Re: [asterisk-users] log incoming calls without answering

2017-04-20 Thread Brandon B. 

On 2017-04-20 02:33 PM, Fabio Moretti wrote:

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


Have you considered the legal implications of what you are trying to 
accomplish? You are describing a pen register 
. Asterisk is not suitable 
for this purpose, even if it technically can be used in a fashion. You 
might want look into high impedance telephone line equipment which solve 
the technical issues.


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[asterisk-users] Spout Communications

2016-11-29 Thread Brandon B. 
Does anyone on this list have any contact information for anyone 
associated with Steven Steffler, the late owner 
 
of the VoIP company Spout Communications ? I'd 
like to help get my services restored and help this company if possible.


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Re: [asterisk-users] Asterisk compatibility with SMS services

2016-11-29 Thread Brandon B. 


Can anyone comment on using SMS in conjunction with VoIP service using 
one of these three VoIP providers: voip.ms, vitelity.com, 
flowroute.com? Are some SMS services more compatible with Asterisk 
(i.e. SMS over SIP works perfectly or not)? Is it best to use a 
different data channel for SMS messages (i.e. SMS via HTTP, SMS via 
XMPP) instead of Asterisk's built in SMS application MessageSend 
? In order to 
develop a web application for sending and receives SMS messages for 
business users, are there any pitfalls in using Asterisk to handle the 
message exchanges?




On 2016-11-29 09:01 AM, Sebastian Nielsen wrote:


Im using SMS successfully over VoIP. No problems at all. You however 
need to use a good codec.


However, I don’t use the MessageSend application, instead I use the 
raw SMS() application.


This works by the SMS centre calling my fixed landline from a specific 
number, I detect the callerid, initiate a SMS reception and then the 
SMS is in the spool files.


If I want to send a outgoing SMS, I push a SMS file in the spool 
folder, then initate a call to the SMS centre.




That's pretty cool, thank you for the details. You are using the builtin 
SMS application that exchanges SMS data over SIP / PSTN connections. I 
don't believe I can get service like that in Canada. Does anyone use the 
SMS applications to send and receive SMS messages in North America? Who 
provides that service?


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[asterisk-users] Asterisk compatibility with SMS services

2016-11-29 Thread Brandon B. 
Can anyone comment on using SMS in conjunction with VoIP service using 
one of these three VoIP providers: voip.ms, vitelity.com, flowroute.com? 
Are some SMS services more compatible with Asterisk (i.e. SMS over SIP 
works perfectly or not)? Is it best to use a different data channel for 
SMS messages (i.e. SMS via HTTP, SMS via XMPP) instead of Asterisk's 
built in SMS application MessageSend 
? In order to 
develop a web application for sending and receives SMS messages for 
business users, are there any pitfalls in using Asterisk to handle the 
message exchanges?


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[asterisk-users] Asterisk 14.0.2 opens a high numbered UDP port

2016-10-13 Thread Brandon B. 
What part of Asterisk 14.0.2 opens the random, high numbered (33094 
currently) UDP port? This port is opened even without any channel 
drivers loaded.


$ sudo netstat -ltunp | grep asterisk
udp0  0 0.0.0.0:51488 
0.0.0.0:*   13830/asterisk
udp0  0 0.0.0.0:5060 0.0.0.0:*   
13830/asterisk
udp0  0 :::42516 :::*
13830/asterisk




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Re: [asterisk-users] combine external video source and audio call to make SIP video call?

2013-11-25 Thread Brandon B. 
Use command line video capable SIP software (linphone / linphonec) with
source and modify it so an incoming call forwards to an outgoing
destination with video added. You might want to talk to developers of
linphone and ask them to make the change for you. Your analog phone calls
to linphonec (probably using Asterisk) so that a call is routed to the SIP
softphone. The softphone adds the video and send the combined audio/video
SIP call back to Asterisk to be routed to the destination.


On Sun, Nov 24, 2013 at 1:44 PM, Eric Cooper e...@cmu.edu wrote:

 I'd like to cobble together a videophone from an analog phone,
 connected to an Asterisk FXS channel, and a co-located video camera,
 connected to a video grabber card on the Asterisk server (so I have a
 Linux video device providing the video stream).  When a call is made
 from the phone, I'd like to somehow add the video and produce a SIP
 video call.  I don't want to use any sort of graphical SIP client;
 ideally this should all be done headlessly in the Asterisk dialplan.
 Any suggestions on where to start?

 --
 Eric Cooper e c c @ c m u . e d u

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[asterisk-users] HD Voice -- connecting Asterisk into HD Voice compatible mobile phone

2013-05-11 Thread Brandon B. 
I've just noticed that mobile phones are starting to support HD Voice.
http://thenextweb.com/apple/2012/09/21/iphone-5s-hd-voice-impressive-theres-still-work-done/Is
there any hardware capable of connecting HD voice quality calls into
Asterisk now? I see that Asterisk compatible GSM hardware is available, but
does anything support HD voice presently or should in the future?
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Re: [asterisk-users] Impromptu conferencing

2012-12-03 Thread Brandon B. 
On Mon, Dec 3, 2012 at 1:09 AM, martin f krafft madd...@madduck.net wrote:

 also sprach Brandon B. bran...@brellsystems.com [2012.12.03.0132 +0100]:
  [all-inbound-for-999]
  ; inbound extension through a conference room
  exten = 999,1,MeetMeCount(999,COUNT-999);
  exten = 999,2,GotoIf($[${COUNT-999}=1]?10);
  exten = 999,3,Dial(SIP/99,999,G(6));
  exten = 999,4,Hangup;
  exten = 999,6,MeetMe(999,FAqx);
  exten = 999,7,MeetMe(999,Fqx);
 
  ; bypass the conference room for multiple inbound calls
  exten = 999,10,Dial(SIP/999);

 This is an interesting approach, but I am still not sure how to add
 the third party. Sure, I can call them up and tell them to dial
 a number, but I'd really rather be able to just switch them in.


The third party has to dial into the system at some point. You can only
switch in existing calls, so your idea is nonsense to me because you
haven't specified how else the third party is in or gets a channel into the
system.

The simplest solution would be to have the third party dial into or be
connected into conference room 999. The first two parties are already in
the conference. The conference room 999 could be available by dialling 8999:

[call-to-conference-room-999]
exten = 8999,1,MeetMeCount(999,COUNT-999);
exten = 8999,2,GotoIf($[${COUNT-999}=0]?10);
exten = 8999,3,MeetMe(999,Fqx);

exten = 8999,10,Hangup;

Anybody who can dial 8999 in the above context will join the conference.

