[asterisk-users] Three-way conference in Asterisk

[Nikhil]

Hi

How to set a threeway conference in asterisk only for VOIP (I am 
using only SIP channel).


Thanks
Nikhil

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Re: [asterisk-users] AMI buffering event output?

[Matt Riddell]

On 1/06/11 11:03 PM, Örn Arnarson wrote:

Hi Matt,

Yes, passing two carriage returns. I login successfully. Here's
example output (with my comments in [])

Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.1
action: login
username: phpagi
secret: supersecretpassword
events: on

Response: Success
Message: Authentication accepted


It seems somewhat impossible that you would be getting different results 
from different hosts.  Are you using the same login?


What if you use the external IP rather than 127.0.0.1

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[asterisk-users] Migration from Mantis to JIRA

[Russell Bryant]

Greetings,

A few weeks ago I posted a message about the upcoming migration from
Mantis to JIRA for issues.asterisk.org [1].  A lot of testing has been
done and all known issues have been resolved.  We have scheduled the
migration for Sunday, June 5th.  The issue tracker will be down most of
the day as the migration takes place.  Once the migration is complete,
the issue tracker will be:

https://issues.asterisk.org/jira/

Mantis will still be available for some time, but will be read-only.  If
you have an account on Mantis, you will be able to log in to JIRA using
the same username.  All of your history will have been migrated.  This
account can also be used on wiki.asterisk.org.

IMPORTANT NOTE: You will have to click the "forgot my password" link to
reset your password before you can log in, though.  It is not possible
to migrate passwords from one to the other as they use a different
hashing algorithm.

For more information about how to use JIRA, see the JIRA user's guide:

http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide

If you run into any problems after the migration has taken place, please
report them in the "JIRA Help" project.  If you would rather report
something via email, email espiceland at digium dot com and me.

Thanks,

[1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html

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www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org

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[asterisk-users] Question about "null routing" calls to DIDs we don't handle

[Jesse Thompson]

Hello, this is Jesse with Webformix.

We are managing an Asterisk installation for residential VOIP service, and
we are having a problem where all inbound calls to DIDs which are assigned
to us by our wholesaler but not yet assigned to a downstream customer get
caught in a routing loop.

** For example: we get assigned DID block 1230-1239 and only 1230-1233 are
given to our customers, then our routing logic sends inbound calls to 1234
back to upstream, which bounce back to us again, tying up all our trunks
before emitting an error.

When we recognized the problem I automatically added ael commands to route
all known unassigned numbers to a friendly error message, but that is a
clumsy approach and the larger issue remains that if we get routed a call
for a number we didn't expect — due to provisioning mistake, premature
porting, or mis-routing at our wholesaler — we still need to avoid a routing
loop.

** For example, we have block 1230-1239, and upstream sends us a call for
4321 for no reason at all.. we're not specifically failing that DID so we
still get a routing loop.

What I would like is to somehow identify all inbound calls (originating from
upstream) that are not terminated inside our network, and give them the
friendly error. Calls originating from our customers should get routed
upstream, but calls originating upstream should not bounce back upstream
without an intervening new Dial command.

** For example: we want...
upstream calls customer @ 1230, rings customer
upstream sends us call for 4321, we play an error
customer calls customer @ 1230, rings customer without bothering upstream
customer calls 4321, call bubbles upstream.
anyone calls customer @ 1233, which has unconditional forwarding set to
4322. Via new Dial command, call should route upstream.

The trouble is that, to our knowledge, all calls from local clients and all
calls we get from upstream have to pass through a context we call "clients"
that routes calls to specific local clients. Local calls can't be sent
upstream until after they've run this gauntlet, and inbound calls can't be
failed until they have been matched against same patterns, but after those
matches it's no longer clear how to separate the remotely sourced calls into
an &Unavailable() bin and send the locally sourced calls upstream.

Here is a simplified version of our configs to give you an idea of the tack
we are presently taking.

Thanks guys! :)

- - Jesse Thompson
Webformix Telephone Services



== sip.conf
[general]
context=clients; default context for all calls

register => skd...@peer.upstreamvoip.com

[upstream]
type=peer
host=peer.upstreamVOIP.com
username=dfjhjkb
secret=redacted
context=clients; context for all inbound calls
call-limit=8

[residential]
type=friend
host=dynamic
context=local

(SIPcust1)[residentia] ; SIP customer name
username=SIPcust1
mailbox=SIPcust1
secret=redacted

; and other sip customers


== extentions.ael
context local
{
  // Does some local cleanup, strips leading 1 off number, _N11, star codes,
fun stuff
  includes { clients; };
}

context clients
{
  custDID1 => Dial(SIP/SIPcust1,35);
  custDID2 => Dial(SIP/SIPcust2,35);
  custDID3 => Dial(SIP/UNCforwardNumber@upstream,120); // inbound calls to
forwarding customers should route back upstream
  unassignedDID => &Unassigned();

