Re: [asterisk-users] BLF and call-limit in 1.8

2012-12-07 Thread Matthew Jordan 
On 12/07/2012 06:16 AM, Pan B. Christensen wrote:
> But Asterisk doesn't send ANY notification regarding change of state
> (like ringing) unless call-limit is 1. In my opinion, ringing status
> (and probably several others) shouldn't take into consideration at what
> point a device is considered as busy.
> This worked fine in 1.4 (we skipped 1.6) and seems broken in 1.8.
> Busy-level is 1 in both configurations.
>  
> After some more investigation, I may have found a clue. When I type sip
> show peer , the following values are reported:
> With DB value 4 or 3 or 2 or 0:
>   Call limit   : 0
> With DB value 1:
>   Call limit   : 2147483647
> This doesn't look correct to me...
>  

When you enable call-limit globally for all peers, it sets the
call-limit for all peers to INT_MAX.  Hence why each device can accept
quite a few calls.

The setting you're actually toggling is 'callcounter'.  When its enabled
(boolean true), it sets call-limit to INT_MAX. If not present, then the
call-limit value is used.  This is why every setting but '1' set the
call-limit to 0.

Since you're using Realtime, there me a number of issues at play. Do you
have this problem with a peer defined in sip.conf?

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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[asterisk-users] Adding custom HTTP headers to Asterisk

2012-12-07 Thread Paul Belanger 
Anybody using Apache to proxy HTTP traffic to Asterisk HTTP? I got a
request from a developer to add some CORS headers[1], for an
application we are writing, and wanted to see if anybody else has had
success.

[1] http://enable-cors.org/

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [asterisk-users] Audio feedback - where to troubleshoot?

2012-12-07 Thread Matthew J. Roth 
Justin Sherrill wrote:

> We occasionally get a sort of feedback/echo noise on our phones
> here.  (Polycom IP550 / Asterisk 1.8).  It lasts for about a second,
> and it's described by users as 'jingle bells'.

That's very appropriate for the season.  I just hope they're not
Todash Chimes.  ; )

> It happens when people are using the speakerphone, especially when
> it's on both ends.  It happens on internal calls, so it's not
> something from the PSTN.
> 
> My theory is that the very sensitive speakerphone microphones in
> those models are picking up small noises, and then picking up their
> own playback of the noise, which leads to the feedback sound.  It
> repeats, which is why it sounds like an echo.

Your theory sounds correct to me.  You've described acoustic echo and
that is the most prevalent type for internal VoIP calls.

> I think turning rxgain/txgain down may make a difference, but I
> haven't tried it yet.  Has anyone else experienced something
> similar?  

I believe rxgain and txgain only apply to DAHDI channels.  The first
thing I would try is reducing the speakerphone volume settings on the
Polycom IP550s.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

2012-12-07 Thread Jeff LaCoursiere 


You can also (in mysql) tag a column as "unique":

alter table blah add unique(column_name);

This doesn't add a key AFAIK unless you ask it to be a key:

alter table blah add unique key(column_name);

If you will never refer to a row using the auto-increment column, why 
have it?


No problem having a table with no keys whatsoever, if you never plan to 
pull up records without dumping the whole table.  But agree, that would 
be odd in this instance, since presumably you would at least want to 
have key on the source or destination columns.


j

On 12/07/2012 12:52 PM, Don Kelly wrote:


This thread confuses me. I've not worked with the Asterisk MySQL CDR, 
but have worked with SQL for years.


Every table should have a primary key. Is no column identified as a PK?

If there is a PK, you will not be able to designate another column as PK.

If there is a PK, you don't need to worry about Asterisk duplicating 
keys in that column---the database will not permit non-unique PKs.


If (and this would be really weird) the table is created with no 
primary key, it shouldn't be a problem to add the primary key 
constraint to a column IF there are no nulls or duplicates existing.


--Don

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Leandro Dardini

*Sent:* Thursday, December 06, 2012 12:50 PM
*To:* rwhee...@artifact-software.com; Asterisk Users Mailing List - 
Non-Commercial Discussion
*Subject:* Re: [asterisk-users] CDR - Freepbx - Safe to add primary 
key to table ?


The reason I add a new column autoincrement is due to the fact I trust 
more mysql about uniquness than asterisk.


Leandro

I am typing from my mobile phone...

Il giorno 06/dic/2012 19:11, "Ron Wheeler" 
> ha scritto:


It seems like a safe thing to do.
You could also ask about the impact of making an existing column a  
primary key, in a MySQL forum.


Leandro's solution seems to be a good one as well and does guarantee 
uniqueness.