What would need to be done for a user to e.g. suspend the
 conference, dial another number and finally merge the channels?


Suspending the conference would mean that phone 999 puts the call on hold,
and then does one of the following: (a) dials a third party and then
transfers the connected third party call into the conference room, (b) uses
the conference feature of their phone to connect a third party to their
channel already in the conference room or (c) invites a third party to dial
a number which connects them to conference room 999.

Do I need the manager API for that, like this:


 http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#Mergingconferences


Sure, that will work in theory.

Look into the Bridge command which is available inside a dialplan using
extensions.conf. You can take a call which will end up in a conference room
and before that call joins the conference room it will find* other calls in
the system meant to be in the conference room, bridge those other calls
into a conference room, and finally join the conference room. In this case
the third party call initiates the conference. Alternatively, in the
example above, the conference room always is used so the third party just
needs to join it.

Good Luck.

* finding calls is not possible unless you make them findable
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Re: [asterisk-users] Impromptu conferencing

2012-12-02 Thread Brandon B. 
The feature can be enabled on outbound and inbound calls by connecting both
channels of a connected call to a conference room after the connection is
made. This accomplishes your need to not do anything vs. giving the called
person instructions, transferring the call to a conference room and then
dialing into it immediately. For example, if the conference room is 999 for
phone 999, and phone 999 uses the context [all-outbound-for-999] for
outbound calls and [all-inbound-for-999], then:

[all-outbound-for-999]
; outbound dialing through a conference room
exten = _NXXNXX,1,MeetMeCount(999,COUNT-999);
exten = _NXXNXX,2,GotoIf($[${COUNT-999}=1]?10);
exten = _NXXNXX,3,Set(CALLERID(num)=555999);
exten = _NXXNXX,4,Set(CALLERID(name)=Phone 999);
exten = _NXXNXX,5,Dial(${TRUNK}/${EXTEN},99,G(6));
exten = _NXXNXX,6,MeetMe(999,FAqx);
exten = _NXXNXX,7,MeetMe(999,Fqx);

; bypass the conference room for multiple outbound calls
exten = _NXXNXX,10,Set(CALLERID(num)=555999);
exten = _NXXNXX,11,Set(CALLERID(name)=Phone 999);
exten = _NXXNXX,12,Dial(${TRUNK}/${EXTEN});

[all-inbound-for-999]
; inbound extension through a conference room
exten = 999,1,MeetMeCount(999,COUNT-999);
exten = 999,2,GotoIf($[${COUNT-999}=1]?10);
exten = 999,3,Dial(SIP/99,999,G(6));
exten = 999,4,Hangup;
exten = 999,6,MeetMe(999,FAqx);
exten = 999,7,MeetMe(999,Fqx);

; bypass the conference room for multiple inbound calls
exten = 999,10,Dial(SIP/999);



On Thu, Nov 8, 2012 at 2:01 AM, martin f krafft madd...@madduck.net wrote:

 also sprach Administrator TOOTAI ad...@tootai.net [2012.11.08.0954
 +0100]:
  Does anyone have a working example they would be willing to
  share?
 
  As said by James, you just have to transfer all parties in
  a conference room and then you call this conference.

 The scenario is usually that we are in a discussion and need a third
 party. I suppose I can tell the initial correspondent I will now
 transfer you to a conference room, enter this PIN when asked, then
 hang up, dial the next, and do the same.

 What I would like to do is to convert the current channel into
 a conference room, go on hold and dial a third party, and when
 I come back to the conference room, I bring along the third party.

 Put differently: I don't really want my correspondents to have to do
 anything, just wait and listen to MOH.

 --
 martin | http://madduck.net/ | http://two.sentenc.es/

 nullum magnum ingenium sine mixtura dementiae fuit.
  -- seneca

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Re: [asterisk-users] Call Recording

2012-08-28 Thread Brandon B. 
You should simplify until you have something that works, then add your 
conditions back in one line at a time.


On 12-08-28 11:05 AM, Josh Hopkins wrote:


-- Executing [s@macro-one-touch-record:3] 
ExecIf(SIP/1010-0161, 1?MacroExit()) in new stack




This is where the inbound call is exiting.

exten = s,n,ExecIf($[${CUT(CALLFILENAME,-,1)}=exten  
${DB(AMPUSER/${THISEXTEN}/recording/ondemand)}!=enabled]?MacroExit())


This condition is causing MacroExit to be called, so fix these conditions.
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[asterisk-users] asterisk crash

2012-07-26 Thread Brandon B. 
Hi, one of our asterisk servers recently crashed. Any direction as to how I
can provide helpful information about this issue would be appreciated:

asterisk*CLI core show version
Asterisk 1.8.13.0 built by root @ asterisk on a i686 running Linux on
2012-06-10 22:22:13 UTC

Jul 26 16:39:39 asterisk kernel: [525204.496674] asterisk[5766]: segfault
at 0 ip b7521c86 sp b57a8f48 error 4 in libc-2.11.3.so[b74ad000+14]

The asterisk server is running on an up to date Debian 6 system, but
without the -g option as per
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace, so I
beleive I cannot provide any more information.

Is this a libc6 problem?
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Re: [asterisk-users] Setting up Outgoing Trunk

2009-08-10 Thread Brandon B. 
Assuming you are configuring your Asterisk using the configuration
files -- if you want the caller ID on phone calls between users to be
the same as the caller id on calls made with the trunk lines, set the
external caller id information for the users in sip.conf (i.e.
callerid=9995551212) or to simply change the caller id information
for calls made on the trunk lines, modify the extensions.conf file by
inserting something like the following line before the Dial command.
An example:

[outgoing-calls-context]
exten = _NXXNXX,1,Set(CALLERID(num)=${EXTCALLERID})
exten = _NXXNXX,2,Dial(${TRUNK}/${EXTEN})

The problem here is that if you are using text files for configuration
you are should know this, so this advice is either almost redundant or
not helpful. What kind of Asterisk GUI tool are you using?


On Mon, Aug 10, 2009 at 9:47 AM,
kumarshantanushantanu1...@rediffmail.com wrote:


 On Thu, 06 Aug 2009 21:28:01 +0530 wrote
On 6 Aug 2009, at 16:32, kumarshantanu wrote:
 Hello Everybody,

 Hi.

 I have a genuine problem in Asterisk setup.

 Ok.

 I have three inbound trunks in my asterisk box, everything is

 What kind of trunks.

 These are sip trunks

 working fine but the only problem is when any user make an out-
 going call through his/her extension it goes with same number labeled
 on this.

