on span 1
> < Protocol Discriminator: Q.931 (8)  len=61
> < Call Ref: len= 1 (reference 23/0x17) (Originator)
> < Message type: SETUP (5)
> < [04 03 80 90 a2]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Speech (0)
> <  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
> (16)
> <  Ext: 1  User information layer 1: u-Law (34)
> < [18 04 e9 80 83 17]
> < Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Exclusive
> Dchan: 0
>  <   Ext: 1  DS1 Identifier: 0  
> <   Ext: 1  Coding: 0   Number Specified   Channel Type:
> 3
> <   Ext: 0  Channel: 23 ]
> < [1c 0a 11 be a1 06 02 01 01 02 01 01]
> < Facility (len=12, codeset=0) [ 0x11, 0xbe, 0xa1, 0x06, 0x02, 0x01, 0x01,
> 0x02, 0x01, 0x01 ]
> < [28 0f b1 54 45 4c 45 50 48 4f 4e 45 20 52 4f 4f 4d]
> < Display (len=15) Charset: 31 [ TELEPHONE ROOM ]
> < [6c 09 09 80 32 30 30 34 30 30 30]
> < Calling Number (len=11) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
> Private Numbering Plan (9)
> <   Presentation: Presentation permitted, user
> number not screened (0) '2004000' ]
> < [70 04 89 31 32 33]
> < Called Number (len= 6) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
> Private Numbering Plan (9) '123' ]
> -- Making new call for cr 23
> -- Processing Q.931 Call Setup
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs

Re: [asterisk-users] Digium Vs sangoma Hradware

[Stephen Bosch]

Michelle Dupuis wrote:
> And just for confusion, one of the guys I work with swears by Sangoma.  (I
> have not done a lot of T1 stuff personally...so maybe as your expertise
> grows Sangoma becomes a better fit).
> 
> Perhaps I should have voted for Heinz...

as for string -- go with fishing line.

-Stephen-

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[asterisk-users] libpri & tie line vs trunk

[Michelle Dupuis]

We are connecting an asterisk box to a Nortel Option 61 via a T1 with PRI.
We have hit a problem we cannot overcome; specifically, the Nortel is asking
for the ROSE information element (IE) over the PRI connection.  This causes
libpri to drop the connection, with cause 100.  The Nortel cannot "turn off"
asking for ROSE.
 
Since our Nortel box does not support QSIG (too old), we have to stick with
"dms100" switchtype in zapata.  (If that's wrong please tell me).  
 
So, the only thing left we can change is using a trunk instead of a tie line
(current setup), to connect the PBX's.  Would using a trunk avoid the PBX
sending the ROSE IE request?  Do tie lines somehow imply Nortel-to-Nortel
(is that why the Nortel is asking for ROSE)?
 
Any other options?  Is changing the libpri code to ignore the ROSE request
feasible?
 
I would greatly appreciate any help I can get!  
 
Many thanks!
Michelle
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Re: [asterisk-users] Help

[Jarga Jallow]

 GXP2000 Firmware/sofware version is 1.1.4.18

Jarga 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
'Xenon' Hanson
Sent: Thursday, November 01, 2007 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help

Jarga Jallow wrote:
> I need help with my grand stream GXP2000 phones they keep freezing 
> randomly. Any ideas?

   What firmware revision?

   Want to buy a used one from me? I'm trying to standardize on Sipura
841s, and I have 
one GXP2000.

> Jarga

-- 
  Chris 'Xenon' Hanson | Xenon @ 3D Nature |
http://www.3DNature.com/
  "I set the wheels in motion, turn up all the machines, activate the
programs,
   and run behind the scenes. I set the clouds in motion, turn up light
and sound,
   activate the window, and watch the world go 'round." -Prime Mover,
Rush.

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Re: [asterisk-users] Help

[Chris 'Xenon' Hanson]

Jarga Jallow wrote:
> I need help with my grand stream GXP2000 phones they keep freezing 
> randomly. Any ideas?

   What firmware revision?

   Want to buy a used one from me? I'm trying to standardize on Sipura 841s, 
and I have 
one GXP2000.