  _NXXNXX => Dial(SIP/${EXTEN}@upstream,120); // numbers not handled
here get routed upstream
  // ... which is a problem if those calls originally came *from* upstream.
:(
}

macro Unassigned() // Audio message played for unavailable numbers, not sure
if there's a machine-friendlier error approach or not. :)
{
  PlayBack(ro_sit);
  PlayBack(unassigned);
  PlayBack(5413);

  PlayBack(ro_sit);
  PlayBack(unassigned);
  PlayBack(5413);


  HangUp();
}
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[asterisk-users] Moh on transfer call

[Asistar srl Ivan Bolognani]

Good morning.

I have a problem with the moh

when I receive an external call, answer and transfer telephone call.

when I transfer the user does not feel the MOH.

if you transfer calls between the internal functioning properly moh

In the users.conf i have
mohinterpret=musiconhold

Kan you help

Ivan






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Re: [asterisk-users] SIP Register DOS attack

[Ira]

At 10:56 AM 6/1/2011, you wrote:
Do you have:
sip.conf
[general]
allowguest=no

So because of this I decided to type "sip show channels" into
my Asterisk and got this:

Peer
User/ANR    Call
ID 
Format Hold  Last Message  Expiry 
Peer
216.xxx.69.xxx   (None) 
f2d8db55-0a7edd  (nothing)  No    Rx:
OPTIONS  

216.xxx.69.xxx   (None) 
2ce0b9a5-6de7f4  (nothing)  No    Rx:
OPTIONS  

64.xxx.41.xxx    6314098389  2a482e4b684a59a 
(nothing) 
No 

192.168.233.xxx  (None)      ioh3fna2aw.n4mz 
(nothing)  No    Rx:
REGISTER 

4 active SIP dialogs
I have allowguest=no and all of those IPs are either my providers or
a SIP phone on my network so why would it show  as the
peer?
I'm running Asterisk SVN-trunk-r319759M  if that matters.
Ira



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Re: [asterisk-users] SIP Register DOS attack

[Paul Belanger]

On 11-05-31 06:24 PM, Al lists wrote:

Hi List
Recently i have noticed this attack on couple of servers,
usually a foreign IP starts sending tons of register request without any
answer to authentication,
if you type sip show channels in cli you will see tons of these:
1.2.3.4  (None)  2389603298   00101/1  0x0 (nothing)No
Rx: REGISTER

since there is no authentication in place, asterisk does not see any failed
register attempt, so there wont be anything added to log file as failed
attempt.
thus fail2ban wont see any activity and wont block the IP.
it simply brings down the internet link and the box due to too many sip
channels.


Do you have:

sip.conf
[general]
allowguest=no

--
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Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] please help

[Camilo Echeverry]

Watch out the dialplan sequence
you have 1 then n, n , anf finally a 2.  try changing the last 2 for "n"

if you are only dialing the number

0678922645

You have to remove the leading _ (underscore) and the ending point "." ,
they are used only when dialing regular patterns. in your case wil
be 0678922645XX where  is as many numbers as you want.




On Mon, May 30, 2011 at 11:54 PM, mahesh katta wrote:

> Remove the _ in front of your dialplan,like
> exten => 0678922645,1,--
>
> On Mon, May 30, 2011 at 11:00 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> Hello list
>>
>> i have configured astersik 1.4 with sip i have a question
>>
>> when i put in dial plan.conf
>>
>> exten => _0678922645.,1,Set(CALLERID(number)=520460587)
>>
>> exten => _0678922645
>> .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
>>
>> exten => _0678922645
>> .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
>>
>> exten => _067892264*5*,2,Hangup()
>>
>> i can not call my number but when i delet the last number '5' i can call
>> without any issue
>>
>> i want to put all the number please any hel to solve this issue
>>
>> thanks and regards
>>
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>
>
>
> --
> Best Regards,
>
> Mahesh Katta
> *BUZZ**WORKS* Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
> (E) Mumbai 400069
> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> Web http://www.buzzworks.com
>
>
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-- 
Camilo A. Echeverry J.
301 7553789
-
Y todo lo que hagáis, hacedlo de corazón, como para el Señor y no para los
hombres.

Colonences 3:23
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[asterisk-users] Immediate 180 Response on Invite

[Elliot Murdock]

Hello,

When an INVITE packet is received, Asterisk immediately responds with
a 180 RINGING response before it receives a response from the
provider.  Furthermore, when this occurs, no ring tone is created on
the caller's end.

I've played around with the progressinband setting, but to no avail.

Any idea what is happening here and how to solve it?