 Ok.

 Can we set each of these lines to have fixed outgoing numbers
 like if extn: 201 make an outgoing call the recipient should get
 different no and if extn: 202 make an outgoing call the recipient
 should
 get different one.

 Ok.

 Please can someone help me in this.

 If you show us some config, tell us trunk types and generally 'giving
 us something to go on.

 What config you want from me. I am not very much friendly to asterisk,
 for now I manage it from freePBX. Let me please know if I can provide you
 some more information

 Thanks
 Shantanu

 Steve


 Heh

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Re: [asterisk-users] Preventing MOH from restarting the tune when a call is parked

2009-06-13 Thread Brandon B. 
There might be a simpler solution, but you could use multiple MOH
files and play them randomly, or run use the icecast musiconhold.conf
configuration to play a network mp3 stream.

http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf#Exampleusingicecastampshoutcaststreams
http://www.icecast.org/


On Sat, Jun 13, 2009 at 5:24 AM, martin f krafft madd...@madduck.net wrote:
 Hi folks,

 When I try to park a call, my SIP phone puts the other party on hold
 and MOH starts to play a tune. I then dial 700 and wait for the
 parking slot announcement.

 As soon as the other party gets put into the parkinglot, the MOH
 tune starts again from the beginning. Is there a way to prevent that
 and just let it play?

 --
 martin | http://madduck.net/ | http://two.sentenc.es/

 http://lavender.cime.net/~ricky/badgers.txt

 spamtraps: madduck.bo...@madduck.net

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Re: [asterisk-users] extensions not being detected consistently

2009-06-02 Thread Brandon B. 
Extensions that are dialed within macros like the following lines
could cause the type of problems as you mentioned:

exten = s,n,Macro(dial-us)
exten = s,n,Macro(hangupcall)

This line:

  exten = s,n,Wait(0.5);

should be changed to exten = s,n,WaitExten(0.5); and  these lines:

exten = Wait(10)
exten = s,n(open),NoOp(open)

are not valid. Try this:

exten = s,1,Set(TIMEOUT(digit)=10)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(cassandra/CustomWelcomeMessage)
exten = s,n,GotoIfTime(09:00-17:30,mon-fri,*,*?open)
exten = s,n,Background(cassandra/OfficeHours)
exten = s,n,Background(cassandra/NextRep)

exten = t,n,Macro(dial-us)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(s,open)
exten = #,1,Macro(hangupcall)






On Mon, Jun 1, 2009 at 1:47 PM, Terry Nathan tnat...@aiinc.ca wrote:
 G'afternoon everybody,

 I'm having a problem with consistently being able to ring our extensions
 from an outside line. I don't have a problem reaching the number, but
 during our calls to Background(msg) that I am having a problem. It seems
 to be an issue with timing. If I press the extension towards the end of
 the Background(msg) the it often works. However, in the middle of the
 message it will not work at all.

 What is also strange is that I can dial an extension any time if I call
 from one of our ip phones. This seems to be strictly a problem with
 regular phones, then the timing of dialing the extension becomes important.

 The fact that the ip phones always work seems to suggest that I need to
 look at tone detection, but after googling and searching the bowels of
 every conf file I could find, I haven't found any magic bullet.
 I should mention that the first call to Background() usually works, even
 for the regular phones, I think this is because it is short enough that
 the timing of dialing the extension is relatively easy.

 I don't know if it is significant or not but it seems that once a callee
 tries to dial an extension and it doesn't work, even the next few calls
 will also not work. And similarly, sometimes it works and then a few
 calls will go through, but then it will go back to not detecting
 properly again. Asterisk is running on its own box and there is nothing
 unusual happening with the system, or even people on other lines, that
 is happening.

 Checking the log files when I call in Asterisk tells me that either it
 only detects 1 of the 3 digits (usually the second or third one) or, if
 I dial the extension at a different point in the message, that the first
 digit was pressed twice e.g. '22' instead of just '2'. The inconsistency
 of the problem is starting to drive me bonkers as I can't accurately
 nail down the problem.

 Ideally I'd like our callees to be able to dial an extension as soon as
 the call to Background() is hit in the context, from any phone that
 calls in, not just ip phones. My setup is an installation of
 asterisk-now (Centos 5 with Asterisk 1.4.24)

 If anyone has seen a problem like this before or has even an inkling of
 what it might be, that would be awesome :D Thanks in advance.

 My dial plan:

 [incoming-our-number]

 exten = s,1,Answer
 exten = s,n,NoOp(incoming-our-number)
 exten = s,n,Background(cassandra/CustomWelcomeMessage)       This
 line is usual fine, I think because the message is short enough that
 timing the dialing of the extension is less of an issue.
 exten = s,n,GotoIfTime(09:00-17:30,mon-fri,*,*?open)
 exten = s,n,Wait(0.5);
 exten = s,n,Background(cassandra/OfficeHours)
 exten = Wait(10)
 exten = s,n(open),NoOp(open)
 exten = s,n,WaitExten(0.5);
 exten = s,n,Set(TIMEOUT(digit)=10)
 exten = s,n,Set(TIMEOUT(response)=15)
 exten = s,n,Background(cassandra/NextRep)
      This is the line where I have a problem with dialing an
 extension. The timing is very fickle and heaps of our callees cannot get
 to the right extension properly.
 exten = s,n,WaitExten(10,m[default])

 exten = s,n,Macro(dial-us)
 exten = s,n,Macro(hangupcall)

 exten = i,1,Playback(pbx-invalid)
 exten = i,2,Goto(s,open)
 exten = t,1,Macro(hangupcall)
 exten = #,1,Macro(hangupcall)


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Re: [asterisk-users] Asterisk maximum user

2009-06-02 Thread Brandon B. 
I think you are looking at this incorrectly, First, if you are
seriously looking at thousands of calls, you won't want to be doing
this without the help of somebody who is experienced with Asterisk and
knows how to solve scaling issues. Secondly, a TDM call between two
PRI channels is going to involve very little overhead, and you
probably could theoretically handle many hundreds of calls (perhaps
even a thousand) like this in a single Asterisk system. But, since
that is very expensive to do because PRI ports are not cheap, that is
probably not what you are doing with your low end single CPU Asterisk
system. Software echo cancellation and transcoding between audio
codecs will cost a lot of CPU time, while call recorded will use a
high level of hard disk I/O, so thse issues must be considered.
Assuming no CPU transcoding, no CPU echo cancelling and no call
recording, your worry should be about likely capacity of a single
server given your specifics and then scaling issues (i.e. I have a
1/2/4/8/16 CPU Asterisk server handling many calls, but the CPU load
is spiking above 2/2/4/8/16. How do I scale to 2+ servers?).

