date:20100105

On Monday 04 January 2010 07:16:49 Joseph L. Casale wrote:
 Looking at the source in the rpms from the asterisk package site
 appears that oslec is not built and enabled for the kmod rpms.
 
 Anyone know an existing repo or have direction on how to enable
 this to built for those rpms?
 

I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but you can 
check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm tool to 
build from an svn checkout if you already have a build setup configured.

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


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Re: [asterisk-users] ZapRAS priviledge error

2010-01-05 Thread Will Payne


Another day, another error..

Am now getting:

Plugin zaptel.so loaded.
Zaptel Plugin Initialized
Using zaptel device 'stdin'
Zaptel device is 'stdin'
Unable to put device 'stdin' into HDLC mode


Should ZapRAS see the channel as stdin and not /dev/zap/x?

Will



On 4 Jan 2010, at 16:46, Will Payne wrote:

 
 On 4 Jan 2010, at 16:28, Kevin P. Fleming wrote:
 
 Will Payne wrote:
 
 I'm looking to periodically nudge Asterisk into making an ISDN
 connection, setting up PPP and then (possibly by then starting an AGI
 script) grabbing a file by FTP over the PPP link.
 
 If I'm overcomplicating it or going about it completely the wrong way, a
 point in the right direction would be nice :)
 
 It is doubtful you'll be able to accomplish that, certainly not without
 some seriously ugly hacking. First off, I don't think that PPPD will
 even be invoked with the proper arguments for it to be the 'client' end
 of the connection, but even if it is, the Asterisk dialplan will halt
 execution until PPPD returns, so there's no way you are going to be able
 to execute an AGI or System() or anything to take actions over the PPP link.
 
 Unfortunately, an ugly hack might have to do..
 
 Params - should be able to work around this, even if I have to use a wrapper.
 
 PPPD halting the dialplan - I'll fork off a different process to watch for a 
 connection and make the transfer. I can just tell pppd to connect for a 
 minimum of 'x' seconds and then let Asterisk hang up.
 
 .. which still leaves me in the same position of wondering why I'm getting 
 this Device 'stdin' does not appear to be a zaptel device error...
 
 Will
 
 
 
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Re: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

2010-01-05 Thread Asterisk

I've tried several different qualify settings (including 10), but it didn't 
change the situation much :(.

Regards, Alex


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, January 05, 2010 2:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming 
randomly UNREACHABLE

Have you tried something like qualify=10 ?
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Re: [asterisk-users] Dahdi and oslec

I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup 
configured.

Anthony,
I appreciate the pointer, and I do have a build environment but am not 100%
sure how to accomplish this under CentOS with your files. Can you elaborate
a bit to get me started?

Thank you very much!
jlc


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[asterisk-users] Realtime LDAP Queues crashes

2010-01-05 Thread Jorge Salamero Sanz

Hi all,

I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other 
attributes needed for a working LDAP backend (I'll open a bug to include these 
changes on svn).

SIP users and dialplan are perfectly working, but when I call a queue the 
whole Asterisk (1.6.2.0) crashes:

on extconfig:

[settings]
sipusers = ldap,dc=nodomain,sip
sippeers = ldap,dc=nodomain,sip
extensions = ldap,dc=nodomain,extensions
voicemail = ldap,dc=nodomain,voicemail
queue_members = ldap,dc=nodomain,queue_member
queues = ldap,dc=nodomain,queue

on res_ldap.conf: see [1]

for the Queues on LDAP I have:

ou=Queues,dc=nodomain
ou: Queues
objectClass: top
objectClass: organizationalUnit

cn=foobar,ou=Queues,dc=nodomain
objectClass: applicationProcess
objectClass: AsteriskQueue
AstQueueName: foobar
AstQueueContext: default
AstQueueTimeout: 180
cn: foobar

the dialplan (on extensions.conf, the same if it's on LDAP):

[frontdesk]
exten = 78,1,Answer
exten = 78,n,Queue(foobar)
exten = 78,n,Hangup

[default]
include = common
include = frontdesk
switch = Realtime

and the user on LDAP:

uid=foo,ou=Users,dc=nodomain
cn: foo foo
uid: foo
sn: foo
uidNumber: 2002
gidNumber: 1901
homeDirectory: /nonexistent
userPassword: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
eboxSha1Password: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
eboxMd5Password: {MD5}rL0Y20zC+Fzt72VPzMSk2A==
eboxLmPassword: 5BFAFBEBFB6A0942AAD3B435B51404EE
eboxNtPassword: AC8E657F83DF82BEEA5D43BDAF7800CC
eboxDigestPassword: {MD5}x0Z+Prb70OIF3iARsuJ3Xg==
eboxRealmPassword: {MD5}c7467e3eb6fbd0e205de2011b2e2775e
givenName: foo
description: foo
AstAccountType: friend
AstAccountContext: users
AstAccountCallerID: 1001
AstAccountMailbox: 1001
AstAccountHost: dynamic
AstAccountNAT: yes
AstAccountQualify: yes
AstAccountCanReinvite: no
AstAccountDTMFMode: rfc2833
AstAccountInsecure: port
AstAccountLastQualifyMilliseconds: 0
AstAccountIPAddress: 0.0.0.0
AstAccountPort: 0
AstAccountExpirationTimestamp: 0
AstAccountRegistrationServer: 0
AstAccountUserAgent: 0
AstAccountFullContact: sip:0.0.0.0
AstContext: users
AstVoicemailMailbox: 1001
AstVoicemailPassword: 1001
AstVoicemailEmail: u...@domain
AstVoicemailAttach: yes
AstVoicemailDelete: no
AstQueueMembername: foobar
AstQueueMemberof: foobar
objectClass: AsteriskQueueMember
objectClass: AsteriskSIPUser
objectClass: AsteriskVoiceMail
objectClass: inetOrgPerson
objectClass: passwordHolder
objectClass: posixAccount
AstQueueInterface: SIP/1001

when i call the queue extension, on slapd I can see how Asterisk fetches the 
AsteriskQueue objectClass, and then fetches the foo user, but then crashes 
like this:

-- Executing [...@users:1] Answer(SIP/demo-, ) in new stack
-- Executing [...@users:2] Queue(SIP/demo-, foobar) in new 
stack
[Jan  5 13:26:28] WARNING[6195]: app_queue.c:1134 create_queue_member: No 
location at interface ''
[1]6124 segmentation fault (core dumped)  asterisk -
vvc

*CLI queue show foobar
[1]6356 segmentation fault (core dumped)  asterisk -
vvc

*CLI queue add member SIP/foo to foobar
[1]6394 segmentation fault (core dumped)  asterisk -
vvc

any clue on what's wrong ? how could i debug this ? maybe there is some 
attribute missing ? or the LDAP schema is wrong ? anyone with a working setup 
like this ?

thanks in advance !

[0] http://people.ebox-platform.com/~bencer/asterisk.ldif
[1] http://people.ebox-platform.com/~bencer/res_ldap.conf.mas

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Re: [asterisk-users] Inquiry:Asterisk different codec schemes?

2010-01-05 Thread Kevin P. Fleming

Olle E. Johansson wrote:

 But it's fairly common to have asymmetric media in the call. If the caller 
 offers A, B and C and the callee responds with B, the caller sends B but the 
 callee might send A.

Only for non-Asterisk endpoints, since Asterisk will never do this.

Is this really that common? I'd be surprised if an endpoint would want
to consume a G.729 encoder (for example) without a corresponding decoder
on the receive path... doing that would make managing DSP resources in
the endpoint much more complicated.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] CallerID on Indian PSTN is not working.

2010-01-05 Thread Arun Sasidhar

Hi,

I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
same problem listed here:  https://issues.asterisk.org/view.php?id=6683
There is one patch on this link but i don't know how to apply patch on
asterisknow. Is this patch will resolve my issue? Kindly help me to fix this
issue.

My log file showing this while an incoming call on PSTN

[Jan  5 18:14:59] DEBUG[9938] dsp.c: dsp busy pattern set to 0,0
[Jan  5 18:14:59] VERBOSE[9986] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan  5 18:15:01] NOTICE[9986] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan  5 18:15:02] NOTICE[9986] chan_dahdi.c: Got event 2 (Ring/Answered)...
[Jan  5 18:15:04] NOTICE[9986] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:1]
Set(DAHDI/1-1, __FROM_DID=s) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:2]
Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new
stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:3]
ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:4]
Set(DAHDI/1-1, FAX_RX=disabled) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:5]
Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:6]
SetCallerPres(DAHDI/1-1, allowed_not_screened) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing [...@from-pstn:7]
Goto(DAHDI/1-1, from-did-direct|104|1) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Goto
(from-did-direct,104,1)
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@from-did-direct:1] Macro(DAHDI/1-1, exten-vm|104|104) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing 
[...@macro-exten-vm:1]
Macro(DAHDI/1-1, user-callerid) in new stack
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in new stack
[Jan  5 18:15:04] DEBUG[9986] app_macro.c: Executed application: Set
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@macro-user-callerid:2] GotoIf(DAHDI/1-1, 0?report) in new stack
[Jan  5 18:15:04] DEBUG[9986] app_macro.c: Executed application: GotoIf
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@macro-user-callerid:3] ExecIf(DAHDI/1-1, 1|Set|REALCALLERIDNUM=) in
new stack
[Jan  5 18:15:04] DEBUG[9986] app_macro.c: Executed application: ExecIf
[Jan  5 18:15:04] DEBUG[9986] app_macro.c: Last app: Set|REALCALLERIDNUM=
[Jan  5 18:15:04] DEBUG[9986] func_db.c: DB: DEVICE//user not found in
database.
[Jan  5 18:15:04] VERBOSE[9986] logger.c: -- Executing
[...@macro-user-callerid:4] Set(DAHDI/1-1, AMPUSER=) in new stack
[Jan  5 18:15:04] DEBUG[9986] app_macro.c: Executed application: Set
.