> Jarga

-- 
  Chris 'Xenon' Hanson | Xenon @ 3D Nature | http://www.3DNature.com/
  "I set the wheels in motion, turn up all the machines, activate the programs,
   and run behind the scenes. I set the clouds in motion, turn up light and 
sound,
   activate the window, and watch the world go 'round." -Prime Mover, Rush.

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[asterisk-users] RTP Read too short

[John Faubion]

Hello,
I'm getting the following logs:

[Nov  1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too
short
[Nov  1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too
short
[Nov  1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too
short

Anyone know how to correct this? I'm using SIPConnect from CBeyond and this
appears on incoming calls. I haven't had any complaints about voice quality
and I haven't seen any dropped calls. Should I be concerned?

Thanks,
John Faubion




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Re: [asterisk-users] AEL2 and Callbacks

[Richard Lyman]

Douglas Garstang wrote:
> I am originating a command via the AMI with this...
>
> Action: Login
> Username: xxx
> Secret: yyy
>
> ACTION: Originate
> Async: yes
> Timeout: 6
> Exten: callback
> Channel: Local/[EMAIL PROTECTED]
> Callerid: 849120
> Context: default
> ActionID: 849120
>
> My LegA context:
> ---
> context LegA {
> _X. => {
> Dial(SIP/[EMAIL PROTECTED]); 
> }
>
> }
>
> And my default context:
> --
> context default {
> callback => {
> NoCDR();
> Wait(1);
> 
> Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
> }
>
> }
>
> The A leg is established, and once Asterisk goes to dial the B leg...
>
> -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2", 
> "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/Provider-09a8cff8 is making progress passing it to Local/[EMAIL 
> PROTECTED],2
> -- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
>   == Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed so 
> falling back to exten 's'
>   == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed so 
> falling back to context 'default'
> [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel 
> 'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context 
> 'default', but no invalid handler
>
> Uhm, why? I have a default context with a callback extension. Of course I 
> have no explicit priority 1 though... this is AEL2 
> What's it complaining for?
>
> Doug.
>
>
>   
originates have always had an issue where it falls back to an 's' 
extension.  and since you do not have one, nor an 'i' for invalid 
extension... it bombs out.



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[asterisk-users] Help

[Jarga Jallow]

  

I need help with my grand stream GXP2000 phones they keep freezing
randomly. Any ideas?

 

Jarga

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Re: [asterisk-users] Connection astrisk to a RAS (portmaster)

[Nicolas Ross]

Thanks,

That's not it.

I juste uninstalled wanpipe, redownloaded zaptel (another version (1.4.5.1) 
to try), re-installed wanpipe, (patching zaptel in the process), recompile 
wanpipe, re-compiled zaptel, recompiled asterisk, re-installed everything, 
and still got the same errors...

Nicolas


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

Sent: Thursday, November 01, 2007 9:15 AM
Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster)


> On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote:
>> I also get sometime :
>>
>>   == Primary D-Channel on span 2 down
>> [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No
>> D-channels available!  Using Primary channel 48 as D-channel anyway!
>>   == Primary D-Channel on span 2 up
>
> I don't claim to be an expert on the Wanpipe drivers, but it's been my
> experience that if your D-channels bounce up and down like that every
> few seconds, that the Wanpipe drivers didn't successfully patch Zaptel,
> or you haven't restarted Zaptel since the Wanpipe drivers patched
> Zaptel.
>
> Hopefully that's enough to get you up and running.
>
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
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[asterisk-users] Parking speed

[Kelly Opal]

Hi
Is it possible to speed up the parking process. We receive a lot of calls 
during the day and it gets to be painful to wait for the call to be parked and 
the number to be played back to you. I would like to speed up the whole process 
including the play back speed.

Thanks 

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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[SIP]

While I can't help with OpenSER (OpenSER and SER are very different 
these days), the SER config info for SER 0.9.6 can be found in:

http://siprouter.onsip.org/doc/gettingstarted/

That will also help you to understand the config methodology.