Some capacity hints: assuming SIP telephones over a high bandwidth
network (g.711 codec by all SIP devices) and multiple SIP or PRI
trunks for connecting calls to the PSTN, you might solve this problem
by using a modern quad CPU server with redundant power supplies and
redundant hard disks and configure 100 SIP phones per server with a
Quad port PRI card (TE420B) or SIP trunking for PSTN connections. This
might work for a business if the system will hit 50% capacity often
with capacity spikes of 75% possible. If the number of calls will
normally reach 10% of capacity, you might be able to configured 5
times more SIP phones/users per server.

Brandon.

On Tue, Jun 2, 2009 at 7:31 AM, M.Monzur Alam mon...@citechco.net wrote:
 How many asterisk voice user concurrently continue their voice only one
 Asterisk sever? Could it possible implementation in big environment
 (more than thousand users)?
 Hardware statistics:

 vendor_id               : GenuineIntel
 cpu family              : 15
 model                   : 2
 model name              : Intel(R) Pentium(R) 4 CPU 2.80GHz
 stepping                : 9
 cpu MHz         : 2799.622
 cache size      : 512 KB
 fdiv_bug                : no
 hlt_bug                 : no
 f00f_bug        : no
 coma_bug        : no
 fpu                     : yes
 fpu_exception           : yes
 cpuid level             : 2
 wp                      : yes
 flags                   : fpu vme de pse tsc msr pae mce cx8 apic sep
 mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
 pbe up pebs bts cid xtpr
 bogomips        : 5604.90
 clflush size            : 64


 Thanks
 Monzur


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Re: [asterisk-users] Not receiving voicemail message in mailbox

2009-05-10 Thread Brandon B. 
mutt will not deliver a email message, so you are using the wrong
command. The email message with attachment is created by Asterisk and
needs msmtp to deliver the message.

On Sun, May 10, 2009 at 9:10 AM, jonas kellens jonas.kell...@telenet.be wrote:
 Dave,

 can you help me with my configuration of mutt (MUA) + msmtp (MTA) ?

 I have included the following in my voicemail.conf :

 mailcmd=/usr/sbin/mutt

 But how will Asterisk know how to use Mutt to attach its voicemail-message
 (.wav-file) ???

 I use Mutt together with msmtp to send me weekly the Asterisk-log files. I
 put the commands in my crontab-file.

 Attachment is possible with mutt -a attachment

 How does Asterisk know to use the -a option ???

 Thanks for he help !

 Jonas.



 On Fri, 2009-05-08 at 13:35 -0700, Dave Walker wrote:

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

 mailcmd Mailcmd allows the administrator to override the default mailer
 command with a defined command. Mailcmd takes a string value set to the
 desired command line to execute when a user needs to be notified of a voice
 mail message. The default command line is: '/usr/sbin/sendmail -t'. A useful
 alternative to sendmail is Exim which many people find easier to configure.
 Examples

  ; - you will need to escape any  or  characters with a \
  mailcmd=/usr/sbin/sendmail -v -t -f asterisk-...@yourdomain.com   ; use -f to prevent r...@localhost.localdomain or similar
  mailcmd=/usr/exim/bin/exim -t
  ; - use the next line for testing
  mailcmd=cat \ /tmp/astvm-mail



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Re: [asterisk-users] inbound filed

2009-04-15 Thread Brandon B. 
You call call to extension '246463' will not match 'exten =
1246463'.

On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez
bayardo.sanc...@gmail.comwrote:

 i create inbound confi my confi is:

 [incoming]
 exten= 1246463,,1,Dial(SIP/8003,60,rT)
 exten= 6463,1,Dial(SIP/8003,60,rT)
 exten= 1246463,,n,Wait(5)
 exten= 1246463,,n,Hangup

 but y calling and send this error in my CLI:

 [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
 Call from '101396_procall' to extension '246463' rejected because
 extension not found.

 --
 Bayardo Sánchez García
 Web Developer - Internet Portals - Asterisk Support - Windows Server
 Support - Proxi Support - Linux
 E-mail: bayardo.sanc...@gmail.com
 Linux User: #418392
 America Central - Managua, NI (505) 249-2853 -  4886876
 IM msn messenger: bjsanch...@hotmail.com
 Skype: bayardo.sanchez
 This email is intended solely for the person or organization to which it is
 addressed. It may contain privileged and confidential information. If you
 are not the intended recipient, you are prohibited from copying, disclosing
 or distributing this email or its contents (as it may be unlawful for you to
 do so) or taking any action in reliance on it. If you have received this
 email by mistake, please delete it. All e-mail sent to this address will be
 received by B.S. Solution e-mail system and is subject to archiving and
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Re: [asterisk-users] inbound filed

2009-04-15 Thread Brandon B. 
Try this:

[incoming]
exten= 246463,1,Dial(SIP/8003,60,rT)
exten= 246463,n,Wait(5)
exten= 246463,n,Hangup
exten= 6463,1,Dial(SIP/8003,60,rT)



On Wed, Apr 15, 2009 at 11:00 AM, Bayardo Sanchez bayardo.sanc...@gmail.com
 wrote:

 i call my tollfree number and send the call to my extension 8003

 On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. bran...@brellsystems.comwrote:

 You call call to extension '246463' will not match 'exten =
 1246463'.

 On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez 
 bayardo.sanc...@gmail.com wrote:

 i create inbound confi my confi is:

 [incoming]
 exten= 1246463,,1,Dial(SIP/8003,60,rT)
 exten= 6463,1,Dial(SIP/8003,60,rT)
 exten= 1246463,,n,Wait(5)
 exten= 1246463,,n,Hangup

 but y calling and send this error in my CLI:

 [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
 Call from '101396_procall' to extension '246463' rejected because
 extension not found.

 --
 Bayardo Sánchez García
 Web Developer - Internet Portals - Asterisk Support - Windows Server
 Support - Proxi Support - Linux
 E-mail: bayardo.sanc...@gmail.com
 Linux User: #418392
 America Central - Managua, NI (505) 249-2853 -  4886876
 IM msn messenger: bjsanch...@hotmail.com
 Skype: bayardo.sanchez
 This email is intended solely for the person or organization to which it
 is addressed. It may contain privileged and confidential information. If you
 are not the intended recipient, you are prohibited from copying, disclosing
 or distributing this email or its contents (as it may be unlawful for you to
 do so) or taking any action in reliance on it. If you have received this
 email by mistake, please delete it. All e-mail sent to this address will be
 received by B.S. Solution e-mail system and is subject to archiving and
 review by someone other than the recipient.