Ron

On 06/12/2012 12:25 PM, Leandro Dardini wrote:

Yes, go for it. However I have added another autoincrement column
and created the primary key on it. On the other columns I need to
search I have created just an index.

Leandro

2012/12/6 Olivier mailto:oza_4...@yahoo.fr>>

Hello,

I need to develop an application that will query (mostly reading)
an existing MySQL CDR database.
This database (named asteriskcdrdb) was created during Freepbx
2.10 install on my asterisk 1.8 setup.
This database has a single CDR table which is filled by Asterisk.

The tools I'm planning to use require this table to include a
Primary Key.
Is it safe to Alter this table telling it to use UniqueID column
as a Primary Key ?

(Sure, I'll test this on a database copy but I'm not confident my
tests will cover everything)

Regards

--



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Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

2012-12-07 Thread Don Kelly 
This thread confuses me. I've not worked with the Asterisk MySQL CDR, but
have worked with SQL for years.

 

Every table should have a primary key. Is no column identified as a PK?

 

If there is a PK, you will not be able to designate another column as PK.

 

If there is a PK, you don't need to worry about Asterisk duplicating keys in
that column-the database will not permit non-unique PKs.

 

If (and this would be really weird) the table is created with no primary
key, it shouldn't be a problem to add the primary key constraint to a column
IF there are no nulls or duplicates existing.

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Thursday, December 06, 2012 12:50 PM
To: rwhee...@artifact-software.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to
table ?

 

The reason I add a new column autoincrement is due to the fact I trust more
mysql about uniquness than asterisk. 

Leandro

I am typing from my mobile phone...

Il giorno 06/dic/2012 19:11, "Ron Wheeler" 
ha scritto:

It seems like a safe thing to do.
You could also ask about the impact of making an existing column a  primary
key, in a MySQL forum.

Leandro's solution seems to be a good one as well and does guarantee
uniqueness.



Ron

On 06/12/2012 12:25 PM, Leandro Dardini wrote:

Yes, go for it. However I have added another autoincrement column and
created the primary key on it. On the other columns I need to search I have
created just an index. 

 

Leandro

2012/12/6 Olivier 

Hello,

I need to develop an application that will query (mostly reading) an
existing MySQL CDR database.
This database (named asteriskcdrdb) was created during Freepbx 2.10 install
on my asterisk 1.8 setup.
This database has a single CDR table which is filled by Asterisk.

The tools I'm planning to use require this table to include a Primary Key.
Is it safe to Alter this table telling it to use UniqueID column as a
Primary Key ?

(Sure, I'll test this on a database copy but I'm not confident my tests will
cover everything)

Regards

--



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Re: [asterisk-users] PRI can receive calls but cannot dial out

2012-12-07 Thread Alex Kauffmann 

On 12/7/2012 6:23 AM, Vieri wrote:




Am 05.12.2012 08:48, schrieb Vieri:

Hi,

I'm trying to call out from a SIP extension to an

outbound destination via a PRI E1 (Digium B410P).


Please take a look at the PRI debug below.



# cat /etc/dahdi/system.conf

# Digium Wildcard TDM400P REV I (WCTDM/4)
fxsks=1
echocanceller=oslec,1
fxsks=2
echocanceller=oslec,2
fxsks=3
echocanceller=oslec,3
fxsks=4
echocanceller=oslec,4

# Digium Wildcard TDM2400P (WCTDM/0)
fxsks=5
echocanceller=oslec,5
fxsks=6
echocanceller=oslec,6
fxsks=7
echocanceller=oslec,7
fxsks=8
echocanceller=oslec,8
fxsks=9
echocanceller=oslec,9
fxsks=10
echocanceller=oslec,10
fxsks=11
echocanceller=oslec,11
fxsks=12
echocanceller=oslec,12

# Digium Wildcard B410P (B4/0/1)
span=3,1,0,CCS,AMI
bchan=29-30
hardhdlc=31
echocanceller=oslec,29-30

# Digium Wildcard B410P (B4/0/2)
span=4,2,0,CCS,AMI
bchan=32-33
hardhdlc=34
echocanceller=oslec,32-33

# Digium Wildcard B410P (B4/0/3)
span=5,3,0,CCS,AMI
bchan=35-36
hardhdlc=37
echocanceller=oslec,35-36

# Digium Wildcard B410P (B4/0/4)
span=6,4,0,CCS,AMI
bchan=38-39
hardhdlc=40
echocanceller=oslec,38-39



# lsmod | grep wcb4xxp
wcb4xxp

   66250  12

dahdi

169899  65
dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm




# cat chan_dahdi.conf

[trunkgroups]