Thanks,
Elliot

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[Steve Davies]

On 1 June 2011 15:10, randall  wrote:
> On 06/01/2011 03:55 PM, randall wrote:
>> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
>>> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
 Hi all,

 After running fine for a few months now asterisk seems to hang
 frequently , still functioning but the DAHDI channels seem busy  (users
 report a busy signal when calling or being called)

 A reboot will allow it to run for another day or maybe 2  or 3 till the
 problem occurs again.


 running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
 2.6.32-5-686

 i get the following errors:
 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 
 2

 (happens on all 4 spans)

 and the following in dmesg:
 [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
 [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:    current
 packet[0..2]: 55 55 FC
 [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71


 Channel 0/1, span 1 got hangup, cause 18
>>>
>>> Is this happening in the middle of a call? Or only a while after the
>>> call ended?
>>>
>>
>> the "bad fcs" messages seem to happen random
> there seems to be a relation indeed, have seen them happen randomly
> quite spurious, but they indeed tend to happen a while after the call is
> made.
>>
>> the hangup happens when a call through DAHDI is attempted,
>> (usually after it has been working fine for a while a day or 2)

In my experience, FCS errors are caused by line quality issues, and
usually (not always) are in the telco's equipment. If they are only
happening occasionally, it may be a marginal, but mostly-OK signal on
the wire.

Do you also get occasional poor-quality audio on calls? The issue will
happen more when a call is being setup, or is progressing because
there are more frames being exchanged when a call is in progress.

I have also seen a bad component or dry solder on a voice card cause
this, and even a badly made ISDN cable can be part of the problem. If
none of that helps, I would ask the telco to put a trace on the line.

Hope that helps,
Steve

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[Tzafrir Cohen]

On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote:
> On 06/01/2011 03:55 PM, randall wrote:
> > On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
> >> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
> >>> Hi all,
> >>>
> >>> After running fine for a few months now asterisk seems to hang
> >>> frequently , still functioning but the DAHDI channels seem busy  (users
> >>> report a busy signal when calling or being called)
> >>>
> >>> A reboot will allow it to run for another day or maybe 2  or 3 till the
> >>> problem occurs again.
> >>>
> >>>
> >>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> >>> 2.6.32-5-686
> >>>
> >>> i get the following errors:
> >>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of 
> >>> span 2
> >>>
> >>> (happens on all 4 spans)
> >>>
> >>> and the following in dmesg:
> >>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
> >>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
> >>> packet[0..2]: 55 55 FC
> >>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
> >>>
> >>>
> >>> Channel 0/1, span 1 got hangup, cause 18
> >>
> >> Is this happening in the middle of a call? Or only a while after the
> >> call ended?
> >>
> > 
> > the "bad fcs" messages seem to happen random
> there seems to be a relation indeed, have seen them happen randomly
> quite spurious, but they indeed tend to happen a while after the call is
> made.

A while after a call is made? A while after a call is ended?

Maybe the provider intentionally sets layer 1 down ("to save power")?

That makes sense on PtMP, though I was not aware of this being used on
PtP.

-- 
   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] [solved] Dahdi_genconfig - "Empty configuration -- no spans"

[Sebastian Arcus]

On 01/06/11 16:11, Sebastian Arcus wrote:

Asterisk - 1.8.4.1
Dahdi-linux - 2.4.1.2
Dahdi-tools - 2.4.1
Kernel: 2.6.37.6
Kernel BKL: enabled

I am upgrading Asterisk on this box. It has an OpenVox A400P PCI analog
card with 1 FXO and 1FXS module.

This server has been running just fine for two years with Asterisk 1.6.1.0

I've just upgraded all the OS and installed Asterisk 1.8.4.1. On trying
to configure the A400P I get:

# modprobe dahdi

# modprobe wctdm

# dahdi_genconf
Empty configuration -- no spans
Empty configuration -- no spans

# dahdi_cfg -vvv
DAHDI Tools Version - 2.4.1

DAHDI Version: 2.4.1.2
Echo Canceller(s):
Configuration
==


Channel map:


0 channels to configure.

# dahdi_hardware
driver should be 'wctdm' but is actually 'netjet'



Answering my own quesiton. The answer was staring me in the face right 
on the line above. Thank you Asterisk team for putting in really helpful 
messages in there (no, that was not sarcasm - that message really is 
helpful).


As the line reads (well, implies), blacklisting netjet in 
/etc/modprobe.d/blacklist.conf and restarting did the trick.


Sebastian

> pci::01:09.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F





Where am I going wrong? Why is dahdi_genconf not seeing the card, when
dahdi_hardware does see it? I've tried "modprobe netjet", but makes no
difference.