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 --
 Bayardo Sánchez García
 Web Developer - Internet Portals - Asterisk Support - Windows Server
 Support - Proxi Support - Linux
 E-mail: bayardo.sanc...@gmail.com
 Linux User: #418392
 America Central - Managua, NI (505) 249-2853 -  4886876
 IM msn messenger: bjsanch...@hotmail.com
 Skype: bayardo.sanchez
 This email is intended solely for the person or organization to which it is
 addressed. It may contain privileged and confidential information. If you
 are not the intended recipient, you are prohibited from copying, disclosing
 or distributing this email or its contents (as it may be unlawful for you to
 do so) or taking any action in reliance on it. If you have received this
 email by mistake, please delete it. All e-mail sent to this address will be
 received by B.S. Solution e-mail system and is subject to archiving and
 review by someone other than the recipient.

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Re: [asterisk-users] inbound filed

2009-04-15 Thread Brandon B. 
Is your system configured to send the dialed calls to the [incoming]
context?


On Wed, Apr 15, 2009 at 11:22 AM, Bayardo Sanchez bayardo.sanc...@gmail.com
 wrote:

 nothing

 [Apr 15 11:24:15] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
 Call from '101396_procall' to extension '246463' rejected because
 extension not found.



 On Wed, Apr 15, 2009 at 11:07 AM, Brandon B. bran...@brellsystems.comwrote:

 Try this:

 [incoming]
 exten= 246463,1,Dial(SIP/8003,60,rT)
 exten= 246463,n,Wait(5)
 exten= 246463,n,Hangup
 exten= 6463,1,Dial(SIP/8003,60,rT)



 On Wed, Apr 15, 2009 at 11:00 AM, Bayardo Sanchez 
 bayardo.sanc...@gmail.com wrote:

 i call my tollfree number and send the call to my extension 8003

 On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. 
 bran...@brellsystems.comwrote:

 You call call to extension '246463' will not match 'exten =
 1246463'.

 On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez 
 bayardo.sanc...@gmail.com wrote:

 i create inbound confi my confi is:

 [incoming]
 exten= 1246463,,1,Dial(SIP/8003,60,rT)
 exten= 6463,1,Dial(SIP/8003,60,rT)
 exten= 1246463,,n,Wait(5)
 exten= 1246463,,n,Hangup

 but y calling and send this error in my CLI:

 [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383
 handle_request_invite: Call from '101396_procall' to extension 
 '246463'
 rejected because extension not found.

 --
 Bayardo Sánchez García
 Web Developer - Internet Portals - Asterisk Support - Windows Server
 Support - Proxi Support - Linux
 E-mail: bayardo.sanc...@gmail.com
 Linux User: #418392
 America Central - Managua, NI (505) 249-2853 -  4886876
 IM msn messenger: bjsanch...@hotmail.com
 Skype: bayardo.sanchez
 This email is intended solely for the person or organization to which
 it is addressed. It may contain privileged and confidential information. 
 If
 you are not the intended recipient, you are prohibited from copying,
 disclosing or distributing this email or its contents (as it may be 
 unlawful
 for you to do so) or taking any action in reliance on it. If you have
 received this email by mistake, please delete it. All e-mail sent to this
 address will be received by B.S. Solution e-mail system and is subject to
 archiving and review by someone other than the recipient.

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 --
 Bayardo Sánchez García
 Web Developer - Internet Portals - Asterisk Support - Windows Server
 Support - Proxi Support - Linux
 E-mail: bayardo.sanc...@gmail.com
 Linux User: #418392
 America Central - Managua, NI (505) 249-2853 -  4886876
 IM msn messenger: bjsanch...@hotmail.com
 Skype: bayardo.sanchez
 This email is intended solely for the person or organization to which it
 is addressed. It may contain privileged and confidential information. If you
 are not the intended recipient, you are prohibited from copying, disclosing
 or distributing this email or its contents (as it may be unlawful for you to
 do so) or taking any action in reliance on it. If you have received this
 email by mistake, please delete it. All e-mail sent to this address will be
 received by B.S. Solution e-mail system and is subject to archiving and
 review by someone other than the recipient.

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 --
 Bayardo Sánchez García
 Web Developer - Internet Portals - Asterisk Support - Windows Server
 Support - Proxi Support - Linux
 E-mail: bayardo.sanc...@gmail.com
 Linux User: #418392
 America Central - Managua, NI (505) 249-2853 -  4886876
 IM msn messenger: bjsanch...@hotmail.com
 Skype: bayardo.sanchez
 This email is intended solely for the person or organization to which it is
 addressed. It may contain privileged and confidential information. If you
 are not the intended recipient, you are prohibited from copying, disclosing
 or distributing this email or its contents (as it may be unlawful for you to
 do so) or taking any action in reliance on it. If you have received this
 email by mistake, please delete it. All e-mail sent to this address will be
 received by B.S. Solution e-mail system

Re: [asterisk-users] problema con una x100p

2009-04-02 Thread Brandon B. 
nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y
/etc/asterisk/chan/dahdi.conf



2009/4/2 Manolet Gmail mano...@gmail.com

 Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic

 Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
 descague compile e instale lo siguiente:

 asterisk-1.4.24
 dahdi-linux-2.1.0.4
 dahdi-tools-2.1.0.2
 libpri-1.4.9

 Sin embargo no logro configurar la tarjeta con exito, ya probe casi  todo.

 Esto aparece si ejecuto lspci:
 04:06.0 Communication controller: Motorola Wildcard X100P

 dahdi_hardware me muestra:
 pci::04:06.0 wcfxo-   1057:5608 Wildcard X100P

 dahdi_cfg -v :
 DAHDI Tools Version - 2.1.0.2

 DAHDI Version: 2.1.0.4
 Echo Canceller(s):
 Configuration
 ==


 0 channels to configure.

 dahdi_scan:
 [1]
 active=yes
 alarms=UNCONFIGURED
 description=DAHDI_DUMMY/1 (source: HRtimer) 1
 name=DAHDI_DUMMY/1
 manufacturer=
 devicetype=DAHDI Dummy Timing
 location=
 basechan=1
 totchans=0
 irq=0

 Nada parece funcionar y realmente no se donde esta el error... alguna idea?