[channels]
transfer = yes
usecallerid = yes
cidsignalling = dtmf
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
canpark = yes
cancallforward = yes
callreturn = yes
callprogress = no
overlapdial = yes
echocancel = yes
facilityenable = yes
immediate = no
busydetect = no

; Digium Wildcard TDM400P REV I (WCTDM/4)
signalling = fxs_ks
txgain = 1.0
rxgain = 14.0
group = 3
context = incoming-dahdi-3
faxdetect = incoming
channel => 1,2,3,4

; Digium Wildcard TDM2400P (WCTDM/0)
group = 4
context = incoming-dahdi-4
faxdetect = incoming
channel => 5,6,7,8,9,10,11,12

; Digium Wildcard B410P (B4/0/1)
signalling = bri_cpe
switchtype = euroisdn
rxgain = 2.0
group = 2
context = incoming-dahdi-2
faxdetect = incoming
channel => 29-30

; Digium Wildcard B410P (B4/0/2)
channel => 32-33

; Digium Wildcard B410P (B4/0/3)
channel => 35-36

; Digium Wildcard B410P (B4/0/4)
channel => 38-39

---

# asterisk -rx "dahdi show status"
Description


   Alarms  IRQbpviol CRC
Fra Codi Options  LBO

Wildcard TDM400P REV I Board 5

  OK  0
 0  0
CAS Unk   0 db
(CSU)/0-133 feet (DSX-1)

Wildcard TDM2400P

   OK
 0  0
0  CAS Unk
0 db (CSU)/0-133 feet (DSX-1)

B4XXP (PCI) Card 0 Span 1

 RED
0  0
   0  CCS AMI
  0 db (CSU)/0-133 feet (DSX-1)

B4XXP (PCI) Card 0 Span 2

 OK  0
 0  0
CCS AMI   0 db
(CSU)/0-133 feet (DSX-1)

B4XXP (PCI) Card 0 Span 3

 OK  0
 0  0
CCS AMI   0 db
(CSU)/0-133 feet (DSX-1)

B4XXP (PCI) Card 0 Span 4

 OK  0
 0  0
CCS AMI   0 db
(CSU)/0-133 feet (DSX-1)


Note that I have 3 cables connected and 1 port is free

(RED).


---

in AEL dialplan, I run:

Dial(DAHDI/g2/XX);

in the *CLI I see the following:

   -- Requested transfer capability:

0x00 - SPEECH

   -- Called DAHDI/g2/XX
   -- Span 4: Channel 0/1 got hangup,

cause 18

   -- Hungup 'DAHDI/i4/XX-7'
 == Everyone is busy/congested at this time

(1:0/0/1)

   -- Auto fallthrough, channel

'SIP/4053-0089' status is 'CHANUNAVAIL'



If I enable PRI debug:

   -- Executing [@company:1]

Dial("SIP/4053-0001", "DAHDI/g2/XX") in new
stack

PRI Span: 4 -- Making new call for cref 32772
   -- Requested transfer capability:

0x00 - SPEECH

PRI Span: 4
PRI Span: 4 > DL-DATA request
PRI Span: 4 > Protocol Discriminator: Q.931

(8)  len=32

PRI Span: 4 > TEI=0 Call Ref: len= 1 (reference

4/0x4) (Sent from originator)

PRI Span: 4 > Message Type: SETUP (5)
PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open

V(A)=6 K=1

PRI Span: 4
PRI Span: 4 > Protocol Discriminator: Q.931

(8)  len=32

PRI Span: 4 > TEI=0 Call Ref: len= 1 (reference

4/0x4) (Sent from originator)

PRI Span: 4 > Message Type: SETUP (5)
PRI Span: 4 > [04 03 80 90 a3]
PRI Span: 4 > Bearer Capability (len= 5) [ Ext:

1  Coding-Std: 0  Info transfer capability: Speech
(0)

PRI Span: 4 >


 Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)

PRI Span: 4 >


   User information layer 1: A-Law (35)

PRI Span: 4 > [18 01 81]
PRI Span: 4 > Channel ID (len= 3) [ Ext: 1

IntID: Implicit  BRI  Spare: 0
Preferred  Dchan: 0

PRI Span: 4 >


ChanSel: B1 channel

PRI Span: 4 >

]

PRI Span: 4 > [6c 06 21 80 34 30 35 33]
PRI Span: 4 > Calling Party Number (len= 8) [ Ext:

0  TON: National Number (2)  NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)

PRI Span: 4 >


Presentation: Presentation
allowed, User-provided, not screened (0)  '4053' ]

PRI Span: 4 > [70 0a 80 36 35 36 36 36 30 34 39 39]
PRI Span: 4 >

Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues

2012-12-07 Thread Tim Nelson 
- Original Message -
> Tim,
> 
> What version are you on? There is a specific upgrade path for pre
> 3.3.
> 

Yes, that was the issue. I needed to upgrade to version 3.3 first, then upgrade 
to latest 4.x was no problem. Thanks!