And in asterisk console

-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new
stack
-- Executing [...@from-pstn:2] Gosub(DAHDI/1-1,
app-blacklist-check|s|1) in new stack
-- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, )
in new stack
-- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1,
0?blacklisted) in new stack
-- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1,
CALLED_BLACKLIST=1) in new stack
-- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new
stack
-- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1
|Set|CALLERID(name)=) in new stack
-- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new
stack
-- Executing [...@from-pstn:5] Set(DAHDI/1-1,
__CALLINGPRES_SV=allowed_not_screened) in new stack
-- Executing [...@from-pstn:6] SetCallerPres(DAHDI/1-1,
allowed_not_screened) in new stack
-- Executing [...@from-pstn:7] Goto(DAHDI/1-1, from-did-direct|104|1)
in new stack
-- Goto (from-did-direct,104,1)
-- Executing [...@from-did-direct:1] Macro(DAHDI/1-1,
exten-vm|104|104) in new stack
-- Executing [...@macro-exten-vm:1] Macro(DAHDI/1-1, user-callerid) in
new stack
-- Executing [...@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in
new stack
-- Executing

[asterisk-users] Newbie: MITEL and Asterisk

2010-01-05 Thread phiroc

Hello,

can the Asterisk API be used to automate a MITEL 5330 telephone?

If not, are there any other API which can used to do that?

Many thanks.

phiroc

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Re: [asterisk-users] Dialplans Holiday Dates

2010-01-05 Thread Danny Nicholas

When it is running, nerdvittles.com is an excellent resource for this kind
of question.  Voip-info.org is almost always up and has more technically
oriented answers to this type of query.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham
Sent: Thursday, December 31, 2009 9:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialplans  Holiday Dates

I have a working dialplan for our phone system with Mon-Fri, business 
hours identification, etc.  But what I'm lacking right now is support 
for company holiday dates.

What I'd like to do is to create a database of these dates and just 
update them as new years rollover.

I suspect others have done this sort of thing with Asterisk before, but 
I've not found any resources so far.

Does anyone have a suggestion as to how to approach this?  I'm running 
Asterisk 1.4.2.

Thanks
Myles
-- 
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona  USA
http://www.techsolusa.com
Phone +1-480-451-7440

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[asterisk-users] DTMF detection on dahdi with b4xxp (again, some more details)

2010-01-05 Thread Christian Theune

Hi,

I tried again getting DTMF detection on my ISDN devices with dahdi going 
again. I used the channel debug to see whether asterisk sees the frames 
and detects them as DTMF.

Interestingly here's what works:

1. GSM phone - chan_dahdi g1 - asterisk - can_sip - SIP phone

Both the GSM phone and the SIP phone can issue DTMF that will be 
detected as features (transfer)

2. GSM phone - chan_dahdi g1 - asterisk - chan_dahdi g4 - ISDN phone

The GSM phone can issue DTMF that will be detected. The ISDN phone 
won't. (That's my issue.) I don't see any messages of asterisk 
recognizing the DTMF frames when pressing the keys. I do hear the DMTF 
sound on both phones.

3. ISDN phone - chan_dahdi g4 - asterisk - chan_dahdi g1 - GSM phone

The ISDN phone can issue DTMF that will be recognized and so does the 
GSM phone.

So. When the ISDN phone is receiving a call on g4 its DTMF sounds won't 
be recognized. OTOH when the GSM phone on g1 is being called it's sounds 
are recognized.

Sounds like a configuration issue to me. Does anybody have an idea what 
to look out for?

Thanks in advance,
Christian

-- 
Christian Theune · c...@gocept.com
gocept gmbh  co. kg · forsterstraße 29 · 06112 halle (saale) · germany
http://gocept.com · tel +49 345 1229889 0 · fax +49 345 1229889 1
Zope and Plone consulting and development


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Re: [asterisk-users] T.38 ITSP?

2010-01-05 Thread David Backeberg

On Mon, Jan 4, 2010 at 6:44 PM, Karl Fife karlf...@gmail.com wrote:
 Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x
 instance AND do it reliably?  If so, I can think of a number of locations
 with copper loops that could be scrapped.  I'm actually quite surprised at
 what an underwhelming number of ITSP's that say they support T.38 (zero so
 far among my normal go-to companies).

 For locations that just want to be able to send (because they use
 fax-to-email for receive), It should be trivial to relay via T-38 to any of
 our asterisk servers that DO have a PRI or copper loop, and send (or
 queue-up) the fax to be sent.  It's the 'local number' for 'traditional
 receive' that looking to be the harder problem.

 Is anybody using an ITSP for inbound T-38 fax with 'local' numbers?

Methinks your difficulty finding an ITSP for this purpose is directly
related to said ITSP's collective experience trying to get this to
work for their customers. They've no doubt discovered that the support
costs for the many times that they have to deal with a broken T.38
implementation or an intermittently lossy end-to-end voip connection
are more than the marginal charges they can assess for the service.

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Re: [asterisk-users] No reply to SIP OPTIONS - sip pee rs becoming randomly UNREACHABLE

2010-01-05 Thread Tilghman Lesher

On Tuesday 05 January 2010 04:54:52 Asterisk wrote:
 I've tried several different qualify settings (including 10), but it didn't
 change the situation much :(.

Realize that qualify=10 is 10ms, not 10 seconds.  You probably want
something on the order of qualify=3000.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

2010-01-05 Thread Danny Nicholas

Hope I'm not the only one who doesn't know this;  is the time value MS
across the board?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Tuesday, January 05, 2010 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]No reply to SIP OPTIONS - sip peers becoming
randomly UNREACHABLE

On Tuesday 05 January 2010 04:54:52 Asterisk wrote:
 I've tried several different qualify settings (including 10), but it
didn't
 change the situation much :(.

Realize that qualify=10 is 10ms, not 10 seconds.  You probably want
something on the order of qualify=3000.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] No reply to SIP OPTIONS - sip pee rs becoming randomly UNREACHABLE

2010-01-05 Thread Tilghman Lesher

On Tuesday 05 January 2010 09:50:42 Danny Nicholas wrote:
 Tilghman Lesher wrote:
  On Tuesday 05 January 2010 04:54:52 Asterisk wrote:
   I've tried several different qualify settings (including 10), but it
   didn't change the situation much :(.
 
  Realize that qualify=10 is 10ms, not 10 seconds.  You probably want
  something on the order of qualify=3000.

 Hope I'm not the only one who doesn't know this;  is the time value MS
 across the board?

No, it's not, but there's a janitor project in trunk (for 1.8) to make the
time units in each setting a bit more clear.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] Inquiry:Asterisk different codec schemes?

2010-01-05 Thread Olle E. Johansson


5 jan 2010 kl. 10.08 skrev hadi motamedi:

 
 
 On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote:
 
 4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
 
  hadi motamedi wrote:
 
  Sorry . I didn't get the point clearly . In the SIP Invite message , it
  says my audio endpoint is IP x.x.x.x port x, and I can use codecs
  A,B,C. The remote endpoint responds with a 200 OK, saying my audio
  stream is at IP y.y.y.y port y, and I choose codec B. Can you please do
  me favor and let me know if my understanding is right or not ?
  Thank you
 
  No, you are not understanding the SDP offer/answer model properly. If
  one endpoint offers codecs A, B and C in its SDP, it is willing to
  *receive* media in those formats. The receiver of that offer can choose
  to send media to the offerer in any of those formats, at any time. If
  the answering endpoint includes only codec B in its SDP, then it is
  willing to *receive* only codec B. In that scenario, it is possible for
  media to flow from endpoint 1 to endpoint 2 using codec B, and from
  endpoint 2 to endpoint 1 using codec A (or C), but this will not happen
  if Asterisk is an endpoint in this scenario.
 
  When Asterisk receives a media frame, if the format of that frame is not
  the format that it is currently sending to the other endpoint, it will
  switch to that format automatically. If it cannot do so because the
  other endpoint did not offer to receive that format, then the call's
  audio will probably fail. This is the reason why I responded before that
  Asterisk does not support asymmetric formats in a media session.
 
  In reality, it is extremely uncommon for a SIP endpoint to want to send
  media in a format that it is not also willing to receive; in fact, I
  can't say I've ever seen this situation arise in any testing I've done
  or in any issues reported in our issue tracker.
 
 But it's fairly common to have asymmetric media in the call. If the caller 
 offers A, B and C and the callee responds with B, the caller sends B but the 
 callee might send A.
 
 /O
 
  
  
 Sorry . You mean we can have asymmetric codecs in Asterisk ?
  
As Kevin stated, for Asterisk, the server switches to the format we receive, so 
no. I just pointed out that it happens quite often that a call is asymmetric, 
and you will see Asterisk trying to follow the other side.

/O
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Re: [asterisk-users] Dahdi and oslec

I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but 
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup 
configured.

Anthony,
So this script builds them with the dahdi-tools-libs package requirement, I
thought the fedora spec built all of these? Any idea?

Thanks!
jlc

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[asterisk-users] Canadian call quality issue

 hello,

 we have been using a couple of US based
 VoIP providers for outbound calls completed
 within the US, without any issues.

 We recently started making calls to Canada
 and have received a few complaints about
 the call quality.

 Questions :

  - Could this be because of the number of
intermediate IP hops between us / our
VoIP provider and the Canadian phone
companies ?

  - Would choosing a Canadian VoIP provider
address / resolve this issue ?

 Thank you in advance.

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Re: [asterisk-users] T.38 ITSP?

2010-01-05 Thread Karl Fife

Sadly I suspect you're right.  I suspect the other business problem would be 
abuse. Anyone in that business would doubtless get their hands dirty trying 
to combat T.38 subscribers whose intention is to send Junk Faxes.

Flat-roof repair!  Employee vacation discounts!  Health insurance for small 
business!  The carrier might incur some duty-of-care ( legal risk) with 
regard to preventing junk faxes originating (for example) from two-bit GPRS 
data connection Nigeria.

-Karl


- Original Message - 
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Tuesday, January 05, 2010 8:37 AM
Subject: Re: [asterisk-users] T.38 ITSP?


On Mon, Jan 4, 2010 at 6:44 PM, Karl Fife karlf...@gmail.com wrote:
 Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x
 instance AND do it reliably? If so, I can think of a number of locations
 with copper loops that could be scrapped. I'm actually quite surprised at
 what an underwhelming number of ITSP's that say they support T.38 (zero so
 far among my normal go-to companies).

 For locations that just want to be able to send (because they use
 fax-to-email for receive), It should be trivial to relay via T-38 to any 
 of
 our asterisk servers that DO have a PRI or copper loop, and send (or
 queue-up) the fax to be sent. It's the 'local number' for 'traditional
 receive' that looking to be the harder problem.

 Is anybody using an ITSP for inbound T-38 fax with 'local' numbers?

Methinks your difficulty finding an ITSP for this purpose is directly
related to said ITSP's collective experience trying to get this to
work for their customers. They've no doubt discovered that the support
costs for the many times that they have to deal with a broken T.38
implementation or an intermittently lossy end-to-end voip connection
are more than the marginal charges they can assess for the service.