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Re: [asterisk-users] problema con una x100p

2009-04-02 Thread Brandon B. 
Follow instructions from the following line to configure Asterisk 1.2 with
zaptel drivers for the X100P. If you are using dahdi-linux drivers instead
of zaptel, then instead of zaptel.conf you need to have a properly
configured /etc/dahdi/system.conf and instead of /etc/asterisk/zapata.conf
use this file instead /etc/asterisk/chan_dahdi.conf.

   http://users.telenet.be/Asterisk-PBX/InstallWildcard.htm

Since it sounds like you have not used Asterisk before, you should
read Asterisk:
the future of telephony http://astbook.asteriskdocs.org/ and look through
the pages at voip-info.org to get started.





2009/4/2 Manolet Gmail mano...@gmail.com

 system.conf:

 # Global data

 loadzone= us
 defaultzone = us

 el archivo /etc/asterisk/chan/dahdi.conf no existe, estoy usando
 asterisk 1.2 de todas maneras como dije ztcfg -v tampoco muestra la
 tarjeta, asi que de ninguna forma la va a ver asterisk, si hago en
 asterisk dahdi show channels no aparece nada.

 2009/4/2 Brandon B. bran...@brellsystems.com:
  nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y
  /etc/asterisk/chan/dahdi.conf
 
 
 
  2009/4/2 Manolet Gmail mano...@gmail.com
 
  Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic
 
  Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
  descague compile e instale lo siguiente:
 
  asterisk-1.4.24
  dahdi-linux-2.1.0.4
  dahdi-tools-2.1.0.2
  libpri-1.4.9
 
  Sin embargo no logro configurar la tarjeta con exito, ya probe casi
 todo.
 
  Esto aparece si ejecuto lspci:
  04:06.0 Communication controller: Motorola Wildcard X100P
 
  dahdi_hardware me muestra:
  pci::04:06.0 wcfxo-   1057:5608 Wildcard X100P
 
  dahdi_cfg -v :
  DAHDI Tools Version - 2.1.0.2
 
  DAHDI Version: 2.1.0.4
  Echo Canceller(s):
  Configuration
  ==
 
 
  0 channels to configure.
 
  dahdi_scan:
  [1]
  active=yes
  alarms=UNCONFIGURED
  description=DAHDI_DUMMY/1 (source: HRtimer) 1
  name=DAHDI_DUMMY/1
  manufacturer=
  devicetype=DAHDI Dummy Timing
  location=
  basechan=1
  totchans=0
  irq=0
 
  Nada parece funcionar y realmente no se donde esta el error... alguna
  idea?
 
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Re: [asterisk-users] PRI problem

2009-03-31 Thread Brandon B. 
Try a T1 crossover cable:

http://www.voip-info.org/wiki/view/crossover+T1+cable

On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.bizwrote:

 Hi guys,

 I've been trying to get my ISDN-10 line up for the past few days, but
 its been going up and down. I am using  OpenVox  D110P  card  on
 asterisk version 1.4.21. It seems to me like a cable problem. I tried
 using Ethernet straight cable (12, 45, 36, 78) and also a straight
 cable where the twisted pairs are on 12, 34, 56 and 78. The problem
 remains the same.

 /*etc/zaptel.conf*
 loadzone=sg
 defaultzone=sg

 # PRI Span
 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15
 dchan=16
 bchan=17-31


 */etc/asterisk/zapata.conf*
 language=en
 progzone=sg
 musiconhold=default

 ; PRI Set Up
 context=inbound-pri1
 switchtype=euroisdn
 signalling=pri_cpe
 pridialplan=national
 overlapdial=yes
 immediate=no
 faxdetect=both
 overlapdial=no
 usecallerid=yes
 usecallingpres=yes
 callerid=asreceived
 group=9
 channel = 1-15
 channel = 17-31


 The following are the messages that keep repeating.

  == Primary D-Channel on span 1 down
 Mar 31 14:34:05 WARNING[2361]: chan_zap.c:2682 pri_find_dchan: No
 D-channels available!  Using Primary channel 16 as D-channel anyway!
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 1
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 2
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 3
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 4
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 5
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 6
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 7
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 8
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 9
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 10
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 11
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 12
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 13
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 14
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 15
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 17
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 18
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 19
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 20
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 21
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 22
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 23
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 24
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 25
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 26
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 27
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 28
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 29
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 30
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 31
 Mar 31 14:34:07 NOTICE[2361]: chan_zap.c:9112 pri_dchannel: PRI got
 event: No more alarm (5) on Primary D-channel of span 1
  == Primary D-Channel on span 1 up
  == Restart on requested on entire span 1
 Mar 31 14:34:08 NOTICE[2361]: chan_zap.c:9112 pri_dchannel: PRI got
 event: HDLC Abort (6) on Primary D-channel of span 1
 Mar 31 14:34:08 WARNING[2362]: chan_zap.c:6899 handle_init_event:
 Detected alarm on channel 1: Red Alarm
 Mar 31 14:34:08 WARNING[2362]: chan_zap.c:6899 handle_init_event:
 Detected alarm on channel 2: Red Alarm
 Mar 31 14:34:08 WARNING[2362]: chan_zap.c:6899 handle_init_event:
 Detected alarm on channel 3: Red Alarm
 Mar 31 14:34:08 WARNING[2362]: chan_zap.c:6899 handle_init_event:
 Detected alarm on channel 4: Red Alarm
 Mar 31 14:34:08 WARNING[2362]: chan_zap.c:6899 handle_init_event:
 Detected alarm on channel 5: Red Alarm
 Mar 31 14:34:08 WARNING[2362]: chan_zap.c:6899

Re: [asterisk-users] IAX problem through intermediate asterisk box

2009-03-26 Thread Brandon B. 
Here's my troubleshooting help -- since the problem sounds like a timing
issue and part of the call is being trunked, then fix your timing problem,
or remove the trunking from A and B then see if the problem goes away.

On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman andrew.hak...@gmail.comwrote:

 So no one else has a problem routing IAX traffic through an
 intermediate Asterisk server? Does anyone else use Asterisk in such a
 configuration?

 On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman andrew.hak...@gmail.com
 wrote:
  I'm having a problem with IAX running through an intermediate asterisk
  box. Perhaps a small diagram will explain the situation better:
 
  *A --- [cloud (public internet)] --- *B [cloud
  (private network)]--- *C
 
  Asterisk server's A, B, and C, are all connected together with IAX
  All asterisk servers are 1.6.0.6
  Server A and B are geographically close, but connected over the public
 internet.
  Server B and C are geographically far, but connected over a private
 network.
  (the latency between A and B, and B and C are roughly equal)
 
  Each server has at least 1 phone hanging off of it, with A and C
  having most of the phones (B only has a couple).
  A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
 
  Phoning from A to B (or vice versa) works well, as does phoning from B
  to C (and vice versa). Calls can be placed for an indefinite amount of
  time and everything works great.
 