Thanks for any suggestions,

Sebastian

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[asterisk-users] Dahdi_genconfig - "Empty configuration -- no spans"

[Sebastian Arcus]

Asterisk - 1.8.4.1
Dahdi-linux - 2.4.1.2
Dahdi-tools - 2.4.1
Kernel: 2.6.37.6
Kernel BKL: enabled

I am upgrading Asterisk on this box. It has an OpenVox A400P PCI analog 
card with 1 FXO and 1FXS module.


This server has been running just fine for two years with Asterisk 1.6.1.0

I've just upgraded all the OS and installed Asterisk 1.8.4.1. On trying 
to configure the A400P I get:


# modprobe dahdi

# modprobe wctdm

# dahdi_genconf
Empty configuration -- no spans
Empty configuration -- no spans

# dahdi_cfg -vvv
DAHDI Tools Version - 2.4.1

DAHDI Version: 2.4.1.2
Echo Canceller(s):
Configuration
==


Channel map:


0 channels to configure.

# dahdi_hardware
driver should be 'wctdm' but is actually 'netjet'
pci::01:09.0 wctdm+   e159:0001 Wildcard TDM400P REV E/F



Where am I going wrong? Why is dahdi_genconf not seeing the card, when 
dahdi_hardware does see it? I've tried "modprobe netjet", but makes no 
difference.



Thanks for any suggestions,

Sebastian

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 03:55 PM, randall wrote:
> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
>> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
>>> Hi all,
>>>
>>> After running fine for a few months now asterisk seems to hang
>>> frequently , still functioning but the DAHDI channels seem busy  (users
>>> report a busy signal when calling or being called)
>>>
>>> A reboot will allow it to run for another day or maybe 2  or 3 till the
>>> problem occurs again.
>>>
>>>
>>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
>>> 2.6.32-5-686
>>>
>>> i get the following errors:
>>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
>>>
>>> (happens on all 4 spans)
>>>
>>> and the following in dmesg:
>>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
>>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
>>> packet[0..2]: 55 55 FC
>>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>>>
>>>
>>> Channel 0/1, span 1 got hangup, cause 18
>>
>> Is this happening in the middle of a call? Or only a while after the
>> call ended?
>>
> 
> the "bad fcs" messages seem to happen random
there seems to be a relation indeed, have seen them happen randomly
quite spurious, but they indeed tend to happen a while after the call is
made.
> 
> the hangup happens when a call through DAHDI is attempted,
> (usually after it has been working fine for a while a day or 2)


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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
>> Hi all,
>>
>> After running fine for a few months now asterisk seems to hang
>> frequently , still functioning but the DAHDI channels seem busy  (users
>> report a busy signal when calling or being called)
>>
>> A reboot will allow it to run for another day or maybe 2  or 3 till the
>> problem occurs again.
>>
>>
>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
>> 2.6.32-5-686
>>
>> i get the following errors:
>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
>>
>> (happens on all 4 spans)
>>
>> and the following in dmesg:
>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
>> packet[0..2]: 55 55 FC
>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>>
>>
>> Channel 0/1, span 1 got hangup, cause 18
> 
> Is this happening in the middle of a call? Or only a while after the
> call ended?
> 

the "bad fcs" messages seem to happen random

the hangup happens when a call through DAHDI is attempted,
(usually after it has been working fine for a while a day or 2)

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[Tzafrir Cohen]

On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
> Hi all,
> 
> After running fine for a few months now asterisk seems to hang
> frequently , still functioning but the DAHDI channels seem busy  (users
> report a busy signal when calling or being called)
> 
> A reboot will allow it to run for another day or maybe 2  or 3 till the
> problem occurs again.
> 
> 
> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> 2.6.32-5-686
> 
> i get the following errors:
> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
> 
> (happens on all 4 spans)
> 
> and the following in dmesg:
> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
> packet[0..2]: 55 55 FC
> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
> 
> 
> Channel 0/1, span 1 got hangup, cause 18

Is this happening in the middle of a call? Or only a while after the
call ended?

-- 
   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] asterisk fails when DNS or internet fails

[nhadie ramos]

I have setup a reverse dns for my local subnet and it seems to have resolved
the issue, i was able to make calls even when my asterisk box is not
connected to the net. thanks for all your help!