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[asterisk-users] Really Silly Question From Total Newbie

Hello All -

I've been poking around the past few weeks, trying to familiarize
myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
be complete.   This is my first exposure to all of these technologies.

I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
2400 and the install went well.   I can log in and poke around in
Linux and I even configured the box to be recognized on my windows
network.  However, is there a GUI that I can access to help me set
things up?  I've gotten so far as what looks to me like DOS windows
that I can change various things in the OS...

I do not have any other hardware installed.  No cards and no VoIP
phones.   I havent got to the point where I can make a test call or
anything like that.  I dont know how to tell if Asterisk is up and
running and how I can tweak it, etc.   I've been reading a lot of
different things, and have become a bit confused. I think that in time
it will come to me but I needed to stop and ask because I need to know
if I am on the wrong path for what I'd like to do someday

My main question is: CAN I make call from that box to my cell phone
using a soft-phone?   If so, how can I do that?   Also, can I use my
cell phone to call into that box?   I dont know if I have to get a
phone number, or do I NEED a phone number?   At the moment, I do not
have any dollars to throw at this project.   Its purely for learning,
proof of concept sort of thing for myself on my spare time in the
evenings.  I would simply like to be able to call out and be able to
call into that box.  Later on down the road maybe I will get into
setting up an IVR using a database so I can call into that system from
wherever and get information read back to me.  But, first things
first  I'd like to know if I am heading down the wrong path here.

Sorry for what might seem as really silly questions, but I am not sure
how to proceed.

Thanks in advance for any insight that you folks can provide!

Mike

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Re: [asterisk-users] Dahdi and oslec

2010-01-05 Thread Tzafrir Cohen

On Mon, Jan 04, 2010 at 01:16:49PM +, Joseph L. Casale wrote:
 Looking at the source in the rpms from the asterisk package site
 appears that oslec is not built and enabled for the kmod rpms.
 
 Anyone know an existing repo or have direction on how to enable
 this to built for those rpms?

  git clone http://git.tzafrir.org.il/git/dahdi-extra.git
  cd dahdi-extra
  make gen-patch

And use the generated dahdi_linux_extra.diff . It includes OSLEC and
some other things. See the Makefile there for more information. The
patch should be applied with -p1 .

This repository includes the extra DAHDI drivers currently included
directly in the Debian package.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread Randy R

On Tue, Jan 5, 2010 at 8:53 PM, UIT DEVELOPMENT uit...@gmail.com wrote:
 I've been poking around the past few weeks, trying to familiarize
 myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
 be complete.   This is my first exposure to all of these technologies.

I think one of the best things to do is to read this book:
http://oreilly.com/catalog/9780596009625

It will allow you to ask specific questions about stuff you may not
get but in the meantime it will tell you all the basic things about
what asterisk can do in terms a newbie can easily assimilate.

There are also a lot of web sites out there with tutorials about the
world of VoIP and Asterisk, and of course the IRC channel #asterisk on
Freenode.net

/r

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread Barry L. Kline

UIT DEVELOPMENT wrote:

 Sorry for what might seem as really silly questions, but I am not sure
 how to proceed.
 
 Thanks in advance for any insight that you folks can provide!

Hello Mike.

Welcome to the wonderful world of Asterisk.  Before you sludge through a
GUI and all the attendant bad habits that can produce, I suggest that
you download what we consider to be the Bible of Asterisk.  The infobot
on IRC says:

thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN
0-596-51048-9) --- Order yours at
http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF
http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at
http://astbook.asteriskdocs.org;

Download that and read the first few chapters.  It will make your
Asterisk experience a lot more enjoyable and will help you understand
what you're doing.

This list, and the IRC channel #asterisk, are good resources when you
finally get to the point where you're stuck and need some help.

Regards,

Barry

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread James A. Shigley

I can't help you two much with configuration of linux, but as to the call 
question. You will need some route for the server to be capable of 
sending/receiving calls. There is a couple of ways to do this cheaply. 

Buy a standard telephone modem (usb, pci, or serial). And plug into wall a 
jack. This will only allow you one call at a time. But if this is just a proof 
of concept that sounds like it will be plenty. Your number would be whatever 
the phone jack corresponds to.

Integrate it with something like skype.

Buy one of the products similar to a magic jack which will work with linux.

Or if you know someone who has a sip server already running. And is amiable to 
letting you piggy back off of it. Your  phone number would point to that 
person's Server (be it one of the friends that the person is lending you, or a 
number that you have forwarded/ported at/to them) with it set to forward to 
your asterisk server. from there you could use your dialplan to do whatever you 
wanted it to. And for outbound you would send the calls out thru the friends 
server via sip or iax.

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps, 
 
CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information =is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by reply to sender 
only message and destroy all electronic and hard copies of the communication, 
including attachments. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
Sent: Tuesday, January 05, 2010 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Really Silly Question From Total Newbie

.

My main question is: CAN I make call from that box to my cell phone
using a soft-phone?   If so, how can I do that?   Also, can I use my
cell phone to call into that box?   I dont know if I have to get a
phone number, or do I NEED a phone number?   At the moment, I do not
have any dollars to throw at this project.   Its purely for learning,
proof of concept sort of thing for myself on my spare time in the
evenings.  I would simply like to be able to call out and be able to
call into that box.  Later on down the road maybe I will get into
setting up an IVR using a database so I can call into that system from
wherever and get information read back to me.  But, first things
first  I'd like to know if I am heading down the wrong path here.
...

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Re: [asterisk-users] Really Silly Question From Total Newbie

On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

 CAN I make call from that box to my cell phone using a soft-phone?  If 
 so, how can I do that?

You need to get an account with a VOIP provider -- someone to accept your 
call via the Internet and place a call on the PSTN to call your cell 
number -- or any other number.

 Also, can I use my cell phone to call into that box?  I dont know if I 
 have to get a phone number, or do I NEED a phone number?

With the same account you can rent a PSTN number. When someone calls that 
number, they will call your server over the Internet.

This is assuming you don't have a fancy-dancy smart phone with a SIP 
client.

I use www.vitelity.net. I don't know their current pricing, but outbound 
calls are less than US$0.015 per minute and renting a PSTN number is 
around US$1.50 per month.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] SIP Listen Multiple Ports

2010-01-05 Thread Vikram Ragukumar

Steve Edwards wrote:
 On Sun, 3 Jan 2010, Steve Edwards wrote:

 You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple 
 addresses and ports and forward to Asterisk on the same or 
 different boxes.
 
 On Mon, 4 Jan 2010, Vikram Ragukumar wrote:

 Would it be more efficient to use libnetfilter_queue() to listen to 
 specific addresses / ports and forward to Asterisk?
 Yes, but the number of SIP control messages are usually insignificant 
 compared to all the RTP packets.
 
 On Mon, 4 Jan 2010, Vikram Ragukumar wrote:
 
 My customer is keen on using a hardware bridge to maximize throughput 
 and also allow multiple servers.  My boss is pressing me to maintain 
 Kamailio and rtpproxy compatibility, and understand the tradeoffs in 
 satisfying both.  The link below has a diagram showing the way I'm going 
 now:

 http://signalogic.com/images/openser_asterisk_sysconfig_dataflow.jpg

 Will the bridge preclude me from gracefully modifying my code to use 
 more Kamailio functionality, if needed?
 
 I won't claim to have any insight into your system, but...
 
 Your diagram shows all SIP messages (unencrypted and decrypted) flowing 
 through Kamailio. My guess is that you would have access to all Kamailio 
 features.

Steve,

Thank you for your prompt response.

If Kamailio is setup to listen on ports 5060 and 9090, port 9090 carries 
unknown SIP signaling information. Is it possible for Kamailio to dump 
these unrecognized signaling packets to a user space application which 
would process and return packets to Kamailio ?

Would it be better to use libnetfilter_queue() to handle the 
unrecognized signaling information prior to Kamailio ?

Thanks for all your help,
Regards,
Vikram.







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Re: [asterisk-users] Really Silly Question From Total Newbie

Will do Barry.  Thanks for the links!   Downloading now..   Mike

On Tue, Jan 5, 2010 at 3:25 PM, Barry L. Kline blkl...@attglobal.net wrote:
 UIT DEVELOPMENT wrote:

 Sorry for what might seem as really silly questions, but I am not sure
 how to proceed.

 Thanks in advance for any insight that you folks can provide!

 Hello Mike.

 Welcome to the wonderful world of Asterisk.  Before you sludge through a
 GUI and all the attendant bad habits that can produce, I suggest that
 you download what we consider to be the Bible of Asterisk.  The infobot
 on IRC says:

 thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN
 0-596-51048-9) --- Order yours at
 http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF
 http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at
 http://astbook.asteriskdocs.org;

 Download that and read the first few chapters.  It will make your
 Asterisk experience a lot more enjoyable and will help you understand
 what you're doing.

 This list, and the IRC channel #asterisk, are good resources when you
 finally get to the point where you're stuck and need some help.

 Regards,

 Barry

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Re: [asterisk-users] Really Silly Question From Total Newbie

Thanks Randy!

On Tue, Jan 5, 2010 at 3:25 PM, Randy R randulo2...@gmail.com wrote:
 On Tue, Jan 5, 2010 at 8:53 PM, UIT DEVELOPMENT uit...@gmail.com wrote:
 I've been poking around the past few weeks, trying to familiarize
 myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
 be complete.   This is my first exposure to all of these technologies.

 I think one of the best things to do is to read this book:
 http://oreilly.com/catalog/9780596009625

 It will allow you to ask specific questions about stuff you may not
 get but in the meantime it will tell you all the basic things about
 what asterisk can do in terms a newbie can easily assimilate.

 There are also a lot of web sites out there with tutorials about the
 world of VoIP and Asterisk, and of course the IRC channel #asterisk on
 Freenode.net

 /r

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Re: [asterisk-users] Really Silly Question From Total Newbie

James,

Thank you for the reply.  I do not have phone service in my home.
I've been 100% cell since 2003.  I do have an old analog phone - big
heavy thing...  If I connect it to the wall outlet there is nothing.
I've tried every outlet in the house.  I didnt expect to find a tone
as we've never connected phone service here.