  The problem arises when phoning from A through B to C (or vice versa).
  For the first small amount of time (which can vary on a call to call
  basis, and lasts from 0 seconds to 3 minutes or so) everything is
  fine. After this, the audio in both directions gets garbled, and
  starts arriving in spurts. Once this happens, it continues forever.
  The audio never returns to normal no matter how long you wait.
 
  A to B uses IAX with trunking. B to C is not using trunking
  (dahdi_dummy is not working well on C for some reason - the module
  loads, but no /dev/dahdi is ever created). The same behavior happens
  when A to B is not using trunking either.
 
  Usually only 1 call is being placed at a time. An interesting thing
  happens when 2 testcalls are in progress at the same time though. If
  there's a call from A to B, and a call from A to C is made, once the
  call from A to C becomes garbled, so does the A to B call. When the A
  to C call is ended, the A to B call clears up. Ending the A to B call
  first does not improve the A to C call.
 
  The dialplans are setup so each server passes all non-local extensions
  to it's neighbor.
 
  Hence, for A, the relevant part of the dialplan is
 
  exten = _2XXX,1,Verbose(1|Extension 2xxx)
  exten = _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
  exten = _2XXX,n,Hangup()
 
  exten = _3XXX,1,Verbose(1|Extension 3xxx)
  exten = _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
  exten = _3xxx,n,Hangup()
 
  For B:
 
  exten = _1XXX,1,NoOp()
  exten = _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
  exten = _1XXX,n,Hangup()
 
  exten = _3xxx,1,NoOp()
  exten = _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
  exten = _3xxx,n,Hangup()
 
 
  For C:
  exten = _2XXX,1,Verbose(1|Extension 2xxx)
  exten = _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
  exten = _2XXX,n,Hangup()
 
  exten = _1XXX,1,Verbose(1|Extension 1xxx)
  exten = _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
  exten = _1XXX,n,Hangup()
 
  Is this the proper way to set such a configuration up? Is there a
  better way to call from A through B to C that would work better?
  Anyone else experience total audio breakup after a while with a
  similar arrangement? Why does it work initially for up to about 3
  minutes, then completely fall apart?
 
  Thanks,
  Andrew
 

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[asterisk-users] Unrecognized prilocaldialplan error when dialing a SIP call to a PRI trunk

2009-03-24 Thread Brandon B. 
Asterisk 1.6.0.6 with dahdi 2.1.0.4 is showing a strange Unrecognized
prilocaldialplan error with the SIP username when a SIP call is dialed to a
PRI trunk. The error shows up like this:

Unrecognized prilocaldialplan TON modifier: a
Unrecognized prilocaldialplan TON modifier: b
Unrecognized prilocaldialplan TON modifier: c

Where abc is the SIP username.

Is this a bug where the SIP username is somehow becoming a dialed number
prefix, as described here in chan_dahdi.conf

; pridialplan may be also set at dialtime, by prefixing the dialled
number with
; one of the following letters:
; U - Unknown
; I - International
; N - National
; L - Local (Net Specific)
; S - Subscriber
; V - Abbreviated
; R - Reserved (should probably never be used but is included for
completeness)

I'm planning on changing the prilocaldialplan to try attempt to resolve this
problem. On this system pridialplan and prilocaldialplan are not changed, so
they are set to the default values. Has anyone else come across this error
and what was the change that fixed it?

Brandon B.
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Re: [asterisk-users] PRI dropping

2009-03-24 Thread Brandon B. 
Is your PRI dropping calls, or are the unused B channels resetting? What is
your resetinterval in the /etc/asterisk/zapata.conf?



On Tue, Mar 24, 2009 at 3:21 PM, Harry Vangberg ha...@vangberg.name wrote:

 Hello,

 I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo
 cancellation. Every 30-60 minutes I experience PRI dropping.

 @@@ /etc/zaptel.conf:
 loadzone=dk
 defaultzone=dk

 span=2,1,0,ccs,hdb3,crc4
 bchan=32-46
 dchan=47
 bchan=48-62
 @@@

 @@@ /etc/asterisk/zapata.conf
 [channels]
 switchtype=euroisdn
 usecallerid=yes

 group=2
 signalling=pri_cpe
 context=incoming
 channel = 32-46
 channel = 48-62
 @@@

 With PRI debug turned out I get this output:

 @@@
 q931.c:2755 q931_restart: call 32768 on channel 1 enters state 62 (Restart)
  Protocol Discriminator: Q.931 (8)  len=13
  Call Ref: len= 2 (reference 0/0x0) (Originator)
  Message type: RESTART (70)
  [18 03 a9 83 81]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive
  Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0  Number Specified  Channel Type:
 3
Ext: 1  Channel: 1 ]
  [79 01 80]
  Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
 Channel (0) ]
  Protocol Discriminator: Q.931 (8)  len=13
  Call Ref: len= 2 (reference 0/0x0) (Terminator)
  Message type: RESTART ACKNOWLEDGE (78)
  [18 03 a1 83 81]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
 Preferred  Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0  Number Specified  Channel Type:
 3
Ext: 1  Channel: 1 ]
  [79 01 80]
  Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
 Channel (0) ]
 -- Processing IE 24 (cs0, Channel Identification)
 -- Processing IE 121 (cs0, Restart Indicator)
 q931.c:3581 q931_receive: call 32768 on channel 1 enters state 0 (Null)
-- B-channel 0/1 successfully restarted on span 2
 @@@

 For each and every channel (see full output at http://sprunge.us/fUEd )

 I have had two technicians from my telco out here and measuring.
 Everything is fine with the connection.

 Best regards,

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Re: [asterisk-users] what is the effect of high LBO settings?

2009-03-18 Thread Brandon B. 
On Mon, Mar 2, 2009 at 3:07 PM, Brandon B. bran...@brellsystems.com wrote:

 On Fri, Feb 27, 2009 at 7:49 PM, Jared Smith jsm...@digium.com wrote:

  As I understand it, the LBO is effectively an attenuation value, with a
 higher number meaning less attenuation.  This way, you don't get too hot
 of a signal with a short cable, or two low of a signal on long cable.

 Just how far is your Asterisk box from the demarcation point?


 This system is connected to a CSU in the same room that provides the
 physical T1 line. I've always set the LBO setting at 0 for this  because
 I've never had a long line to deal with. Since 0 works for me, I'm going to
 assume it's the correct setting with the demarc point (i.e. the Paradyne
 CSU) in the same room -- right? It's slightly confusing with settings 5,6,7
 labelled CSU and no description as to when to use those levels. Could you
 provide any suggestion for when levels 5,6,7 would be appropriate?