On Wed, Jun 1, 2011 at 5:57 AM, Hans Witvliet  wrote:

> On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote:
> > As far as I can tell it is trying to do a reverse lookup on the IPs
> configured on the system.  With the internet down, does the command "host
> 10.10.10.1" (or whatever IPs you have on the system) take a while to come
> back?  Unless you can do a reverse lookup of all the IPs on the system don't
> expect Asterisk to be able to.   If your /etc/hosts is set up correct, you
> should be able to look up any IP configured on any interface on the system
> without delay.
> >
> > I'm sure there are other places Asterisk tries to do DNS lookups, but the
> above info has solved this issue for me in the past.
> >
>
> I'm not sure if that's all is true.
> Sure, if you add a line in /etc/hosts, that works for most applications,
> as not all commands follow /etc/resolv.conf
>
> i just tried, adding a line to /etc/hosts.
> ping hostname works, but host hostname fails, just as host ip-address.
> So even when you only put ip-addresses (brrr) into your config files,
> the reversed-lookup will still spoil the party.
>
>
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Re: [asterisk-users] Free CNAM

[Michael R. Wally]

I've been toying around with the idea of starting some kind of 'Open 
CNAM' project to destroy the current money hustle BS that dominates this 
industry.  The ever-growing FreeCNAM database may be a good starting 
point for such a project.


I would also like to use Bitcoin (BTC) as the micropayment solution for 
user-requested updates.  Some nominal fee.


If anyone wants to get involved, contact me.



On 06/01/2011 07:51 AM, Skyler wrote:

Hi,

  The junk in CNAM databases like "FLORIDA", "ONTARIO" etc. is IMO the
carrier's way to isolate their users and another excuse to charge more money
for 'the better plan'. In the end, it's the carrier that inputs the info so
if it shows "FLORIDA" with one database I can't see how any other database
would be different as the carrier is the only one that controls the outbound
CID info. Calling me from POTS to snatch the CID will result in the same.

...unless there were a user friendly CNAM service, where info could be
updated by the end-user and queried freely by voip providers. I would update
my cellular numbers for sure and know at least a dozen people that would do
the same. Everyone is going VoIP so why not?

  Talking about 'where's the money or angle'... here is one, vanity. Charge
$1/yr to a user per DID, if I don't renew then delete it and re-query the
original carrier.


S.



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Re: [asterisk-users] Free CNAM

[Skyler]

Hi,

 The junk in CNAM databases like "FLORIDA", "ONTARIO" etc. is IMO the
carrier's way to isolate their users and another excuse to charge more money
for 'the better plan'. In the end, it's the carrier that inputs the info so
if it shows "FLORIDA" with one database I can't see how any other database
would be different as the carrier is the only one that controls the outbound
CID info. Calling me from POTS to snatch the CID will result in the same.

...unless there were a user friendly CNAM service, where info could be
updated by the end-user and queried freely by voip providers. I would update
my cellular numbers for sure and know at least a dozen people that would do
the same. Everyone is going VoIP so why not?

 Talking about 'where's the money or angle'... here is one, vanity. Charge
$1/yr to a user per DID, if I don't renew then delete it and re-query the
original carrier.


S.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R.
Wally
Sent: Sunday, May 29, 2011 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Free CNAM

The system uses real Telco CNAM Dips.  Any generic names you get are 
from the subscriber's carrier itself.  We can only provide what we 
ourselves get.
> I tried it, but it returns the same kind of junk that some of the
databases
> do.  For example, on a Florida number, it just says "FLORIDA" instead of
> the proper name (some of the CNAM databases have the right name).
>

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-
No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1375 / Virus Database: 1509/3666 - Release Date: 05/28/11


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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[Karsten Wemheuer]

Hi randall,

Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
> > > i get the following errors:
> > > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
> > > of span 2
> > >
> > >
> > > Your telco provider has crc on or off , that is not matching with
> > > your server cross check with them.
> > > and this problem solve 4 problems
> > >
> > >
> > thanks for the reply,
> > 
> > what is crc (same as crc-4?)
> > and where can i set this?
> > 
> > same crc or crc4
> > 
> > --
> adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
> at all

As I can see from Your first post, You are using BRI in
point-to-multipoint mode. On BRI lines there is nothing like CRC/CRC4
and that is the reason, why the config is not loading any more.

On a PTMP line there may be some CRC-errors from time to time, when the
provider shuts down the line, which is normal in some countries. But
this has nothing to do with Your initial problem.

Unfortunately I don't know a solution for Your problem. It may be a
hardware issue.

HTH,

Karsten



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Re: [asterisk-users] AMI buffering event output?

[Örn Arnarson]

Hi Alex,

In the case of php, there's just an open socket. I'm using that php
script on multiple other hosts with no buffering.

The telnet client on the server is just "telnet", and I've tested
telnetting to AMI on another server to make sure that the client isn't
buffering with no problem. The problem only presents itself when
telnetting to this particular AMI.

Regards,
Örn

On Wed, Jun 1, 2011 at 11:02 AM, Alex Balashov
 wrote:
> Are you using the same telnet client in both cases?
>
> On 06/01/2011 06:54 AM, Örn Arnarson wrote:
>
>> No, because the same is happening with telnet. If I telnet to AMI, I
>> observe exactly the same behavior. Otherwise I would put it down to
>> PHP.
>>
>> On Tue, May 31, 2011 at 5:41 PM, Alex Balashov
>>   wrote:
>>>
>>> On 05/31/2011 01:38 PM, Örn Arnarson wrote:

 Hi,

 I'm seeing weird behavior with AMI where no events are output until
 some input is detected (can be an empty line), at which time all the
 buffered output is spewed out at once.