My assumption, however stupid it might be, is that I could set this up
to make calls but as I began to read more and more I started reading
about gateways and other costly services that I was hoping to avoid --
for now at least.  This all began when my wife went food shopping
around the holiddays and wanted to know if we had something or not
already, and I wasnt home to confirm or not.   So I got to thinking,
if she could call the VoIP box and get one of those press 1 for
spices... press 2 for dry foods, press 3 for canned goods, press 4 for
snacks, press 5 for drinks. and it would access a
database that I already have set up with our groceries already in
there.   Yea - geeky, I know.  :-)

So that was the plan but first I needed to be able to get this thing
set up.  I THINK you're saying I need to purchase another service to
get myself to make calls.   I dont know anyone with a SIP server..
I'd rather keep it all inside our home if possible.  I've got a lot of
old hardware laying around and I do have MODEMs - internal and
external 56k types.   Thanks for the hints.  I'll check into Magic
Jack type stuff and see how it can help.

-M


On Tue, Jan 5, 2010 at 3:19 PM, James A. Shigley j...@answeringserv.com wrote:
 I can't help you two much with configuration of linux, but as to the call 
 question. You will need some route for the server to be capable of 
 sending/receiving calls. There is a couple of ways to do this cheaply.

 Buy a standard telephone modem (usb, pci, or serial). And plug into wall a 
 jack. This will only allow you one call at a time. But if this is just a 
 proof of concept that sounds like it will be plenty. Your number would be 
 whatever the phone jack corresponds to.

 Integrate it with something like skype.

 Buy one of the products similar to a magic jack which will work with linux.

 Or if you know someone who has a sip server already running. And is amiable 
 to letting you piggy back off of it. Your  phone number would point to that 
 person's Server (be it one of the friends that the person is lending you, or 
 a number that you have forwarded/ported at/to them) with it set to forward to 
 your asterisk server. from there you could use your dialplan to do whatever 
 you wanted it to. And for outbound you would send the calls out thru the 
 friends server via sip or iax.

 James Shigley
 Monroe Telephone Answering Service
 409-981-9213
 Infinity 5.51,UC 4.02.3803, Blink 3.0.104
 Ecreator:2.21, eResponse 1.1.7
 Webportal,WebApps,

 CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
 information which may be confidential or privileged. The information =is 
 intended to be for the use of the individual or entity named above. If you 
 are not the intended recipient, be aware that any disclosure, copying, 
 distribution or use of the contents of this information is prohibited. If you 
 have received this email in error, please notify the sender immediately by 
 reply to sender only message and destroy all electronic and hard copies of 
 the communication, including attachments.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
 Sent: Tuesday, January 05, 2010 1:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Really Silly Question From Total Newbie

 .

 My main question is: CAN I make call from that box to my cell phone
 using a soft-phone?   If so, how can I do that?   Also, can I use my
 cell phone to call into that box?   I dont know if I have to get a
 phone number, or do I NEED a phone number?   At the moment, I do not
 have any dollars to throw at this project.   Its purely for learning,
 proof of concept sort of thing for myself on my spare time in the
 evenings.  I would simply like to be able to call out and be able to
 call into that box.  Later on down the road maybe I will get into
 setting up an IVR using a database so I can call into that system from
 wherever and get information read back to me.  But, first things
 first  I'd like to know if I am heading down the wrong path here.
 ...

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Re: [asterisk-users] [asterisk-speech-rec] AGI and embargeability

2010-01-05 Thread Quinn Weaver

On Tue, Jan 5, 2010 at 12:20 AM, Trevor Benson tben...@a-1networks.com wrote:
 Its called speechbackground.  from asterisk console type 'core show 
 applications speech' (and hit the tab key) these are the speech applications 
 used.  Speechbackground being similar to background.

Thanks, Trevor, Steve, and Alex.  I should have been clearer about
what I want to do.  I want to play a sound file while listening for
*either* speech *or* DTMF input.  Either should affect control flow.
Think for customer service, press or say 2.  Is there any way to do
this?

My second constraint is that I want to do this within a single AGI
script.  I'd prefer using AGI commands rather than EXEC'ing dialplan
applications like Background and speechbackground, but I guess I can
use apps if I need to.

Thanks again,



 Trevor Benson
 A1 Networks  |  Network Engineer
 dCAP- Digium Certified Asterisk Professional
 LPIC-1, Network+, CNA, MCP
 DID (707)703-1041
 Fax (707)703-1983
 tben...@a-1networks.com





 On Jan 4, 2010, at 5:41 PM, Quinn Weaver wrote:

 Hi,

 This is a naive question, but is there a way in my AGI script to
 simultaneously play audio and listen for DTMF or voice responses?
 I've heard VOIP hackers call this inbargeability; it's the ability
 to barge in to a playing audio clip.

 I'm planning to use Lumenvox for the DTMF and voice recognition, BTW.
 Not sure if that matters.

 Many thanks to anyone who can lend me a clue about this,

 --
 Quinn Weaver Consulting, LLC
 Full-stack web design and development
 http://quinnweaver.com/
 510-520-5217

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-- 
Quinn Weaver Consulting, LLC
Full-stack web design and development
http://quinnweaver.com/
510-520-5217

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread Danny Nicholas

There are some free-trial and low-cost services out there.  Gizmo comes to
mind but buyer beware;  look through this site for recommendations and
warnings.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT
DEVELOPMENT
Sent: Tuesday, January 05, 2010 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Really Silly Question From Total Newbie

James,

Thank you for the reply.  I do not have phone service in my home.
I've been 100% cell since 2003.  I do have an old analog phone - big
heavy thing...  If I connect it to the wall outlet there is nothing.
I've tried every outlet in the house.  I didnt expect to find a tone
as we've never connected phone service here.

My assumption, however stupid it might be, is that I could set this up
to make calls but as I began to read more and more I started reading
about gateways and other costly services that I was hoping to avoid --
for now at least.  This all began when my wife went food shopping
around the holiddays and wanted to know if we had something or not
already, and I wasnt home to confirm or not.   So I got to thinking,
if she could call the VoIP box and get one of those press 1 for
spices... press 2 for dry foods, press 3 for canned goods, press 4 for
snacks, press 5 for drinks. and it would access a
database that I already have set up with our groceries already in
there.   Yea - geeky, I know.  :-)

So that was the plan but first I needed to be able to get this thing
set up.  I THINK you're saying I need to purchase another service to
get myself to make calls.   I dont know anyone with a SIP server..
I'd rather keep it all inside our home if possible.  I've got a lot of
old hardware laying around and I do have MODEMs - internal and
external 56k types.   Thanks for the hints.  I'll check into Magic
Jack type stuff and see how it can help.

-M

On Tue, Jan 5, 2010 at 3:19 PM, James A. Shigley j...@answeringserv.com
wrote:
 I can't help you two much with configuration of linux, but as to the call
question. You will need some route for the server to be capable of
sending/receiving calls. There is a couple of ways to do this cheaply.

 Buy a standard telephone modem (usb, pci, or serial). And plug into wall a
jack. This will only allow you one call at a time. But if this is just a
proof of concept that sounds like it will be plenty. Your number would be
whatever the phone jack corresponds to.

 Integrate it with something like skype.

 Buy one of the products similar to a magic jack which will work with
linux.

 Or if you know someone who has a sip server already running. And is
amiable to letting you piggy back off of it. Your  phone number would point
to that person's Server (be it one of the friends that the person is lending
you, or a number that you have forwarded/ported at/to them) with it set to
forward to your asterisk server. from there you could use your dialplan to
do whatever you wanted it to. And for outbound you would send the calls out
thru the friends server via sip or iax.

 James Shigley
 Monroe Telephone Answering Service
 409-981-9213
 Infinity 5.51,UC 4.02.3803, Blink 3.0.104
 Ecreator:2.21, eResponse 1.1.7
 Webportal,WebApps,

 CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT
DEVELOPMENT
 Sent: Tuesday, January 05, 2010 1:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Really Silly Question From Total Newbie

 .

 My main question is: CAN I make call from that box to my cell phone
 using a soft-phone?   If so, how can I do that?   Also, can I use my
 cell phone to call into that box?   I dont know if I have to get a
 phone number, or do I NEED a phone number?   At the moment, I do not
 have any dollars to throw at this project.   Its purely for learning,
 proof of concept sort of thing for myself on my spare time in the
 evenings.  I would simply like to be able to call out and be able to
 call into that box.  Later on down the road maybe I will get into
 setting up an IVR using a database so I can call into that system from
 wherever and get information read back to me.  But, first things
 first  I'd like to know if I am heading down the wrong path here.

Re: [asterisk-users] Really Silly Question From Total Newbie

Steve-

Got an iPhone but no SIP client that I am aware of. I just make
regular calls to other others/receive calls as usual. Nothing fancy.
I was hoping to create the fancy stuff in my home here.

As I got to reading I began to see things like provider, as you've
said here, and unfortunately if that is the only way then I will have
to stop here as I do not have funds to further this little experiment.
I guess I was under the impression that with a lot of configuring and
reading and technical assistance, etc - I could create what I suppose
the VoIP provider is basically doing, trying to avoid yet another
monthly charge...

I checked the link you provided but no pricing info and I started to
search on VoIP Service and found this, amoung hundreds+ others,
http://www.whichvoip.com/Cheap-VoIP-Phone.htm and see monthly charges
as I feared.As I said, I was hoping to create it all in-house (not
have to create an account anywhere or pay fees for fooling around with
a test setup, etc) but it is beginning to appear that I have not
researched enough. Its much bigger an experiment than I had imagined!

Thanks!
-M

On Tue, Jan 5, 2010 at 3:30 PM, Steve Edwards asterisk@sedwards.com wrote:
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

CAN I make call from that box to my cell phone using a soft-phone? If
so, how can I do that?

You need to get an account with a VOIP provider -- someone to accept your
call via the Internet and place a call on the PSTN to call your cell
number -- or any other number.

Also, can I use my cell phone to call into that box? I dont know if I
have to get a phone number, or do I NEED a phone number?

With the same account you can rent a PSTN number. When someone calls that
number, they will call your server over the Internet.

This is assuming you don't have a fancy-dancy smart phone with a SIP
client.

I use www.vitelity.net. I don't know their current pricing, but outbound
calls are less than US$0.015 per minute and renting a PSTN number is
around US$1.50 per month.

--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] SIP Listen Multiple Ports

On Tue, 5 Jan 2010, Vikram Ragukumar wrote:

 If Kamailio is setup to listen on ports 5060 and 9090, port 9090 carries 
 unknown SIP signaling information. Is it possible for Kamailio to dump 
 these unrecognized signaling packets to a user space application which 
 would process and return packets to Kamailio ?

 Would it be better to use libnetfilter_queue() to handle the 
 unrecognized signaling information prior to Kamailio ?