 From what you say an LBO setting of 5 would boost the signal level, which
 could be hot. Is there any chance this would cause the card to fail after
 a while? It appears this site just had 4 port Digium card fail today.


Turns out this problem was not the card, but another hardware issue. The
hard disk with Reiserfs eventually cratered and took the system down, and
the entire filesystem appears to be unrecoverable.

I've changed the LBO settings from 5 to 0, and the system is working fine.
If anyone has anything to add regarding the effect of different LBO settings
I think it might be helpful to have this documented.

Brandon B.
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Re: [asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

2009-03-03 Thread Brandon B. 
Use something like this:

exten = 100,1,Dial(SIP/100)
exten = 1001,1,Set(CALLER=${CALLERID(NUM)})
exten = 1001,2,Dial(SIP/100)
exten = 1001,3,Goto(default,${CALLER})

Brandon B.

On Tue, Mar 3, 2009 at 8:06 PM, James Mutuku jnmut...@gmail.com wrote:

 Hellos,

 I want to configure asterisk so that if exten A transfers a call to exten
 B, and B is either busy or the call is not answered, the call returns back
 to A. Is this possible?

 Please help
 James



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Re: [asterisk-users] what is the effect of high LBO settings?

2009-03-02 Thread Brandon B. 
On Fri, Feb 27, 2009 at 7:49 PM, Jared Smith jsm...@digium.com wrote:

 On Fri, 2009-02-27 at 14:07 -0700, Brandon B. wrote:
  As of yet, I am unwilling to change the LBO to 0 to where it probably
  should be because the system is working and I'm not sure exactly what
  the LBO does. I'm aware some changes were made to deal with low audio
  levels.

 LBO stands for Line Built Out... it's essentially a measurement of the
 distance between your demarcation point (d-marc/smart jack/NIU) and your
 Asterisk box.  As you can see from a sample system.conf (from DAHDI) or
 zaptel.conf (from Zaptel), it's an integer value from the following
 table:

 0: 0 db (CSU) / 0-133 feet (DSX-1)
 1: 133-266 feet (DSX-1)
 2: 266-399 feet (DSX-1)
 3: 399-533 feet (DSX-1)
 4: 533-655 feet (DSX-1)
 5: -7.5db (CSU)
 6: -15db (CSU)
 7: -22.5db (CSU)

 As I understand it, the LBO is effectively an attenuation value, with a
 higher number meaning less attenuation.  This way, you don't get too hot
 of a signal with a short cable, or two low of a signal on long cable.

 Just how far is your Asterisk box from the demarcation point?


This system is connected to a CSU in the same room that provides the
physical T1 line. I've always set the LBO setting at 0 for this  because
I've never had a long line to deal with. Since 0 works for me, I'm going to
assume it's the correct setting with the demarc point (i.e. the Paradyne
CSU) in the same room -- right? It's slightly confusing with settings 5,6,7
labelled CSU and no description as to when to use those levels. Could you
provide any suggestion for when levels 5,6,7 would be appropriate?

From what you say an LBO setting of 5 would boost the signal level, which
could be hot. Is there any chance this would cause the card to fail after
a while? It appears this site just had 4 port Digium card fail today.

 Also, I am trying to cross connect with another Asterisk system with
  the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the
  systems aren't seeing each other at all. Could the side with the high
  LBO be confusing the other side somehow?

 Shouldn't be... you did use a T1-crossover cable to cross-connect the
 two Asterisk boxes, right?  I've got a little T1 cross-connect diagram
 at http://www.asteriskdocs.org/cables/ if you need a reference.


Yes, it's a T1 cross over and this problem was resolved after plugging into
a working Digium card on the other end.

Brandon B.
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Re: [asterisk-users] changing /etc/dahdi/system.conf

2009-02-27 Thread Brandon B. 
Great -- thanks for that.

Brandon.

On Fri, Feb 27, 2009 at 8:34 AM, John Todd jt...@digium.com wrote:


 On Feb 26, 2009, at 3:05 PM, Tzafrir Cohen wrote:

  On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote:
  At the top of my /etc/dahdi/system.conf file is this line:
 
 # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25
  18:25:10 2009
  -- do not hand edit
 
  OK, so how do I adjust the timing source and LBO numbers, and echo
  cancellers if I'm not supposed to edit this file?
 
  Hmm I guess you should n't take that too seriously :-(
 
  Maybe this text should be changed.



 Go, go, gadget bugtracker!

 http://bugs.digium.com/view.php?id=14569

 JT

 ---
 John Todd   
 email:jt...@digium.comemail%3ajt...@digium.com
 Digium, Inc. | Asterisk Open Source Community Director
 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
 direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] what is the effect of high LBO settings?

2009-02-27 Thread Brandon B. 
I'm working on an Asterisk system with all of it's PRI ports configured with
the LBO setting at 5, like this:

span=3,0,5,esf,b8zs

As of yet, I am unwilling to change the LBO to 0 to where it probably should
be because the system is working and I'm not sure exactly what the LBO does.
I'm aware some changes were made to deal with low audio levels. Does anybody
know  what audio effects could this possibly have with short cabling? Also,
I am trying to cross connect with another Asterisk system with the normal
LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the systems aren't
seeing each other at all. Could the side with the high LBO be confusing the
other side somehow?

Brandon.
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[asterisk-users] changing /etc/dahdi/system.conf

2009-02-26 Thread Brandon B. 
At the top of my /etc/dahdi/system.conf file is this line:

# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009
-- do not hand edit

OK, so how do I adjust the timing source and LBO numbers, and echo
cancellers if I'm not supposed to edit this file?

Brandon.
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[asterisk-users] codec_dahdi and Asterisk 1.6.0.6

2009-02-25 Thread Brandon B. 
I've got a question about codec_dahdi witrh a system running Asterisk
1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to
route calls between different PRI connections, so no transcoding between
codecs is happening as far as I am aware.

1) How can I use codec_dahdi? Would it be useful when passing a call from
one dahdi channel to another dahdi channel?

2) I'm getting the following error on the Asterisk CLI unless I disable the
codec_dahdi module:

ERROR[18854]: codec_dahdi.c:399 find_transcoders: Failed to open
/dev/dahdi/transcode: No such file or directory

Is this because I do not have a hardware trancoding device? Can I safely
ignore this error or is it a bug?

Brandon.
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