 I am maintaining multiple Asterisk installations, and with one
 installation I have run into a weird buffering problem with AMI.
 The version is 1.6.1.11 in this particular case, which I am running at
 multiple locations, all without this problem. Additionally I have
 tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent
 across these versions.

 manager.conf is identical across all these installations.

 The problem presents with a php script that opens a socket directly to
 AMI, a telnet client from the local machine to the AMI, but not when I
 telnet from the machine to a remote machine running AMI.

 Does anyone have any input as to what I can try?

 Best regards,
 Örn
>>>
>>> This is because of some blocking and/or read-write order issue in your
>>> PHP
>>> script.
>>>
>>> PHP is a web language, not for standalone scripts doing network I/O.
>>>
>>> --
>>> Alex Balashov - Principal
>>> Evariste Systems LLC
>>> 260 Peachtree Street NW
>>> Suite 2200
>>> Atlanta, GA 30303
>>> Tel: +1-678-954-0670
>>> Fax: +1-404-961-1892
>>> Web: http://www.evaristesys.com/
>>>
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>>
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>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
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Re: [asterisk-users] AMI buffering event output?

[Örn Arnarson]

Hi Matt,

Yes, passing two carriage returns. I login successfully. Here's
example output (with my comments in [])

Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.1
action: login
username: phpagi
secret: supersecretpassword
events: on

Response: Success
Message: Authentication accepted

[No output after this point until I pass carriage return]

[Now I press carriage return and two events appear, along with an
error message for missing action]

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/2170
PeerStatus: Unregistered
Cause: Expired

Event: ExtensionStatus
Privilege: call,all
Exten: 2170
Context: default
Hint: SIP/2170
Status: 4

Response: Error
Message: Missing action in request


On Tue, May 31, 2011 at 10:56 PM, Matt Riddell  wrote:
> On 1/06/11 5:38 AM, Örn Arnarson wrote:
>>
>> The problem presents with a php script that opens a socket directly to
>> AMI, a telnet client from the local machine to the AMI, but not when I
>> telnet from the machine to a remote machine running AMI.
>
> Are you 100% sure it's happening when you use telnet on the local machine?
>
> Are you passing two carriage returns after logging in?
>
> --
> Cheers,
>
> Matt Riddell
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/exchange.php (Full ITSP Solution)
> http://www.venturevoip.com/cc.php (Call Centre Solutions)
>
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Re: [asterisk-users] AMI buffering event output?

[Alex Balashov]

Are you using the same telnet client in both cases?

On 06/01/2011 06:54 AM, Örn Arnarson wrote:


No, because the same is happening with telnet. If I telnet to AMI, I
observe exactly the same behavior. Otherwise I would put it down to
PHP.

On Tue, May 31, 2011 at 5:41 PM, Alex Balashov
  wrote:

On 05/31/2011 01:38 PM, Örn Arnarson wrote:


Hi,

I'm seeing weird behavior with AMI where no events are output until
some input is detected (can be an empty line), at which time all the
buffered output is spewed out at once.

I am maintaining multiple Asterisk installations, and with one
installation I have run into a weird buffering problem with AMI.
The version is 1.6.1.11 in this particular case, which I am running at
multiple locations, all without this problem. Additionally I have
tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent
across these versions.

manager.conf is identical across all these installations.

The problem presents with a php script that opens a socket directly to
AMI, a telnet client from the local machine to the AMI, but not when I
telnet from the machine to a remote machine running AMI.

Does anyone have any input as to what I can try?

Best regards,
Örn


This is because of some blocking and/or read-write order issue in your PHP
script.

PHP is a web language, not for standalone scripts doing network I/O.

--
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Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] AMI buffering event output?

[Örn Arnarson]

No, because the same is happening with telnet. If I telnet to AMI, I
observe exactly the same behavior. Otherwise I would put it down to
PHP.

On Tue, May 31, 2011 at 5:41 PM, Alex Balashov
 wrote:
> On 05/31/2011 01:38 PM, Örn Arnarson wrote:
>>
>> Hi,
>>
>> I'm seeing weird behavior with AMI where no events are output until
>> some input is detected (can be an empty line), at which time all the
>> buffered output is spewed out at once.
>>
>> I am maintaining multiple Asterisk installations, and with one
>> installation I have run into a weird buffering problem with AMI.
>> The version is 1.6.1.11 in this particular case, which I am running at
>> multiple locations, all without this problem. Additionally I have
>> tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent
>> across these versions.
>>
>> manager.conf is identical across all these installations.
>>
>> The problem presents with a php script that opens a socket directly to
>> AMI, a telnet client from the local machine to the AMI, but not when I
>> telnet from the machine to a remote machine running AMI.
>>
>> Does anyone have any input as to what I can try?
>>
>> Best regards,
>> Örn
>
> This is because of some blocking and/or read-write order issue in your PHP
> script.
>
> PHP is a web language, not for standalone scripts doing network I/O.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
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Re: [asterisk-users] To know if the ISDN PRI E1 is UP?