You're asking a blind man to describe an elephant. My knowledge in this 
area is so shallow you would be a fool to dive in based on anything I say 
:)

I would be very surprised if Kamailio can deal with anything other that 
SIP, regardless of what port it is sent on.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Really Silly Question From Total Newbie

Thank you Danny.  I shall investigate that.

On Tue, Jan 5, 2010 at 3:57 PM, Danny Nicholas da...@debsinc.com wrote:
 There are some free-trial and low-cost services out there.  Gizmo comes to
 mind but buyer beware;  look through this site for recommendations and
 warnings.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT
 DEVELOPMENT
 Sent: Tuesday, January 05, 2010 2:48 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Really Silly Question From Total Newbie

 James,

 Thank you for the reply.  I do not have phone service in my home.
 I've been 100% cell since 2003.  I do have an old analog phone - big
 heavy thing...  If I connect it to the wall outlet there is nothing.
 I've tried every outlet in the house.  I didnt expect to find a tone
 as we've never connected phone service here.

 My assumption, however stupid it might be, is that I could set this up
 to make calls but as I began to read more and more I started reading
 about gateways and other costly services that I was hoping to avoid --
 for now at least.  This all began when my wife went food shopping
 around the holiddays and wanted to know if we had something or not
 already, and I wasnt home to confirm or not.   So I got to thinking,
 if she could call the VoIP box and get one of those press 1 for
 spices... press 2 for dry foods, press 3 for canned goods, press 4 for
 snacks, press 5 for drinks. and it would access a
 database that I already have set up with our groceries already in
 there.   Yea - geeky, I know.  :-)

 So that was the plan but first I needed to be able to get this thing
 set up.  I THINK you're saying I need to purchase another service to
 get myself to make calls.   I dont know anyone with a SIP server..
 I'd rather keep it all inside our home if possible.  I've got a lot of
 old hardware laying around and I do have MODEMs - internal and
 external 56k types.   Thanks for the hints.  I'll check into Magic
 Jack type stuff and see how it can help.

 -M


 On Tue, Jan 5, 2010 at 3:19 PM, James A. Shigley j...@answeringserv.com
 wrote:
 I can't help you two much with configuration of linux, but as to the call
 question. You will need some route for the server to be capable of
 sending/receiving calls. There is a couple of ways to do this cheaply.

 Buy a standard telephone modem (usb, pci, or serial). And plug into wall a
 jack. This will only allow you one call at a time. But if this is just a
 proof of concept that sounds like it will be plenty. Your number would be
 whatever the phone jack corresponds to.

 Integrate it with something like skype.

 Buy one of the products similar to a magic jack which will work with
 linux.

 Or if you know someone who has a sip server already running. And is
 amiable to letting you piggy back off of it. Your  phone number would point
 to that person's Server (be it one of the friends that the person is lending
 you, or a number that you have forwarded/ported at/to them) with it set to
 forward to your asterisk server. from there you could use your dialplan to
 do whatever you wanted it to. And for outbound you would send the calls out
 thru the friends server via sip or iax.

 James Shigley
 Monroe Telephone Answering Service
 409-981-9213
 Infinity 5.51,UC 4.02.3803, Blink 3.0.104
 Ecreator:2.21, eResponse 1.1.7
 Webportal,WebApps,

 CONFIDENTIALITY NOTICE: This email, including any attachments, contains
 information which may be confidential or privileged. The information =is
 intended to be for the use of the individual or entity named above. If you
 are not the intended recipient, be aware that any disclosure, copying,
 distribution or use of the contents of this information is prohibited. If
 you have received this email in error, please notify the sender immediately
 by reply to sender only message and destroy all electronic and hard copies
 of the communication, including attachments.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT
 DEVELOPMENT
 Sent: Tuesday, January 05, 2010 1:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Really Silly Question From Total Newbie

 .

 My main question is: CAN I make call from that box to my cell phone
 using a soft-phone?   If so, how can I do that?   Also, can I use my
 cell phone to call into that box?   I dont know if I have to get a
 phone number, or do I NEED a phone number?   At the moment, I do not
 have any dollars to throw at this project.   Its purely for learning,
 proof of concept sort of thing for myself on my spare time in the
 evenings.  I would simply like to be able to call out and be able to
 call into that box.  Later on down the road maybe I will get into
 setting up an IVR using a database so I can call into that

Re: [asterisk-users] Really Silly Question From Total Newbie

On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

 I've got a lot of old hardware laying around and I do have MODEMs - 
 internal and external 56k types.

None of your externals will be of any use and I suspect you will spend 
more time than it is worth trying to get any of your internals working. A 
buck-fifty a month and a penny a minute is pretty darn cheap. You will 
have to be consuming thousands of minutes per month before you will reach 
the break-even point of the cost of the land-line you need to plug into 
the modem.

 Thanks for the hints.  I'll check into Magic Jack type stuff and see how 
 it can help.

MagicJack does not work with Linux or Asterisk unless you plan on spending 
a bunch of time hacking (see google.com) to extract the SIP credentials 
out of your device.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Really Silly Question From Total Newbie

On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.

Really? A buck-fifty a month is going to kill the project? I live in San 
Diego, California where SDGE screws us for thirty cents a kilowatt-hour.

How much is that dual Pentium space heater costing you a month :)

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Canadian call quality issue

2010-01-05 Thread Kyle Kienapfel

Going along the internet between us and canada doesn't add much
distance, but bouncing back and forth between east and west coast
does.

On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:
  hello,

  we have been using a couple of US based
  VoIP providers for outbound calls completed
  within the US, without any issues.

  We recently started making calls to Canada
  and have received a few complaints about
  the call quality.

  Questions :

  - Could this be because of the number of
    intermediate IP hops between us / our
    VoIP provider and the Canadian phone
    companies ?

  - Would choosing a Canadian VoIP provider
    address / resolve this issue ?

  Thank you in advance.

 --

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Re: [asterisk-users] Really Silly Question From Total Newbie

Gotcha on the MODEMs.. thanks.

On Tue, Jan 5, 2010 at 4:12 PM, Steve Edwards asterisk@sedwards.com wrote:
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

I've got a lot of old hardware laying around and I do have MODEMs -
internal and external 56k types.

None of your externals will be of any use and I suspect you will spend
more time than it is worth trying to get any of your internals working. A
buck-fifty a month and a penny a minute is pretty darn cheap. You will
have to be consuming thousands of minutes per month before you will reach
the break-even point of the cost of the land-line you need to plug into
the modem.

Thanks for the hints. I'll check into Magic Jack type stuff and see how
it can help.

MagicJack does not work with Linux or Asterisk unless you plan on spending
a bunch of time hacking (see google.com) to extract the SIP credentials
out of your device.

--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Canadian call quality issue

2010-01-05 Thread jon pounder

Kyle Kienapfel wrote:
 Going along the internet between us and canada doesn't add much
 distance, but bouncing back and forth between east and west coast
 does.
   

I'm not so sure that is the case, what I do know is both Rogers and Shaw 
can never seem to fix complaint issues with voip unless you are using 
their phone service. We just gave up on it and I will not ever spend a 
penny with Rogers as a result since I am convinced they are deliberately 
filtering things so you are locked into their voice services.

Other than that, voip works just fine.

 On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:
   
  hello,

  we have been using a couple of US based
  VoIP providers for outbound calls completed
  within the US, without any issues.

  We recently started making calls to Canada
  and have received a few complaints about
  the call quality.

  Questions :

  - Could this be because of the number of
intermediate IP hops between us / our
VoIP provider and the Canadian phone
companies ?

  - Would choosing a Canadian VoIP provider
address / resolve this issue ?

  Thank you in advance.

 --

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[asterisk-users] send faxes as 3,1 kHz Audio

2010-01-05 Thread achris

Hi,

I have installed Asterisk with iaxmodem to send faxes with Hylafax.

But I have problems to send some faxes because the receiver does not accept 
speech. I must send the faxes as 3,1 kHz Audio

But I do not find a possibility to do this. I need urgent help!

Chris
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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread Jamie A. Stapleton

Could use the free http://www.sipgate.com/one for some testing (assuming that 
Asterisk is connected to the Internet)

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
Sent: Tuesday, January 05, 2010 2:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Really Silly Question From Total Newbie

Hello All -

I've been poking around the past few weeks, trying to familiarize
myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
be complete.   This is my first exposure to all of these technologies.

I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
2400 and the install went well.   I can log in and poke around in
Linux and I even configured the box to be recognized on my windows
network.  However, is there a GUI that I can access to help me set
things up?  I've gotten so far as what looks to me like DOS windows
that I can change various things in the OS...

I do not have any other hardware installed.  No cards and no VoIP
phones.   I havent got to the point where I can make a test call or
anything like that.  I dont know how to tell if Asterisk is up and
running and how I can tweak it, etc.   I've been reading a lot of
different things, and have become a bit confused. I think that in time
it will come to me but I needed to stop and ask because I need to know
if I am on the wrong path for what I'd like to do someday

My main question is: CAN I make call from that box to my cell phone
using a soft-phone?   If so, how can I do that?   Also, can I use my
cell phone to call into that box?   I dont know if I have to get a
phone number, or do I NEED a phone number?   At the moment, I do not
have any dollars to throw at this project.   Its purely for learning,
proof of concept sort of thing for myself on my spare time in the
evenings.  I would simply like to be able to call out and be able to
call into that box.  Later on down the road maybe I will get into
setting up an IVR using a database so I can call into that system from
wherever and get information read back to me.  But, first things
first  I'd like to know if I am heading down the wrong path here.

Sorry for what might seem as really silly questions, but I am not sure
how to proceed.

Thanks in advance for any insight that you folks can provide!

Mike

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Re: [asterisk-users] Dahdi and oslec

  git clone http://git.tzafrir.org.il/git/dahdi-extra.git
  cd dahdi-extra
  make gen-patch

And use the generated dahdi_linux_extra.diff . It includes OSLEC and
some other things. See the Makefile there for more information. The
patch should be applied with -p1 .

This repository includes the extra DAHDI drivers currently included
directly in the Debian package.

Tzafrir,
Thank you very much for this. It's been ages since I had to do this,
and previously I was downloading a recent kernel source and copying
drivers/staging/echo to the dahdi source, then modifying the dahdi
kbuild and adding an echo kbuild. This really isn't an area I am all
that familiar with, but should I assume this patch includes the source
for that recent kernel echo code, and as a result I could apply this to
Jason Parkers srpm for dahdi-linux-2.2.0.2, then rebuild the whole
set to leverage the kmod under CentOS?