The BASIC meshing Asterisk with SER is pretty straightforward. SER 
accepts the registration and the calls, and when it needs to forward 
something to Asterisk, you just add a forwarding/rewrite block to point 
to the extension(s) on Asterisk you need the SIP messages to go.

Some caveats (which may be different for OpenSER, so someone else can 
chime in):

SER doesn't do voicemail.  You'd need to either use a separate app 
called SEMS (which meshes with SER) or forward to Asterisk to handle 
voicemail (which works easily and well). Call Forwarding can be a bit 
tricky, but easy to handle in SER.  SER's 'billing' capability is... not 
ideal. It has no CDR engine of any kind, as it is just a SIP proxy. It 
has no concept of a whole call from start to finish... only the messages 
passed in a call (INVITEs, ACKs, BYEs, etc). While it will log all this 
information and allow you to piece together a call's start time and 
finish time, it is not what one might call elegant.

N.


Antoine Megalla wrote:
> Hi,
>
> I have a client who requires an Asterisk system with 1500 SIP 
> clients.All clients will have ATAs (mostly Grandstream), so I think a 
> single Asterisk server will not be able to handle all 1500 
> registrations, plus
> typical applications like Voicemail, call forwarding, etc.. and the 
> billing needs for all the clients.
>
> I have searched all over, and it seems that the perfect solution is 
> using SER/OpenSER as registration server for the SIP clients, and then 
> use Asterisk (one or more servers in load balancing mode) for 
> everything else.
>
> The problem is that I cannot find any configuration files for such a 
> setup. I can do all the Asterisk configuration, dial plan, AGIs, apps, 
> etc.. but for SER/OpenSER I cannot find anything.
>
> Can anyone please point me in the right direction, provide me with 
> OpenSER configuration, or any pointers on the subject. I tried to read 
> all the material on how to write configuration files for OpenSER, but 
> it is incomprehensible to me, and it is much harder that when I 
> learning Asterisk 3 years ago.
>
> Your help is much appreciated.
>
> Regards,
>
> Antoine Megalla.
>
> __
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>
> 
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Re: [asterisk-users] Druid

[Juan Sandro]

Hi Dean,

I am using Bicom Systems PBXware . In my opinion there is no asterisk GI that 
can compare to it.

Juan



Date: Wed, 31 Oct 2007 19:35:48 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Druid


Is anyone out there using Druid?

After the switchbox announcement today I’ve been looking into some other gui’s 
and as I’ll probably do a trial install this weekend of the free switchvox iso 
but I thought I’d ask is there any other guis I should be burning trial ISO’s 
of as well?

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).



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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[satish patel]

You can use freeswitch for this kind of setup its working on asterisk technology

asterisk + SER intergration URL 

http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER



Antoine Megalla <[EMAIL PROTECTED]> wrote:  Hi,

I have a client who requires an  Asterisk system with 1500 SIP clients.All 
clients will have ATAs (mostly  Grandstream), so I think a single Asterisk 
server will not be able to handle all  1500 registrations, plus
typical applications like Voicemail, call  forwarding, etc.. and the billing 
needs for all the clients.

I have  searched all over, and it seems that the perfect solution is using 
SER/OpenSER  as registration server for the SIP clients, and then use Asterisk 
(one or more  servers in load balancing mode) for everything else.

The problem is that  I cannot find any configuration files for such a setup. I 
can do all the  Asterisk configuration, dial plan, AGIs, apps, etc.. but for 
SER/OpenSER I  cannot find anything.

Can anyone please point me in the right direction,  provide me with OpenSER 
configuration, or any pointers on the subject. I tried  to read all the 
material on how to write configuration files for OpenSER, but it  is 
incomprehensible to me, and it is much harder that when I learning Asterisk 3  
years ago.

Your help is much appreciated.