--Tim

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Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues

2012-12-07 Thread Dave Fullerton 

On 12/06/2012 04:09 PM, Tim Nelson wrote:

I have a site with Polycom handsets on all the desks, mostly IP650s, some 
IP550s, and some IP450s as well.

I need to update the firmware on the IP450s. However, the firmware simply won't 
load.

The latest firmware (4.0.3 Rev F) supports all phones at this site, and was 
downloaded from here: 
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html

The phone pulls the firmware from the PBX via TFTP (as expected), but always 
results in 'Error: Image is not compatible with the phone'.

As a troubleshooting step, *ALL* firmware has been removed from the TFTP root, 
and *ONLY* the new firmware placed there. So, is the Polycom firmware matrix 
wrong about this phone/firmware compatibility, or am I missing something? The 
bootrom has also been upgraded to the latest without any problems.

Thoughts? My head is getting sore from banging it on my desk... :/

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105




Without knowing what version of the SIP application is already on the 
phones it is hard to say. But it sounds as if your phones are loaded 
with SIP 3.2.x or lower. To upgrade beyond 3.2.x you will need to put 
the 4.4 version of the BootROM on your provisioning server. With SIP 4.0 
and later (and I believe 3.3 as well) the "BootROM" (now called the 
updater) is included in the sip.ld file itself. The 4.4 version of the 
BootROM updates the phone to look for the new updater inside the sip.ld 
file. It should ONLY be used to upgrade phones from SIP 3.2 or lower to 
SIP 3.3 or higher.


Hope this helps.

-Dave


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Re: [asterisk-users] PRI can receive calls but cannot dial out