Thanks again!
jlc

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Re: [asterisk-users] Really Silly Question From Total Newbie

Really? A buck-fifty a month is going to kill the project? I live in San
Diego, California where SDGE screws us for thirty cents a kilowatt-hour.

Yea. Sort of. I am recently unemployed. Got plenty of time on my
hands now and I am trying to not incur any more costs than I need.

How much is that dual Pentium space heater costing you a month :)

We're not paying much over $50 a month right now in the dead of
winter. Nat Gas is a different storyWe're in Charlotte, NC -
Duke Power. We're quite diligent about our electric use. Right now
nothing is on except the PC I am sending from - not even that server.
I have the temp set to 68. At night it goes to 63. At the height of
summer with AC running nearly 24x7 we'll be suprised if its over $100.
I guess nuclear out here is cheap, I dont know. WATER is a
different story. The water bill sucks.

Actually, I've had that thing on 24x7 in the past and the difference
in electric costs is not much. That doesnt mean I would want to tack
on even more, another subscription that I really dont wish to have
right now... Its just for fun. If need be I have another Dell - a
500SC w/2GB RAM, 1 CPU at 1400mhz.Not sure if its capable.

Thanks for the advice and such..
-Mike

On Tue, Jan 5, 2010 at 4:17 PM, Steve Edwards asterisk@sedwards.com wrote:
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Really Silly Question From Total Newbie

 On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

 wow !  are you serious ?

 you can afford an iPhone but your entire project is
 dead in the water over a couple of bucks.

 I know there is a contradiction somewhere in there.

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Re: [asterisk-users] Really Silly Question From Total Newbie

Jamie - I will check that out!  Thanks!   It is just for testing and
yes, the Asterisk box is connected to the Internet.  Cool.

-M

On Tue, Jan 5, 2010 at 4:39 PM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
 Could use the free http://www.sipgate.com/one for some testing (assuming that 
 Asterisk is connected to the Internet)

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
 Sent: Tuesday, January 05, 2010 2:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Really Silly Question From Total Newbie

 Hello All -

 I've been poking around the past few weeks, trying to familiarize
 myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
 be complete.   This is my first exposure to all of these technologies.

 I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
 2400 and the install went well.   I can log in and poke around in
 Linux and I even configured the box to be recognized on my windows
 network.  However, is there a GUI that I can access to help me set
 things up?  I've gotten so far as what looks to me like DOS windows
 that I can change various things in the OS...

 I do not have any other hardware installed.  No cards and no VoIP
 phones.   I havent got to the point where I can make a test call or
 anything like that.  I dont know how to tell if Asterisk is up and
 running and how I can tweak it, etc.   I've been reading a lot of
 different things, and have become a bit confused. I think that in time
 it will come to me but I needed to stop and ask because I need to know
 if I am on the wrong path for what I'd like to do someday

 My main question is: CAN I make call from that box to my cell phone
 using a soft-phone?   If so, how can I do that?   Also, can I use my
 cell phone to call into that box?   I dont know if I have to get a
 phone number, or do I NEED a phone number?   At the moment, I do not
 have any dollars to throw at this project.   Its purely for learning,
 proof of concept sort of thing for myself on my spare time in the
 evenings.  I would simply like to be able to call out and be able to
 call into that box.  Later on down the road maybe I will get into
 setting up an IVR using a database so I can call into that system from
 wherever and get information read back to me.  But, first things
 first  I'd like to know if I am heading down the wrong path here.

 Sorry for what might seem as really silly questions, but I am not sure
 how to proceed.

 Thanks in advance for any insight that you folks can provide!

 Mike

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Re: [asterisk-users] Canadian call quality issue

 Jon/Kyle,

 thank you for the feedback.

 I checked with someone who manages a much
 higher volume of calls to Canada and he said
 there are some pockets  some providers that
 report issues with call quality. Overall the calls
 sound the same as they do in the US.

--

  On Tue, Jan 5, 2010,  jon pounder   wrote:

 Kyle Kienapfel wrote:
 Going along the internet between us and canada doesn't add much
 distance, but bouncing back and forth between east and west coast
 does.


 I'm not so sure that is the case, what I do know is both Rogers and Shaw
 can never seem to fix complaint issues with voip unless you are using
 their phone service. We just gave up on it and I will not ever spend a
 penny with Rogers as a result since I am convinced they are deliberately
 filtering things so you are locked into their voice services.

 Other than that, voip works just fine.

 On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:

  hello,

  we have been using a couple of US based
  VoIP providers for outbound calls completed
  within the US, without any issues.

  We recently started making calls to Canada
  and have received a few complaints about
  the call quality.

  Questions :

  - Could this be because of the number of
    intermediate IP hops between us / our
    VoIP provider and the Canadian phone
    companies ?

  - Would choosing a Canadian VoIP provider
    address / resolve this issue ?

  Thank you in advance.

 --

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Re: [asterisk-users] Really Silly Question From Total Newbie

Yep.  Its called unemployment.   Got the iPhone a little less than a
year ago.   Someone in India got my job in mid-November.   I got stuck
holding the 2-year contract.Oh well.   Such is life.

Look - I am going to retire from this thread.   Everyone's been a
great help and I know you and others dont know my situation and I am
not one to broadcast it - but when prodded.  there it is.
Yes, several bucks leads to more than several bucks and being
unemployed and living off the wife's income - its not an option.
Hopefully you'll not encounter sucky times, else you'd know..That
couple of bucks a month will never just be a couple of bucks..  :-)


On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
  On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

  wow !  are you serious ?

  you can afford an iPhone but your entire project is
  dead in the water over a couple of bucks.

  I know there is a contradiction somewhere in there.

 --

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Re: [asterisk-users] Really Silly Question From Total Newbie

 my apologies, I do understand.

 sorry.

--

  On Tue, Jan 5, 2010,  UIT DEV  wrote:

 Yep.  Its called unemployment.   Got the iPhone a little less than a
 year ago.   Someone in India got my job in mid-November.   I got stuck
 holding the 2-year contract.    Oh well.   Such is life.

 Look - I am going to retire from this thread.   Everyone's been a
 great help and I know you and others dont know my situation and I am
 not one to broadcast it - but when prodded.  there it is.
 Yes, several bucks leads to more than several bucks and being
 unemployed and living off the wife's income - its not an option.
 Hopefully you'll not encounter sucky times, else you'd know..    That
 couple of bucks a month will never just be a couple of bucks..  :-)


 On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
  On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

  wow !  are you serious ?

  you can afford an iPhone but your entire project is
  dead in the water over a couple of bucks.

  I know there is a contradiction somewhere in there.

 --

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[asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-05 Thread Doug

Hi,

Having problems with getting either RxFax or FaxReceive
to compile.  Running Asterisk 1.4 on CentOS 5.

Does anyone have the free/open source executables
that you could send me?

Thanks for your help!

P. S.: TxFax and FaxSend would also be appreciated.


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Re: [asterisk-users] [asterisk-speech-rec] AGI and embargeability

2010-01-05 Thread Kevin P. Fleming

Quinn Weaver wrote:
 On Tue, Jan 5, 2010 at 12:20 AM, Trevor Benson tben...@a-1networks.com 
 wrote:
 Its called speechbackground.  from asterisk console type 'core show 
 applications speech' (and hit the tab key) these are the speech applications 
 used.  Speechbackground being similar to background.
 
 Thanks, Trevor, Steve, and Alex.  I should have been clearer about
 what I want to do.  I want to play a sound file while listening for
 *either* speech *or* DTMF input.  Either should affect control flow.
 Think for customer service, press or say 2.  Is there any way to do
 this?
 
 My second constraint is that I want to do this within a single AGI
 script.  I'd prefer using AGI commands rather than EXEC'ing dialplan
 applications like Background and speechbackground, but I guess I can
 use apps if I need to.

AGI in Asterisk 1.6.0 and later has SPEECH RECOGNIZE which does
exactly what you want.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-05 Thread Tzafrir Cohen

On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
 Hi,
 
 Having problems with getting either RxFax or FaxReceive
 to compile.  Running Asterisk 1.4 on CentOS 5.

What version of SpanDSP do you use?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Dahdi and oslec

2010-01-05 Thread Tzafrir Cohen

On Tue, Jan 05, 2010 at 09:42:48PM +, Joseph L. Casale wrote:
   git clone http://git.tzafrir.org.il/git/dahdi-extra.git
   cd dahdi-extra
   make gen-patch
 
 And use the generated dahdi_linux_extra.diff . It includes OSLEC and
 some other things. See the Makefile there for more information. The
 patch should be applied with -p1 .
 
 This repository includes the extra DAHDI drivers currently included
 directly in the Debian package.
 
 Tzafrir,
 Thank you very much for this. It's been ages since I had to do this,
 and previously I was downloading a recent kernel source and copying
 drivers/staging/echo to the dahdi source, then modifying the dahdi
 kbuild and adding an echo kbuild. This really isn't an area I am all
 that familiar with, but should I assume this patch includes the source
 for that recent kernel echo code, and as a result I could apply this to
 Jason Parkers srpm for dahdi-linux-2.2.0.2, then rebuild the whole
 set to leverage the kmod under CentOS?

Basically - yes. It's an extra patch to add to your source RPM. Are you
familiar with modifying them?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Dahdi and oslec

On Tuesday 05 January 2010 12:21:15 Joseph L. Casale wrote:
 So this script builds them with the dahdi-tools-libs package requirement, I
 thought the fedora spec built all of these? Any idea?
 
Fedora packages the dahdi-tools* suff, but can't include the kernel modules.  
I did not realize you were using CentOS.  You'll need to change some of the 
definitions at the top of the file to match whatever version of dahdi-tools 
you have installed (if CentOS has them).  If not, the Fedora specs and patches 
are here: http://cvs.fedoraproject.org/viewvc/rpms/dahdi-tools/


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


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Re: [asterisk-users] Dahdi and oslec

Basically - yes. It's an extra patch to add to your source RPM. Are you
familiar with modifying them?

Tzafrir,
Vaguely, I would very graciously take any suggestions you could provide:)
The whole dahdi package routine has change since the last time I used it,
was shortly Jason Parker started providing the dahdi linux/tools.

From what I can tell so far, I can continue to use his user tools unchanged
but I need to apply this patch to the tar file in the 
dahdi-linux-2.2.0.2-1_centos5.src.rpm
and rebuild it, but that , `dahdi-linux` pulls in:

dahdi-firmware
dahdi-firmware-oct6114-064
dahdi-firmware-oct6114-128
dahdi-firmware-tc400m
kmod-dahdi-linux
kmod-dahdi-linux-fwload-
yum-kmod

That of which contain dahdi-firmware and kmod-dahdi-linux-fwload-vpmadt032 
which don't have
srpms available to me.