[James zhu]

hello:
Make sure the setting is correct and no Irq LOCKED. 

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 




> Date: Tue, 31 May 2011 05:32:53 -0700
> From: bilmar...@yahoo.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] To know if the ISDN PRI E1 is UP?
> 
> Hi All;
> 
> This is the output of the pri show status, so I appreciate if to know if that 
> means the E1 is UP? What does it means that the status us (Status: In Alarm, 
> Down, Active)? What in the below result give an indication that it is UP?
> 
> CC*CLI> pri show span 1
> Primary D-channel: 16
> Status: In Alarm, Down, Active
> Switchtype: EuroISDN
> Type: Network
> Overlap Dial: 0
> Logical Channel Mapping: 0
> Timer and counter settings:
>   N200: 3
>   N202: 3
>   K: 7
>   T200: 1000
>   T202: 1
>   T203: 1
>   T303: 4000
>   T305: 3
>   T308: 4000
>   T309: 6000
>   T313: 4000
>   T-HOLD: 4000
>   T-RETRIEVE: 4000
>   T-RESPONSE: 4000
> Overlap Recv: No
> 
> Regards
> Bilal
> 
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[asterisk-users] DBdeltree: Error deleting key from database

[Satish Barot]

Hi Everybody,

Don't know why this DBdeltree error appears on Asterisk CLI.Good part is, it
does remove family entry from AstDB.

Sample Dialplan

exten => 1212,1,Noop()
same => n,Set(TEST=1234)
same => n,Set(DB(${TEST}/TESTSTART)=${STRFTIME(${EPOCH},,%Y-%m-%d
%H:%M:%S)})
same => n,DBdeltree(${TEST})
same => n,Hangup()

Asterisk CLI output

[Jun  1 14:30:39]   == Using SIP RTP CoS mark 5
[Jun  1 14:30:39] -- Executing [1212@incoming-did:1]
NoOp("SIP/29-002f", "") in new stack
[Jun  1 14:30:39] -- Executing [1212@incoming-did:2]
Set("SIP/29-002f", "TEST=1234") in new stack
[Jun  1 14:30:39] -- Executing [1212@incoming-did:3]
Set("SIP/29-002f", "DB(1234/TESTSTART)=2011-06-01 14:30:39") in

new stack
[Jun  1 14:30:39] -- Executing [1212@incoming-did:4]
DBdeltree("SIP/29-002f", "1234") in new stack
[Jun  1 14:30:39] -- DBdeltree: family=1234
[Jun  1 14:30:39] -- DBdeltree: Error deleting key from database.
[Jun  1 14:30:39] -- Executing [1212@incoming-did:5]
Hangup("SIP/29-002f", "") in new stack
[Jun  1 14:30:39]   == Spawn extension (incoming-did, 1212, 5) exited
non-zero on 'SIP/29-002f'


Any idea?

TYIA

[SATISH]
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 10:07 AM, mahesh katta wrote:
> 
> 
> On Wed, Jun 1, 2011 at 1:30 PM, randall  > wrote:
> 
> 
> > > i get the following errors:
> > > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
> D-channel
> > > of span 2
> > >
> > >
> > > Your telco provider has crc on or off , that is not
> matching with
> > > your server cross check with them.
> > > and this problem solve 4 problems
> > >
> > >
> > thanks for the reply,
> >
> > what is crc (same as crc-4?)
> > and where can i set this?
> >
> > same crc or crc4
> >
> > --
> adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
> at all
> 
>span=1,1,0,ccs,ami,crc4
crc4 has the same effect

> 
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> 
> 
> 
> -- 
> Best Regards,
> 
> Mahesh Katta
> *BUZZ**WORKS*Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
> (E) Mumbai 400069
> GSM+91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> Webhttp://www.buzzworks.com
> 
> 
> 
> 
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[mahesh katta]

On Wed, Jun 1, 2011 at 1:30 PM, randall  wrote:

>
> > > i get the following errors:
> > > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
> D-channel
> > > of span 2
> > >
> > >
> > > Your telco provider has crc on or off , that is not matching
> with
> > > your server cross check with them.
> > > and this problem solve 4 problems
> > >
> > >
> > thanks for the reply,
> >
> > what is crc (same as crc-4?)
> > and where can i set this?
> >
> > same crc or crc4
> >
> > --
> adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
> at all
>
>span=1,1,0,ccs,ami,crc4