Regards,

Antoine  Megalla.
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PGP Signature--

Satish Patel
mobile:- +91-9818875535

http://www.linuxbug.org
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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[satish patel]

This is enough one to setup ser with asterisk 

http://www.voip-info.org/wiki/view/Asterisk+at+large

Antoine Megalla <[EMAIL PROTECTED]> wrote: Hi,

I have a client who requires an Asterisk system with
1500 SIP clients.
All clients will have ATAs (mostly Grandstream), so I
think a single 
Asterisk server will not be able to handle all 1500
registrations, plus 
typical applications like Voicemail, call forwarding,
etc.. and the billing 
needs for all the clients.

I have searched all over, and it seems that the
perfect solution is using 
SER/OpenSER as registration server for the SIP
clients, and then use 
Asterisk (one or more servers in load balancing mode)
for everything else.

The problem is that I cannot find any configuration
files for such a setup.
I can do all the Asterisk configuration, dial plan,
AGIs, apps, etc.. but 
for SER/OpenSER I cannot find anything.

Can anyone please point me in the right direction,
provide me with OpenSER 
configuration, or any pointers on the subject. I tried
to read all the 
material on how to write configuration files for
OpenSER, but it is 
incomprehensible to me, and it is much harder that
when I learning Asterisk 
3 years ago.

Your help is much appreciated.

Regards,

Antoine Megalla. 



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Satish Patel
mobile:- +91-9818875535

http://www.linuxbug.org
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Re: [asterisk-users] Spam Filter News

[Jared Smith]

On Thu, 2007-11-01 at 08:05 +, Thomas Kenyon wrote:
> Is there any news on getting the Spam Filter fixed for this mailing list?

We've tweaked it quite a bit in the past few weeks, and it seems to be
much better.  If you're still experiencing issues, please send me a
private email and I'll look into it in more detail.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Connection astrisk to a RAS (portmaster)

[Jared Smith]

On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote:
> I also get sometime :
> 
>   == Primary D-Channel on span 2 down
> [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No 
> D-channels available!  Using Primary channel 48 as D-channel anyway!
>   == Primary D-Channel on span 2 up

I don't claim to be an expert on the Wanpipe drivers, but it's been my
experience that if your D-channels bounce up and down like that every
few seconds, that the Wanpipe drivers didn't successfully patch Zaptel,
or you haven't restarted Zaptel since the Wanpipe drivers patched
Zaptel.

Hopefully that's enough to get you up and running.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Mobile phone codecs ...

[Steve Kennedy]

On Thu, Nov 01, 2007 at 01:09:24PM +0100, Benny Amorsen wrote:

> > "AM" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes:
> AM> Maybe the GSM codec is implanted to the "GSM chip" and that one
> AM> does alaw, ulaw...
> Also, modern handsets like the E90 rarely use the plain GSM codec.
> They use newer codecs such as EFR whenever possible. Asterisk probably
> won't support EFR anytime soon; it is patent encumbered and not
> particularly suited for LAN or WLAN use.

GSM networks are circuit switched and use 13Kb/s slots (i.e. GSM codec
is 13Kb/s codec).

3G should just be data, so there's a wider choice of codecs.


Steve

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UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
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[asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[Antoine Megalla]

Hi,

I have a client who requires an Asterisk system with 1500 SIP clients.All 
clients will have ATAs (mostly Grandstream), so I think a single Asterisk 
server will not be able to handle all 1500 registrations, plus
typical applications like Voicemail, call forwarding, etc.. and the billing 
needs for all the clients.

I have searched all over, and it seems that the perfect solution is using 
SER/OpenSER as registration server for the SIP clients, and then use Asterisk 
(one or more servers in load balancing mode) for everything else.

The problem is that I cannot find any configuration files for such a setup. I 
can do all the Asterisk configuration, dial plan, AGIs, apps, etc.. but for 
SER/OpenSER I cannot find anything.

Can anyone please point me in the right direction, provide me with OpenSER 
configuration, or any pointers on the subject. I tried to read all the material 
on how to write configuration files for OpenSER, but it is incomprehensible to 
me, and it is much harder that when I learning Asterisk 3 years ago.

Your help is much appreciated.

Regards,

Antoine Megalla.