I'm just unclear on how the patching of the dahdi-linux rpm affects the rest.

Thanks for any guidance!
jlc

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Re: [asterisk-users] Dahdi and oslec

On Tuesday 05 January 2010 17:09:32 Joseph L. Casale wrote:
 From what I can tell so far, I can continue to use his user tools unchanged
 but I need to apply this patch to the tar file in the
  dahdi-linux-2.2.0.2-1_centos5.src.rpm and rebuild it, but that ,
  `dahdi-linux` pulls in
 
atrpms.net also provides packages for RHEL5, if those would work.

http://atrpms.net/dist/el5/

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


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Re: [asterisk-users] Dahdi and oslec

atrpms.net also provides packages for RHEL5, if those would work.

http://atrpms.net/dist/el5/

Just on my way to work on this server now, this would be great! That
way I don't have to work all night:) Does the atrpms ones finally do oslec?

Thanks!
jlc

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread meetmecall

Siax is a pretty good working sip and iax2 softphone for the iPhone.  
Easy to connect to your own Asterisk box

If you have an Android phone (I have HTC Hero with Android 1.5)  ASip  
is a good choice. It is working and and calls using umts are working  
surprisingly well.

Erik


They are both available on the markets fot his phones.
On 5 jan 2010, at 22:03, UIT DEVELOPMENT wrote:

 Steve-

 Got an iPhone but no SIP client that I am aware of.  I just make
 regular calls to other others/receive calls as usual.  Nothing fancy.
 I was hoping to create the fancy stuff in my home here.

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
 I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge...

 I checked the link you provided but no pricing info and I started to
 search on VoIP Service and found this, amoung hundreds+ others,
 http://www.whichvoip.com/Cheap-VoIP-Phone.htm  and see monthly charges
 as I feared.As I said, I was hoping to create it all in-house (not
 have to create an account anywhere or pay fees for fooling around with
 a test setup, etc) but it is beginning to appear that I have not
 researched enough.  Its much bigger an experiment than I had imagined!

 Thanks!
 -M

 On Tue, Jan 5, 2010 at 3:30 PM, Steve Edwards asterisk@sedwards.com 
  wrote:
 On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

 CAN I make call from that box to my cell phone using a soft- 
 phone?  If
 so, how can I do that?

 You need to get an account with a VOIP provider -- someone to  
 accept your
 call via the Internet and place a call on the PSTN to call your cell
 number -- or any other number.

 Also, can I use my cell phone to call into that box?  I dont know  
 if I
 have to get a phone number, or do I NEED a phone number?

 With the same account you can rent a PSTN number. When someone  
 calls that
 number, they will call your server over the Internet.

 This is assuming you don't have a fancy-dancy smart phone with a SIP
 client.

 I use www.vitelity.net. I don't know their current pricing, but  
 outbound
 calls are less than US$0.015 per minute and renting a PSTN number is
 around US$1.50 per month.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice:  
 +1-760-468-3867 PST
 Newline  Fax:  
 +1-760-731-3000

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread Ira

At 12:48 PM 1/5/2010, you wrote:
So that was the plan but first I needed to be able to get this thing
set up.  I THINK you're saying I need to purchase another service to
get myself to make calls.   I dont know anyone with a SIP server..

There are services that will give you free incoming minutes and a 
number. I just got one from www.ipcomms.net which seems to work. The 
number is in nowheresville, NY and I'm in Los Angeles, but cell 
phones have free long distance so I don't care.

Ira 


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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread hin lee

You can practice Asterisk using free SIP phones.   This way you can call from 
extension to extension.


SJ Phone
http://www.sjlabs.com/sjp.html

X Lite
http://www.counterpath.com/x-lite.html




From: UIT DEVELOPMENT uit...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Tue, January 5, 2010 2:04:20 PM
Subject: Re: [asterisk-users] Really Silly Question From Total Newbie

Yep.  Its called unemployment.   Got the iPhone a little less than a
year ago.   Someone in India got my job in mid-November.   I got stuck
holding the 2-year contract.Oh well.   Such is life.

Look - I am going to retire from this thread.   Everyone's been a
great help and I know you and others dont know my situation and I am
not one to broadcast it - but when prodded.  there it is.
Yes, several bucks leads to more than several bucks and being
unemployed and living off the wife's income - its not an option.
Hopefully you'll not encounter sucky times, else you'd know..That
couple of bucks a month will never just be a couple of bucks..  :-)


On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
  On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

  wow !  are you serious ?

  you can afford an iPhone but your entire project is
  dead in the water over a couple of bucks.

  I know there is a contradiction somewhere in there.

 --

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Re: [asterisk-users] Really Silly Question From Total Newbie

Thanks and no problem.  There was no way you would have known.   Thank
you for the info - it really is helpful and I have learned a LOT in
this thread.  This is a great list with a lot of helpful folks on it!

Mike

On Tue, Jan 5, 2010 at 5:16 PM, Max McGraw max.mcg...@gmail.com wrote:
  my apologies, I do understand.

  sorry.

 --

  On Tue, Jan 5, 2010,  UIT DEV  wrote:

 Yep.  Its called unemployment.   Got the iPhone a little less than a
 year ago.   Someone in India got my job in mid-November.   I got stuck
 holding the 2-year contract.    Oh well.   Such is life.

 Look - I am going to retire from this thread.   Everyone's been a
 great help and I know you and others dont know my situation and I am
 not one to broadcast it - but when prodded.  there it is.
 Yes, several bucks leads to more than several bucks and being
 unemployed and living off the wife's income - its not an option.
 Hopefully you'll not encounter sucky times, else you'd know..    That
 couple of bucks a month will never just be a couple of bucks..  :-)


 On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
  On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

  wow !  are you serious ?

  you can afford an iPhone but your entire project is
  dead in the water over a couple of bucks.

  I know there is a contradiction somewhere in there.

 --

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Re: [asterisk-users] Really Silly Question From Total Newbie

No Android phone. But I will read up on this anyhow.  The softphone is
probably all that I need then, and of course a functioning Asterisk
setup.

On Tue, Jan 5, 2010 at 7:29 PM, meetmecall i...@meetmecall.nl wrote:
 Siax is a pretty good working sip and iax2 softphone for the iPhone.
 Easy to connect to your own Asterisk box

 If you have an Android phone (I have HTC Hero with Android 1.5)  ASip
 is a good choice. It is working and and calls using umts are working
 surprisingly well.

 Erik


 They are both available on the markets fot his phones.
 On 5 jan 2010, at 22:03, UIT DEVELOPMENT wrote:

 Steve-

 Got an iPhone but no SIP client that I am aware of.  I just make
 regular calls to other others/receive calls as usual.  Nothing fancy.
 I was hoping to create the fancy stuff in my home here.

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
 I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge...

 I checked the link you provided but no pricing info and I started to
 search on VoIP Service and found this, amoung hundreds+ others,
 http://www.whichvoip.com/Cheap-VoIP-Phone.htm  and see monthly charges
 as I feared.    As I said, I was hoping to create it all in-house (not
 have to create an account anywhere or pay fees for fooling around with
 a test setup, etc) but it is beginning to appear that I have not
 researched enough.  Its much bigger an experiment than I had imagined!

 Thanks!
 -M

 On Tue, Jan 5, 2010 at 3:30 PM, Steve Edwards asterisk@sedwards.com
  wrote:
 On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

 CAN I make call from that box to my cell phone using a soft-
 phone?  If
 so, how can I do that?

 You need to get an account with a VOIP provider -- someone to
 accept your
 call via the Internet and place a call on the PSTN to call your cell
 number -- or any other number.

 Also, can I use my cell phone to call into that box?  I dont know
 if I
 have to get a phone number, or do I NEED a phone number?

 With the same account you can rent a PSTN number. When someone
 calls that
 number, they will call your server over the Internet.

 This is assuming you don't have a fancy-dancy smart phone with a SIP
 client.

 I use www.vitelity.net. I don't know their current pricing, but
 outbound
 calls are less than US$0.015 per minute and renting a PSTN number is
 around US$1.50 per month.

 --
 Thanks in advance,
 -
 Steve Edwards       sedwa...@sedwards.com      Voice:
 +1-760-468-3867 PST
 Newline                                              Fax:
 +1-760-731-3000

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Re: [asterisk-users] Really Silly Question From Total Newbie

Ah, good idea.  :-)   Are you saying that if I got a number that was
in my parents area code then they could be making a local call to my
Asterisk, which is physically a 1000+ miles from them?   Now that is
cool.

On Tue, Jan 5, 2010 at 7:51 PM, Ira i...@extrasensory.com wrote:
 At 12:48 PM 1/5/2010, you wrote:
So that was the plan but first I needed to be able to get this thing
set up.  I THINK you're saying I need to purchase another service to
get myself to make calls.   I dont know anyone with a SIP server..

 There are services that will give you free incoming minutes and a
 number. I just got one from www.ipcomms.net which seems to work. The
 number is in nowheresville, NY and I'm in Los Angeles, but cell
 phones have free long distance so I don't care.

 Ira


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Re: [asterisk-users] Really Silly Question From Total Newbie

Thank you for these.  I will be reading up on these sites shortly.

On Tue, Jan 5, 2010 at 7:59 PM, hin lee hi...@yahoo.com wrote:
 You can practice Asterisk using free SIP phones.   This way you can call
 from extension to extension.

 SJ Phone
 http://www.sjlabs.com/sjp.html

 X Lite
 http://www.counterpath.com/x-lite.html

 
 From: UIT DEVELOPMENT uit...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tue, January 5, 2010 2:04:20 PM
 Subject: Re: [asterisk-users] Really Silly Question From Total Newbie

 Yep.  Its called unemployment.  Got the iPhone a little less than a
 year ago.  Someone in India got my job in mid-November.  I got stuck
 holding the 2-year contract.    Oh well.  Such is life.

 Look - I am going to retire from this thread.  Everyone's been a
 great help and I know you and others dont know my situation and I am
 not one to broadcast it - but when prodded.  there it is.
 Yes, several bucks leads to more than several bucks and being
 unemployed and living off the wife's income - its not an option.
 Hopefully you'll not encounter sucky times, else you'd know..    That
 couple of bucks a month will never just be a couple of bucks..  :-)


 On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
  On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

  wow !  are you serious ?

  you can afford an iPhone but your entire project is
  dead in the water over a couple of bucks.