2012-12-07 Thread Steve Totaro 
On Wed, Dec 5, 2012 at 2:48 AM, Vieri  wrote:
> Hi,
>
> I'm trying to call out from a SIP extension to an outbound destination via a 
> PRI E1 (Digium B410P).
>
> Please take a look at the PRI debug below.
>
> 
>
> # cat /etc/dahdi/system.conf
>
> # Digium Wildcard TDM400P REV I (WCTDM/4)
> fxsks=1
> echocanceller=oslec,1
> fxsks=2
> echocanceller=oslec,2
> fxsks=3
> echocanceller=oslec,3
> fxsks=4
> echocanceller=oslec,4
>
> # Digium Wildcard TDM2400P (WCTDM/0)
> fxsks=5
> echocanceller=oslec,5
> fxsks=6
> echocanceller=oslec,6
> fxsks=7
> echocanceller=oslec,7
> fxsks=8
> echocanceller=oslec,8
> fxsks=9
> echocanceller=oslec,9
> fxsks=10
> echocanceller=oslec,10
> fxsks=11
> echocanceller=oslec,11
> fxsks=12
> echocanceller=oslec,12
>
> # Digium Wildcard B410P (B4/0/1)
> span=3,1,0,CCS,AMI
> bchan=29-30
> hardhdlc=31
> echocanceller=oslec,29-30
>
> # Digium Wildcard B410P (B4/0/2)
> span=4,2,0,CCS,AMI
> bchan=32-33
> hardhdlc=34
> echocanceller=oslec,32-33
>
> # Digium Wildcard B410P (B4/0/3)
> span=5,3,0,CCS,AMI
> bchan=35-36
> hardhdlc=37
> echocanceller=oslec,35-36
>
> # Digium Wildcard B410P (B4/0/4)
> span=6,4,0,CCS,AMI
> bchan=38-39
> hardhdlc=40
> echocanceller=oslec,38-39
>
> 
>
> # lsmod | grep wcb4xxp
> wcb4xxp66250  12
> dahdi 169899  65 
> dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm
>
> 
>
> # cat chan_dahdi.conf
>
> [trunkgroups]
>
> [channels]
> transfer = yes
> usecallerid = yes
> cidsignalling = dtmf
> callwaiting = yes
> usecallingpres = yes
> callwaitingcallerid = yes
> threewaycalling = yes
> canpark = yes
> cancallforward = yes
> callreturn = yes
> callprogress = no
> overlapdial = yes
> echocancel = yes
> facilityenable = yes
> immediate = no
> busydetect = no
>
> ; Digium Wildcard TDM400P REV I (WCTDM/4)
> signalling = fxs_ks
> txgain = 1.0
> rxgain = 14.0
> group = 3
> context = incoming-dahdi-3
> faxdetect = incoming
> channel => 1,2,3,4
>
> ; Digium Wildcard TDM2400P (WCTDM/0)
> group = 4
> context = incoming-dahdi-4
> faxdetect = incoming
> channel => 5,6,7,8,9,10,11,12
>
> ; Digium Wildcard B410P (B4/0/1)
> signalling = bri_cpe
> switchtype = euroisdn
> rxgain = 2.0
> group = 2
> context = incoming-dahdi-2
> faxdetect = incoming
> channel => 29-30
>
> ; Digium Wildcard B410P (B4/0/2)
> channel => 32-33
>
> ; Digium Wildcard B410P (B4/0/3)
> channel => 35-36
>
> ; Digium Wildcard B410P (B4/0/4)
> channel => 38-39
>
> ---
>
> # asterisk -rx "dahdi show status"
> Description  Alarms  IRQbpviol CRCFra 
> Codi Options  LBO
> Wildcard TDM400P REV I Board 5   OK  0  0  0  CAS Unk 
>   0 db (CSU)/0-133 feet (DSX-1)
> Wildcard TDM2400POK  0  0  0  CAS Unk 
>   0 db (CSU)/0-133 feet (DSX-1)
> B4XXP (PCI) Card 0 Span 1RED 0  0  0  CCS AMI 
>   0 db (CSU)/0-133 feet (DSX-1)
> B4XXP (PCI) Card 0 Span 2OK  0  0  0  CCS AMI 
>   0 db (CSU)/0-133 feet (DSX-1)
> B4XXP (PCI) Card 0 Span 3OK  0  0  0  CCS AMI 
>   0 db (CSU)/0-133 feet (DSX-1)
> B4XXP (PCI) Card 0 Span 4OK  0  0  0  CCS AMI 
>   0 db (CSU)/0-133 feet (DSX-1)
>
> Note that I have 3 cables connected and 1 port is free (RED).
>
> ---
>
> in AEL dialplan, I run:
>
> Dial(DAHDI/g2/XX);
>
> in the *CLI I see the following:
>
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called DAHDI/g2/XX
> -- Span 4: Channel 0/1 got hangup, cause 18
> -- Hungup 'DAHDI/i4/XX-7'
>   == Everyone is busy/congested at this time (1:0/0/1)
> -- Auto fallthrough, channel 'SIP/4053-0089' status is 'CHANUNAVAIL'
>
>
> If I enable PRI debug:
>
> -- Executing [@company:1] Dial("SIP/4053-0001", 
> "DAHDI/g2/XX") in new stack
> PRI Span: 4 -- Making new call for cref 32772
> -- Requested transfer capability: 0x00 - SPEECH
> PRI Span: 4
> PRI Span: 4 > DL-DATA request
> PRI Span: 4 > Protocol Discriminator: Q.931 (8)  len=32
> PRI Span: 4 > TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator)
> PRI Span: 4 > Message Type: SETUP (5)
> PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open V(A)=6 K=1
> PRI Span: 4
> PRI Span: 4 > Protocol Discriminator: Q.931 (8)  len=32
> PRI Span: 4 > TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator)
> PRI Span: 4 > Message Type: SETUP (5)
> PRI Span: 4 > [04 03 80 90 a3]
> PRI Span: 4 > Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info 
> transfer capability: Speech (0)
> PRI Span: 4 >  Ext: 1  Trans mode/rate: 64kbps, 
> circuit-mode (16)
> PRI Span: 4 >User information layer 1: A-Law 
> (35)
> PRI Span: 4 > [18 01 81]
> PRI Span: 4 > Channel ID (len= 3) [ Ext: 1  IntID: Implicit  BRI  Spare: 0  
> Preferred

Re: [asterisk-users] - configure ring group

2012-12-07 Thread Paolo De Michele 
hi Leandro,

thank to you for your reply but I think I shall apply the configurations
that advised me AJS
many thanks for your help
cheers