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[asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

[Antoine Megalla]

Hi,

I have a client who requires an Asterisk system with
1500 SIP clients.
All clients will have ATAs (mostly Grandstream), so I
think a single 
Asterisk server will not be able to handle all 1500
registrations, plus 
typical applications like Voicemail, call forwarding,
etc.. and the billing 
needs for all the clients.

I have searched all over, and it seems that the
perfect solution is using 
SER/OpenSER as registration server for the SIP
clients, and then use 
Asterisk (one or more servers in load balancing mode)
for everything else.

The problem is that I cannot find any configuration
files for such a setup.
I can do all the Asterisk configuration, dial plan,
AGIs, apps, etc.. but 
for SER/OpenSER I cannot find anything.

Can anyone please point me in the right direction,
provide me with OpenSER 
configuration, or any pointers on the subject. I tried
to read all the 
material on how to write configuration files for
OpenSER, but it is 
incomprehensible to me, and it is much harder that
when I learning Asterisk 
3 years ago.

Your help is much appreciated.

Regards,

Antoine Megalla. 



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Re: [asterisk-users] Mobile phone codecs ...

[Benny Amorsen]

> "AM" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes:

AM> Maybe the GSM codec is implanted to the "GSM chip" and that one
AM> does alaw, ulaw...

Also, modern handsets like the E90 rarely use the plain GSM codec.
They use newer codecs such as EFR whenever possible. Asterisk probably
won't support EFR anytime soon; it is patent encumbered and not
particularly suited for LAN or WLAN use.


/Benny



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Re: [asterisk-users] SER with Asterisk intergration

[J. Oquendo]

satish patel wrote:


> machine ?? i have asterisk with 200 SIP device but i voice
> qulity and load of asterisk is bit high so i need to implement

Why do you think you need SER? You must have something horrible on your
set up

# asterisk -rx "sip show peers"|tail -n 2
325 sip peers [299 online , 26 offline]
Verbosity is at least 10

Not one user has complained



J. Oquendo

SGFA (FW+VPN v4.1)
SGFE (FW+VPN v4.1)

"I hear much of people's calling out to punish the
guilty, but very few are concerned to clear the
innocent." Daniel Defoe

http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xF684C42E



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Re: [asterisk-users] SER with Asterisk intergration

[satish patel]

have you configuration file of SER to send request to asterisk i m not much 
familer with SER i have knowledge of asterik

Arun Kumar <[EMAIL PROTECTED]> wrote: just configure SER on another port and 
use.

On 11/1/07, satish patel <[EMAIL PROTECTED] > wrote:Dear all

  anybody have implement SER with Asterisk in single machine ?? i have 
asterisk with 200 SIP device but i voice qulity and load of asterisk is bit 
high so i need to implement SER for SIP registra and asterisk for feature 

Rgerads




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Re: [asterisk-users] Druid

[Dean Collins]

Nice - I like where it's heading, finally someone came up with a half
decent looking UI for a User Portal (though not perfect)

 

I've consulted to 3 different companies about User Portals and not a
single one has implemented anything near what it should look like.

 

I just don't get why Asterisk product designers don't talk hire proper
UI consultants to build the right solution.

 

That's what you get for being cheap, all the pieces available and still
no clue.

 

 

 

Regards,

 

Dean Collins

Cognation Pty Ltd

[EMAIL PROTECTED] 

+1-212-203-4357 Ph

+61-2-9016-5642 (Sydney in-dial).

 

 

 

> -Original Message-

> From: [EMAIL PROTECTED] [mailto:asterisk-users-

> [EMAIL PROTECTED] On Behalf Of Alan Lord

> Sent: Thursday, 1 November 2007 5:07 AM

> To: asterisk-users@lists.digium.com

> Subject: Re: [asterisk-users] Druid

> 

> Alex Epshteyn wrote:

> 

> I came across this one the other day:

> 

> http://www.voiceone.it/index.php?synSiteLang=2

> 

> It's GPL (Important to me!) and it looks very slick. It also has a

> "user" interface as well as an "admin" interface. Although quite young

> from what I have seen of it - it looks like the dogs b's.

> 

> Al

> 

> --

> The way out is open!