  I know there is a contradiction somewhere in there.

 --

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Re: [asterisk-users] Really Silly Question From Total Newbie

On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

 Are you saying that if I got a number that was in my parents area code 
 then they could be making a local call to my Asterisk, which is 
 physically a 1000+ miles from them?  Now that is cool.

See http://www.voip-info.org/wiki/view/DID+Service+Providers

Setting up IAX has fewer potholes than SIP. If your Asterisk server 
registers with the provider you can skip all of the firewall and routing 
issues as well.

You can have any number of PSTN numbers ring your Asterisk server. You can 
assign (in your dial plan) custom ring tones to each so you know if it is 
your friends  family number, your wife's business number, your I'm 
looking for a job number, etc.

A lot of the free DIDs are in the middle of nowhere because of the funny 
FCC tariffs that say that the long distance carrier has to pay the rural 
telephone company above market rates for the call. That's how some of 
the cheesy, late-night cable TV chat services work.

You can get DIDs in other countries as well. I have 5 in England so that 
when my wife is home she can call me or each of our kids with a local 
call.

The numbers are registered to my Asterisk server in San Diego. When a call 
comes in, it dials (using Vitelity) the real cell numbers.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Dahdi and oslec

On Tuesday 05 January 2010 17:30:31 Joseph L. Casale wrote:
 Just on my way to work on this server now, this would be great! That
 way I don't have to work all night:) Does the atrpms ones finally do oslec?

I don't use them myself, but I was thinking that the RHEL5 spec files might be 
another place to look for what you need to build with OSLEC included, more 
specifically for CentOS.  I just tried taking a look at ATrpms, but the site 
is having some connection issues at the moment.

How about this -- another CentOS repo:
http://www.zultron.com/2009/03/dahdi-rpms/

Otherwise I'm afraid you'll need to patch and compile.

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


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[asterisk-users] Merlin Legend integration not routing calls back to PSTN.

2010-01-05 Thread Shane Brath

Folks,

I have a Merlin Legend R7 V10.0 with a 2 100D cards.

I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going
to a flip cable to a TE110P card in a Asterisk 1.6.x box.

I have routing setup on the Merlin to send a block of numbers to the
Asterisk.

Currently the PSTN can dial the Asterisk Extensions.
The Asterisk can dial Merlin Extensions.
The Merlin can Dial Asterisk extensions.

But the Asterisk can't dial out to the PSTN?

I have tried everything, and I'm hoping someone else can shed some light on
this. I'm open to ideas.
I've already removed the barrier codes, and disable access code requirements
on Tie and Non-Tie lines, with no effect.
I made sure that the Asterisk is dialing 9XXX when sending the call over
the DAHDI trunk to the Merlin.

Whenever you call from the Asterisk to the Merlin you are redirected to the
Unassigned Extension extension, and dropped to the Operator. I have a
suspicion that this might have something to do with the NetwkService on the
Slot 4 100D card ( out to PSTN ).

Here are some relavant files for comment:

Merlin PRIINFO:
A PRI INFORMATION



A Slot 4 Switch: 5ESS

A Slot 10 Switch: Legend-Ntwk

A System: By line

A BchnlGrp #: Slot: TestTelNum: NtwkServ: Incoming Routing:
A 1 4 CallbyCall By Dial Plan

A Channel ID: 23 22 21 20 19 18 17 16 15 14
A 13 12 11 10 9 8 7 6 5 4
A 3 2 1

A Line PhoneNumber NumberToSend
A 801 NPANXX
A 802 NPANXX
A 803 NPANXX
A 804 NPANXX
A 805 NPANXX
A 806 NPANXX
A 807 NPANXX
A 808 NPANXX
A 809 NPANXX
A 810 NPANXX
A 811 NPANXX
A 812 NPANXX
A 813 NPANXX
A 814 NPANXX
A 815 NPANXX
A 816 NPANXX
A 817 NPANXX
A 818 NPANXX
A 819 NPANXX
A 820 NPANXX
A 821 NPANXX
A 822 NPANXX
A 823 NPANXX

A BchnlGrp #: Slot: TestTelNum: NtwkServ: Incoming Routing:
A 80 10 ElecTandNtwk Route Directly to UDP

A Channel ID: 23 22 21 20 19 18 17 16 15 14
A 13 12 11 10 9 8 7 6 5 4
A 3 2 1

A Line PhoneNumber NumberToSend
A 829 NPANXX
A 830 NPANXX
A 831 NPANXX
A 832 NPANXX
A 833 NPANXX
A 834 NPANXX
A 835 NPANXX
A 836 NPANXX
A PRI INFORMATION


A 837 NPANXX
A 838 NPANXX
A 839 NPANXX
A 840 NPANXX
A 841 NPANXX
A 842 NPANXX
A 843 NPANXX
A 844 NPANXX
A 845 NPANXX
A 846 NPANXX
A 847 NPANXX
A 848 NPANXX
A 849 NPANXX
A 850 NPANXX
A 851 NPANXX

A Network Selection Table

A Entry Number: 0 1 2 3
A Pattern to Match: 101 10***

A Special Service Table

A Entry Number: 0 1 2 3 4 5 6 7
A Pattern to Match: 011 010 01 00 1
A Operator: none OP OP OP/P OP none none none
A Type of Number: I I I N N N N N
A Digits to Delete: 3 0 0 0 0 0 0 0

A Call-By-Call Service Table

A Entry Number: 0 1 2 3 4
A Pattern 0: 0
A Pattern 1: 1
A Pattern 2: 2
A Pattern 3: 3
A Pattern 4: 4
A Pattern 5: 5
A Pattern 6: 6
A Pattern 7: 7
A Pattern 8: 8
A Pattern 9: 9
A Call Type: BOTH BOTH BOTH BOTH BOTH
A NtwkServ: No Service
A DeleteDigits: 0 0 0 0 0

A Entry Number: 5 6 7 8 9
A Call Type: BOTH BOTH BOTH BOTH BOTH
A NtwkServ:
A DeleteDigits: 0 0 0 0 0

A Dial Plan Routing Table

A Entry Number: 0 1 2 3
A NtwkServ: Any service Any service
A PRI INFORMATION


A Expected Digits: 4 4 0 0
A Pattern to Match: 
A Digits to Delete: 0 4 0 0
A Digits to Add: 

A Entry Number: 4 5 6 7
A NtwkServ:
A Expected Digits: 0 0 0 0
A Pattern to Match:
A Digits to Delete: 0 0 0 0
A Digits to Add:

A Entry Number: 8 9 10 11
A NtwkServ:
A Expected Digits: 0 0 0 0
A Pattern to Match:
A Digits to Delete: 0 0 0 0
A Digits to Add:

A Entry Number: 12 13 14 15
A NtwkServ:
A Expected Digits: 0 0 0 0
A Pattern to Match:
A Digits to Delete: 0 0 0 0
A Digits to Add:



Merlin TRUNKINFO

A GENERAL TRUNK INFORMATION



A QCC QCC Extern
A Trk SS/PP RemAccess Pool TlPrfx HldDisc Principal Prty Oper Switch
A 801 4/ 1 No Remote 70 Yes Long 4
A 802 4/ 2 No Remote 70 Yes Long 4
A 803 4/ 3 No Remote 70 Yes Long 4
A 804 4/ 4 No Remote 70 Yes Long 4
A 805 4/ 5 No Remote 70 Yes Long 4
A 806 4/ 6 No Remote 70 Yes Long 4
A 807 4/ 7 No Remote 70 Yes Long 4
A 808 4/ 8 No Remote 70 Yes Long 4
A 809 4/ 9 No Remote 70 Yes Long 4
A 810 4/10 No Remote 70 Yes Long 4
A 811 4/11 No Remote 70 Yes Long 4
A 812 4/12 No Remote 70 Yes Long 4
A 813 4/13 No Remote 70 Yes Long 4
A 814 4/14 No Remote 70 Yes Long 4
A 815 4/15 No Remote 70 Yes Long 4
A 816 4/16 No Remote 70 Yes Long 4
A 817 4/17 No Remote 70 Yes Long 4
A 818 4/18 No Remote 70 Yes Long 4
A 819 4/19 No Remote 70 Yes Long 4
A 820 4/20 No Remote 70 Yes Long 4
A 821 4/21 No Remote 70 Yes Long 4
A 822 4/22 No Remote 70 Yes Long 4
A 823 4/23 No Remote 70 Yes Long 4
A 824 4/24 No Remote Yes Long 4
A 825 5/ 1 No Remote Yes Long 4
A 826 5/ 2 No Remote Yes Long 4
A 827 5/ 3 No Remote Yes Long 4
A 828 5/ 4 No Remote Yes Long 4
A 829 10/ 1 No Remote 890 Yes Long 4
A 830 10/ 2 No Remote 890 Yes Long 4
A 831 10/ 3 No Remote 890 Yes Long 4
A 832 10/ 4 No Remote 890 Yes Long 4
A 833 10/ 5 No

Re: [asterisk-users] Really Silly Question From Total Newbie

Thank you Steve. It is clear that I've only hit the tip of a massive
iceberg with this stuff. Its all very cool, I've got the time so I
might as well make good use of it when I am not out on interviews and
such. It is all such an interesting topic.

On Tue, Jan 5, 2010 at 9:12 PM, Steve Edwards asterisk@sedwards.com wrote:
On Tue, 5 Jan 2010, UIT DEVELOPMENT wrote:

Are you saying that if I got a number that was in my parents area code
then they could be making a local call to my Asterisk, which is
physically a 1000+ miles from them? Now that is cool.

See http://www.voip-info.org/wiki/view/DID+Service+Providers

Setting up IAX has fewer potholes than SIP. If your Asterisk server
registers with the provider you can skip all of the firewall and routing
issues as well.

You can have any number of PSTN numbers ring your Asterisk server. You can
assign (in your dial plan) custom ring tones to each so you know if it is
your friends family number, your wife's business number, your I'm
looking for a job number, etc.

A lot of the free DIDs are in the middle of nowhere because of the funny
FCC tariffs that say that the long distance carrier has to pay the rural
telephone company above market rates for the call. That's how some of
the cheesy, late-night cable TV chat services work.

You can get DIDs in other countries as well. I have 5 in England so that
when my wife is home she can call me or each of our kids with a local
call.

The numbers are registered to my Asterisk server in San Diego. When a call
comes in, it dials (using Vitelity) the real cell numbers.

--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Dahdi and oslec