On 12/06/2012 09:13 AM, Leandro Dardini wrote:
> 100 extension on a row is not feasible... the queue strategy is the
> only possible solution. If you check the queue.conf file you'll find
> you can define a "Queue" and add as many members you like. One of the
> strategy available is the "Ring all" where all the members in the
> queue will be ring. You can let your peers to log in/log out of the
> queue via dialplan
>
> Leandro
>
> 2012/12/6 Paolo De Michele  >
>
> hi all,
>
> thanks for your replies
> if you have 100 extensions, put them all into a single string?
> so: (SIP/1001&SIP/1002&SIP/1003...until you get to 100?
>
> It is very difficult to manage such a thing, no?
>
> I don't understand the queues,ringall. can someone explain?
> thanks in advance
>
>
> On 12/05/2012 10:59 PM, Danny Nicholas wrote:
>>
>> You “can” do the queues/ringall, but you’re increasing your pay
>> grade by doing so.
>>
>>  
>>
>> *From:*asterisk-users-boun...@lists.digium.com
>> 
>> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
>> *Carlos Rojas
>> *Sent:* Wednesday, December 05, 2012 3:58 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] - configure ring group
>>
>>  
>>
>> Maybe, 
>>
>>  
>>
>> You can do that, with queues, and ringall strategy.
>>
>> On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini
>> mailto:ldard...@gmail.com>> wrote:
>>
>> You can dial all the extensions at once, putting all them in the
>> dial string, separated by &. There is no other method.
>>
>>  
>>
>> Leandro
>>
>> 2012/12/5 Paolo De Michele > >
>>
>> hi all,
>>
>> I want have an information about ring group in asterisk
>> (1.8.16 - centos 6.3)
>> I have configured skypeforasterisk for incoming call to one
>> extension and it works
>>
>> now,my chan_skype.conf is:
>>
>> [general]
>>
>> default_user=user-skype
>>
>> [user-skype]
>> secret=x
>> context=from-skype
>> exten=
>> disallow=all
>> allow=ulaw
>> allow=alaw
>>
>> my extensions.conf:
>>
>> [from-skype]
>>
>> exten => ,1,Verbose(2,Incoming Skype Call)
>>same => n,Answer()
>>same => n,Dial(SIP/1000&SIP/2000&SIP/3000,30)
>>same => n,Playback(user&is-curntly-unavail)
>>same => n,Hangup()
>>
>> at right time the internal ring are 1000, 2000 and 3000
>> I have the extension from 1000 to 1005, 2000 to 2005 and from
>> 3000 to 3005
>> I can ring him all? I can group the configuration into a
>> single string?
>>
>> let me know something
>> thanks in advance
>>
>>
>>  
>>
>> --
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Re: [asterisk-users] [SOLVED] - configure ring group

2012-12-07 Thread Paolo De Michele 

On 12/06/2012 10:21 AM, A J Stiles wrote:
> On Wednesday 05 December 2012, Paolo De Michele wrote:
>> hi all,
>>
>> I want have an information about ring group in asterisk (1.8.16 - centos
>> 6.3)
>> I have configured skypeforasterisk for incoming call to one extension
>> and it works
>> . [stuff deleted] .
>> at right time the internal ring are 1000, 2000 and 3000
>> I have the extension from 1000 to 1005, 2000 to 2005 and from 3000 to 3005
>> I can ring him all? I can group the configuration into a single string?
> If you need to ring groups of phones, try something like
>
> [globals]
> ONES=SIP/1000&SIP/1001&SIP/1002&SIP/1003&SIP/1004&SIP/1005
> TWOS=SIP/2000&SIP/2001&SIP/2002&SIP/2003&SIP/2004&SIP/2005
> THREES=SIP/3000&SIP/3001&SIP/3002&SIP/3003&SIP/3004&SIP/3005
>
> [some-context]
> exten => 4000,1,Dial(${ONES}&${TWOS}&${THREES})
> exten => 4000,n,Hangup()
>
> That should make things more manageable.
>
> Also:  Be sure to use *meaningful* variable names!  If your numbers are split 
> according to department  (management, accounts, sales, purchasing, lab, qa, 
> factory floor, training, &c.)  then name your groups accordingly.  If you get 
> your dialplan right, you won't have to touch it for ages; and when you come 
> back to have to edit it, you'll forget what anything was for.
>
AJS,
thank you for you reply
I think I shall apply this type of configurations - It's more manageable

thank you very much
cheers
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Re: [asterisk-users] PRI can receive calls but cannot dial out