> http://www.theopensourcerer.com

> 

> 

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[asterisk-users] Video Call

[voip Server asterisk]

Hi..

Iam new with asterisk PBX, and i have read about asterisk video call.: my
question:

1. Is imposible to establish system video call (from Phone with
GPRS/3G enabled
to Computer Running Softphone like X-Lite) over Asterisk Gateway..
2. If posible what requirement (Hardware and Software on my Asterisk,PC or
My Phone)


Thanks

Joko Pitoyo
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Re: [asterisk-users] Call Failed

[Doug Lytle]

Robert La Ferla wrote:
> After so many rings when the party does not answer, my SIP phone says  
> Call Failed.  Why doesn't it just keep ringing?
>
> Here's the dial plan rule:
>
> exten => _NX,1,Dial(SIP/[EMAIL PROTECTED],,r)
> exten => _NX,n,Hangup()
>   


Not that it's the cause, but get rid of the r.

Doug

-- 
Ben Franklin quote:

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



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[asterisk-users] Autodialing

[Bhrugu Mehta]

hi,all
I want make Autodialer in c++ using Asterisk Mangager Interfase;
how to syncronize originate action i.e. at a time one call made and
this time asterisk wait for some second to generate new call.

thnks in advance.
Bhrugu Mehta (SAI INFO SYSTEM)

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Re: [asterisk-users] Druid

[Alan Lord]

Alex Epshteyn wrote:
> Dean,
> 
>  
> 
> If you are looking for a non-restricting and extensible Asterisk GUI 
> please look at Thirdlane http://www.thirdlane.com 
> . If you are comfortable installing OS, 
> Webmin and Asterisk, I would suggest installing PBX Manager GUI 
> (packaged as a Webmin module), otherwise Thirdlane Advantage (CentOS 
> based ISO) may be a good option.  

I came across this one the other day:

http://www.voiceone.it/index.php?synSiteLang=2

It's GPL (Important to me!) and it looks very slick. It also has a 
"user" interface as well as an "admin" interface. Although quite young 
from what I have seen of it - it looks like the dogs b's.

Al

-- 
The way out is open!
http://www.theopensourcerer.com


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Re: [asterisk-users] SER with Asterisk intergration

[Arun Kumar]

just configure SER on another port and use.

On 11/1/07, satish patel <[EMAIL PROTECTED]> wrote:
>
> Dear all
>
>   anybody have implement SER with Asterisk in single machine ?? i
> have asterisk with 200 SIP device but i voice qulity and load of asterisk is
> bit high so i need to implement SER for SIP registra and asterisk for
> feature
>
> Rgerads
>
>
>
>
> PGP Signature--
>
> Satish Patel
> mobile:- +91-9818875535
>
> http://www.linuxbug.org
>
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> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
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Re: [asterisk-users] SparkLan WVPR-100 Wireless Handset.

[Thomas Kenyon]

Thomas Kenyon wrote:
> Is there any news on getting the Spam Filter fixed for this mailing list?
> 
It just figures that this should get through.
Anyway, back to my original Post.

Has anyone here used a SparkLan WVPR-100 Wireless Handset?

 http://www.sparklan.com/product_details.php?prod_id=4

They look quite good value for money. (I nearly got put off by the lack 
of WPA on the product detail page, but it states support for it in the 
Datasheet, not sure how reliable this is, in the page above the standby 
time is quoted as 70 hours whereas on the datasheet it is quoted as 50).

Weirdly also supports G.729a/b (which I would have thought would drain 
the battery too much and not be worth the extra licensing outlay when 
manufacturing the product).

No mention of Call Transfers etc. Still looks interesting.

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[asterisk-users] Spam Filter News

[Thomas Kenyon]

Is there any news on getting the Spam Filter fixed for this mailing list?

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[asterisk-users] SER with Asterisk intergration

[satish patel]

Dear all

  anybody have implement SER with Asterisk in single machine ?? i have 
asterisk with 200 SIP device but i voice qulity and load of asterisk is bit 
high so i need to implement SER for SIP registra and asterisk for feature

Rgerads




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http://www.linuxbug.org
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