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>



-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

> > i get the following errors:
> > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
> > of span 2
> >
> >
> > Your telco provider has crc on or off , that is not matching with
> > your server cross check with them.
> > and this problem solve 4 problems
> >
> >
> thanks for the reply,
> 
> what is crc (same as crc-4?)
> and where can i set this?
> 
> same crc or crc4
> 
> --
adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
at all

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[mahesh katta]

On Wed, Jun 1, 2011 at 1:07 PM, randall  wrote:

> On 06/01/2011 09:04 AM, mahesh katta wrote:
> >
> >
> > On Wed, Jun 1, 2011 at 11:36 AM, randall  > > wrote:
> >
> > Hi all,
> >
> > After running fine for a few months now asterisk seems to hang
> > frequently , still functioning but the DAHDI channels seem busy
>  (users
> > report a busy signal when calling or being called)
> >
> > A reboot will allow it to run for another day or maybe 2  or 3 till
> the
> > problem occurs again.
> >
> >
> > running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> > 2.6.32-5-686
> >
> > i get the following errors:
> > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
> > of span 2
> >
> >
> > Your telco provider has crc on or off , that is not matching with
> > your server cross check with them.
> > and this problem solve 4 problems
> >
> >
> thanks for the reply,
>
> what is crc (same as crc-4?)
> and where can i set this?
>
> same crc or crc4

> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 09:04 AM, mahesh katta wrote:
> 
> 
> On Wed, Jun 1, 2011 at 11:36 AM, randall  > wrote:
> 
> Hi all,
> 
> After running fine for a few months now asterisk seems to hang
> frequently , still functioning but the DAHDI channels seem busy  (users
> report a busy signal when calling or being called)
> 
> A reboot will allow it to run for another day or maybe 2  or 3 till the
> problem occurs again.
> 
> 
> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> 2.6.32-5-686
> 
> i get the following errors:
> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
> of span 2
>  
> 
> Your telco provider has crc on or off , that is not matching with
> your server cross check with them.
> and this problem solve 4 problems
>  
> 
thanks for the reply,

what is crc (same as crc-4?)
and where can i set this?

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[mahesh katta]

On Wed, Jun 1, 2011 at 11:36 AM, randall  wrote:

> Hi all,
>
> After running fine for a few months now asterisk seems to hang
> frequently , still functioning but the DAHDI channels seem busy  (users
> report a busy signal when calling or being called)
>
> A reboot will allow it to run for another day or maybe 2  or 3 till the
> problem occurs again.
>
>
> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> 2.6.32-5-686
>
> i get the following errors:
> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span
> 2
>
>
Your telco provider has crc on or off , that is not matching with your
server cross check with them.
and this problem solve 4 problems


> (happens on all 4 spans)
>
> and the following in dmesg:
> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
> packet[0..2]: 55 55 FC
> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>
>
> Channel 0/1, span 1 got hangup, cause 18
>
> According to my searching on the web the last two weeks i understand
> that the first mentioned error can indicate a wrong configuration
> setting or buggy hardware.
>
> Have tried many options but the dahdi config as copied over from the
> same machine (with elastix 1.4) seems the only one that seem to work
>
> anybody has a clue?
> much appreciated
>
> some gory details below,
>
> HARDWARE:
> D201GLY2 motherboard
> usb:001/002  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
>
>
> DAHDI CONFIG:
>
> dahdi-channels.conf
>
> ; Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: BRI_TE" (MASTER)
> group=0,11
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> channel => 1-2
> context = default
> group = 63
>
> ; Span 2: XBUS-00/XPD-01 "Xorcom XPD #00/01: BRI_TE"
> group=0,12
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> channel => 4-5
> context = default
> group = 63
>
> ; Span 3: XBUS-00/XPD-02 "Xorcom XPD #00/02: BRI_TE"
> group=0,13
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> channel => 7-8
> context = default
> group = 63
>
> ; Span 4: XBUS-00/XPD-03 "Xorcom XPD #00/03: BRI_TE"
> group=0,14
> context=from-pstn
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> channel => 10-11
> context = default
> group = 63
>
>
> chan_dahdi.conf
>
> [channels]
> context=from-pstn
> signalling=fxs_ks
> rxwink=300  ; Atlas seems to use long (250ms) winks
> pridialplan=local
> prilocaldialplan=local
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=no
> faxdetect=incoming
> echotraining=800
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> ;Uncomment these lines if you have problems with the disconection of
> your analog lines
> ;busydetect=yes
> ;busycount=3
> immediate=no
>
>
>
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-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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