2012-12-07 Thread Vieri 


> Am 05.12.2012 08:48, schrieb Vieri:
> > Hi,
> >
> > I'm trying to call out from a SIP extension to an
> outbound destination via a PRI E1 (Digium B410P).
> >
> > Please take a look at the PRI debug below.
> >
> > 
> >
> > # cat /etc/dahdi/system.conf
> >
> > # Digium Wildcard TDM400P REV I (WCTDM/4)
> > fxsks=1
> > echocanceller=oslec,1
> > fxsks=2
> > echocanceller=oslec,2
> > fxsks=3
> > echocanceller=oslec,3
> > fxsks=4
> > echocanceller=oslec,4
> >
> > # Digium Wildcard TDM2400P (WCTDM/0)
> > fxsks=5
> > echocanceller=oslec,5
> > fxsks=6
> > echocanceller=oslec,6
> > fxsks=7
> > echocanceller=oslec,7
> > fxsks=8
> > echocanceller=oslec,8
> > fxsks=9
> > echocanceller=oslec,9
> > fxsks=10
> > echocanceller=oslec,10
> > fxsks=11
> > echocanceller=oslec,11
> > fxsks=12
> > echocanceller=oslec,12
> >
> > # Digium Wildcard B410P (B4/0/1)
> > span=3,1,0,CCS,AMI
> > bchan=29-30
> > hardhdlc=31
> > echocanceller=oslec,29-30
> >
> > # Digium Wildcard B410P (B4/0/2)
> > span=4,2,0,CCS,AMI
> > bchan=32-33
> > hardhdlc=34
> > echocanceller=oslec,32-33
> >
> > # Digium Wildcard B410P (B4/0/3)
> > span=5,3,0,CCS,AMI
> > bchan=35-36
> > hardhdlc=37
> > echocanceller=oslec,35-36
> >
> > # Digium Wildcard B410P (B4/0/4)
> > span=6,4,0,CCS,AMI
> > bchan=38-39
> > hardhdlc=40
> > echocanceller=oslec,38-39
> >
> > 
> >
> > # lsmod | grep wcb4xxp
> > wcb4xxp             
>   66250  12
> > dahdi             
>    169899  65
> dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm
> >
> > 
> >
> > # cat chan_dahdi.conf
> >
> > [trunkgroups]
> >
> > [channels]
> > transfer = yes
> > usecallerid = yes
> > cidsignalling = dtmf
> > callwaiting = yes
> > usecallingpres = yes
> > callwaitingcallerid = yes
> > threewaycalling = yes
> > canpark = yes
> > cancallforward = yes
> > callreturn = yes
> > callprogress = no
> > overlapdial = yes
> > echocancel = yes
> > facilityenable = yes
> > immediate = no
> > busydetect = no
> >
> > ; Digium Wildcard TDM400P REV I (WCTDM/4)
> > signalling = fxs_ks
> > txgain = 1.0
> > rxgain = 14.0
> > group = 3
> > context = incoming-dahdi-3
> > faxdetect = incoming
> > channel => 1,2,3,4
> >
> > ; Digium Wildcard TDM2400P (WCTDM/0)
> > group = 4
> > context = incoming-dahdi-4
> > faxdetect = incoming
> > channel => 5,6,7,8,9,10,11,12
> >
> > ; Digium Wildcard B410P (B4/0/1)
> > signalling = bri_cpe
> > switchtype = euroisdn
> > rxgain = 2.0
> > group = 2
> > context = incoming-dahdi-2
> > faxdetect = incoming
> > channel => 29-30
> >
> > ; Digium Wildcard B410P (B4/0/2)
> > channel => 32-33
> >
> > ; Digium Wildcard B410P (B4/0/3)
> > channel => 35-36
> >
> > ; Digium Wildcard B410P (B4/0/4)
> > channel => 38-39
> >
> > ---
> >
> > # asterisk -rx "dahdi show status"
> > Description           
>                
>   Alarms  IRQ    bpviol CRC   
> Fra Codi Options  LBO
> > Wildcard TDM400P REV I Board 5     
>      OK      0 
>     0      0     
> CAS Unk           0 db
> (CSU)/0-133 feet (DSX-1)
> > Wildcard TDM2400P         
>               OK 
>     0      0     
> 0      CAS Unk       
>    0 db (CSU)/0-133 feet (DSX-1)
> > B4XXP (PCI) Card 0 Span 1       
>         RED 
>    0      0   
>   0      CCS AMI     
>      0 db (CSU)/0-133 feet (DSX-1)
> > B4XXP (PCI) Card 0 Span 2       
>         OK      0 
>     0      0     
> CCS AMI           0 db
> (CSU)/0-133 feet (DSX-1)
> > B4XXP (PCI) Card 0 Span 3       
>         OK      0 
>     0      0     
> CCS AMI           0 db
> (CSU)/0-133 feet (DSX-1)
> > B4XXP (PCI) Card 0 Span 4       
>         OK      0 
>     0      0     
> CCS AMI           0 db
> (CSU)/0-133 feet (DSX-1)
> >
> > Note that I have 3 cables connected and 1 port is free
> (RED).
> >
> > ---
> >
> > in AEL dialplan, I run:
> >
> > Dial(DAHDI/g2/XX);
> >
> > in the *CLI I see the following:
> >
> >      -- Requested transfer capability:
> 0x00 - SPEECH
> >      -- Called DAHDI/g2/XX
> >      -- Span 4: Channel 0/1 got hangup,
> cause 18
> >      -- Hungup 'DAHDI/i4/XX-7'
> >    == Everyone is busy/congested at this time
> (1:0/0/1)
> >      -- Auto fallthrough, channel
> 'SIP/4053-0089' status is 'CHANUNAVAIL'
> >
> >
> > If I enable PRI debug:
> >
> >      -- Executing [@company:1]
> Dial("SIP/4053-0001", "DAHDI/g2/XX") in new
> stack
> > PRI Span: 4 -- Making new call for cref 32772
> >      -- Requested transfer capability:
> 0x00 - SPEECH
> > PRI Span: 4
> > PRI Span: 4 > DL-DATA request
> > PRI Span: 4 > Protocol Discriminator: Q.931
> (8)  len=32
> > PRI Span: 4 > TEI=0 Call Ref: len= 1 (reference
> 4/0x4) (Sent from originator)
> > PRI Span: 4 > Message Type: SETUP (5)
> > PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open
> V(A)=6 K=1
> > PRI Span: 4
> > PRI Span: 4 > Protocol Discriminator: Q.931
> (8)  len=32
> > PRI Span: 4 > TEI=0 Call Ref: len= 1 (reference
> 4/0x4) (Sent from originator)
> > PRI Span: 4 > Message Type: SETUP (5)
> > PRI Span: 4 > [04 03 80 90 a3]
> > PRI Span

Re: [asterisk-users] BLF and call-limit in 1.8

2012-12-07 Thread Pan B. Christensen 
But Asterisk doesn't send ANY notification regarding change of state (like 
ringing) unless call-limit is 1. In my opinion, ringing status (and probably 
several others) shouldn't take into consideration at what point a device is 
considered as busy.
This worked fine in 1.4 (we skipped 1.6) and seems broken in 1.8. Busy-level is 
1 in both configurations.

After some more investigation, I may have found a clue. When I type sip show 
peer , the following values are reported:
With DB value 4 or 3 or 2 or 0:
  Call limit   : 0
With DB value 1:
  Call limit   : 2147483647

This doesn't look correct to me...

-Pan
  - Original Message - 
  From: Olle E. Johansson 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Friday, December 07, 2012 8:20 AM
  Subject: Re: [asterisk-users] BLF and call-limit in 1.8




  6 dec 2012 kl. 16:54 skrev Danny Nicholas :


Not sure about this since I use the 10/11 branches and not 1.8, but I think 
you need to use the deprecated call-limit for BLF and the new busylimit for the 
other features you need.
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf




  Call-limit is the limit on the number of calls you can take and also sets a 
device to BUSY. Since you want to be able to transfer calls, you need at least 
two. But this did not set the phone to busy on one call. That's why we added 
busy-limit that can be set to the level you want device states to signal busy, 
but still give the ability to the phone to set up more calls.


  counteronpeer is the same as limitonpeer, just a new name.


  /O



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen
Sent: Thursday, December 06, 2012 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] BLF and call-limit in 1.8

Hello

We have recently upgraded our internal PBX from 1.4 to 1.8. This made the 
BLF lamps on our Polycom phones stop working. After a lot of googling and a lot 
of testing, I have been unable to find a solution.

I did try to change the call-limit value from 4 to 1, and this actually 
made BLF work (noone suggested this, and what documantation I can find states 
that this option is deprecated). This change has other implications, however. 
Call waiting stops working, queues don't offer calls if the user is in a 
private call etc.

We have customers that require both BLF and call waiting at the same time.


We are running Asterisk 1.8.11-cert7

I've made the following additions to sip.conf [general]:
callcounter=yes
counteronpeer=yes (undocumented? Supposed to replace limitonpeers?)

(old relevant values, unchanged)
allowsubscribe=yes
subscribecontext=blf
notifyringing=yes
notifyhold=yes
limitonpeers=yes 

I also tried may other suggestions I've found like placing the hints in the 
same context as the extensions and removing subscribecontext.

Is there something I'm missing? Is something not working correctly?

Thanks in advance,
Pan
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