[asterisk-users] Digium IP Phones

2012-05-09 Thread Danny Dias 
Hello,

Im looking to buy a digium phone D70 unit just for testing on lab; to
really understand the phone and features.

I cant find any website with opinions; any here? Are they really valuable
to the price? (D70 quite expensive)

Does the SDK for building apps is usable? Can you build powerfull apps?
Examples?

Many thanks
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[asterisk-users] Why do I get call twice in one go?

2012-05-09 Thread Shahid H 
I understand why do I get call twice to my mobile when I execute the
following AMI command sets:

ACTION: Originate
Channel: Local/800@test
Timeout: 6
Priority: 1

and my dialplan look like this:

[test]
exten => 800,1,DIAL(SIP/447xx@voip);
exten => 800,n,Hangup()


How to prevent getting called twice in one go when I execute this AMI
command?

Thanks...
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Re: [asterisk-users] Broadvoice Got SIP response 503 Service Unavailable

2012-05-09 Thread Ing CIP. Alejandro Celi 

The solution:

Stop the Asterisk, made a backup of configuration files, delete files
and modules previously installed.

Then Reinstall with default config files with "make samples" and I only
insert the configuration of Broadvoice and immediately worked. Then I
restore the configuration files /etc/asterisk (thinking it was a problem
of modules) and stopped working again, It mean that the problem was in
the configuration files, or too much or too less lines.

So what I made is transfer line x line what I needed from old to new
configuration file.

As soon detected where it was what was missing or left over, I will
publish the list

regards,

-- 
Ing CIP. Alejandro Celi Mariátegui 




El vie, 04-05-2012 a las 09:23 +, isr...@gmail.com escribió:

> Broadvoice has a lot of problems for the last 2 months 
> 
> -Original Message-
> From: "Ing. CIP Alejandro Celi Mariategui" 
> Sender: asterisk-users-boun...@lists.digium.com
> Date: Fri, 04 May 2012 02:11:11 
> To: 
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Subject: [asterisk-users] Broadvoice Got SIP response 503 Service Unavailable
> 
> Hi,
> 
> I'm running Asterisk 1.8.11.1 @office.
> 
> The Broadvoice service work fine with all 1.6 version and early 1.8  
> behind a NAT but about 2 months ago stop working.
> 
> No made changes in the firewall NAT rules. Right now I'm @home via my  
> Xlite softphone working fine without problems
> 
> Any suggestions or thoughts?
> 
> Alex Celi
> 
> 
> 
> This is the info
> 
> 
> central*CLI> sip show peers
> Name/username  HostDyn  
> Forcerport ACL Port Status
> 488/488181.64.96.122D   
> 11037OK (182 ms)
> sip.broadvoice.com/305422  206.15.148.221   
>5060 OK (131 ms)
> 
> 
> sip.conf
>  externip=190.12.68.20
>  localnet=192.168.20.0/255.255.255.0
>  localnet=192.168.10.0/255.255.255.0
>  nat=comedia
> 
>  pedantic=no
>  register =>  
> 3054221...@sip.broadvoice.com:XX:3054221...@sip.broadvoice.com
> 
>  [sip.broadvoice.com]
>  type=friend
>  host=sip.broadvoice.com
>  fromdomain=sip.broadvoice.com
>  fromuser=3054221494
>  defaultuser=3054221494
>  authname=3054221494
>  secret=X
>  context=entrantes
>  dtmfmode=inband
>  dtmf=inband
>  nat=comedia
>  directmedia=no
>  qualify=yes
>  callgroup=1
>  pickupgroup=1
>  disallow=all
>  allow=ulaw
>  allow=alaw
> 
> 
> 
> I turned on sip debug. This is what I received
> 
> 181.64.96.122: Is my home IP
> 190.12.68.20 or central.cipher.pe: is office IP
> 206.15.148.221: Broadvoice Server
> 
> 
>  <--- SIP read from UDP:181.64.96.122:11037 --->
>  INVITE sip:90018006273...@central.cipher.pe SIP/2.0
>  Via: SIP/2.0/UDP  
> 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
>  Max-Forwards: 70
>  Contact: 
>  To: "90018006273999"
>  From: "488";tag=93cce179
>  Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
>  CSeq: 1 INVITE
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,  
> SUBSCRIBE, INFO
>  Content-Type: application/sdp
>  User-Agent: X-Lite release 1014k stamp 56015
>  Content-Length: 235
> 
>  v=0
>  o=- 8 2 IN IP4 192.168.7.33
>  s=CounterPath X-Lite 3.0
>  c=IN IP4 192.168.7.33
>  t=0 0
>  m=audio 2424 RTP/AVP 0 8 3 101
>  a=fmtp:101 0-15
>  a=rtpmap:101 telephone-event/8000
>  a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424
>  a=sendrecv
>  <->
>  --- (12 headers 10 lines) ---
>  Sending to 181.64.96.122:11037 (NAT)
>  Using INVITE request as basis request -  
> ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
>  Found peer '488' for '488' from 181.64.96.122:11037
> 
>  <--- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --->
>  SIP/2.0 401 Unauthorized
>  Via: SIP/2.0/UDP  
> 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;received=181.64.96.122;rport=11037
>  From: "488";tag=93cce179
>  To: "90018006273999";tag=as77d2f824
>  Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
>  CSeq: 1 INVITE
>  Server: Asterisk PBX 1.8.11.1
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,  
> NOTIFY, INFO, PUBLISH
>  Supported: replaces, timer
>  WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> nonce="0a1fded4"
>  Content-Length: 0
> 


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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Richard Mudgett 
Please just reply to the mailing list.

- Original Message -
> Right you are,
> 
> but when using bri_cpe I get:
> 
> [May  9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI
> Error
> on span 4: Received MDL/TEI managemement message, but configured for
> mode other than PTMP!
> 
> This repeats itself every second.
> 
> The
> 
> bri_cpe_ptmp
> 
> settings seems to give the least troubles, but no calling possible,

Are you able to make calls when in PTP mode?  The warning message is just
complaining about receiving unexpected TEI management messages because
the span is in PTP mode.  It is otherwise benign if the line is really PTP.

If you can make calls, please create a JIRA issue on the PRI project so the
message level can be reduced.  Please attach an intense pri debug output
showing the received MDL messages.

pri set debug 2 span 4

https://issues.asterisk.org/jira

Richard

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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Bart Coninckx 

Right you are,

but when using bri_cpe I get:

[May  9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI Error 
on span 4: Received MDL/TEI managemement message, but configured for 
mode other than PTMP!


This repeats itself every second.

The

bri_cpe_ptmp

settings seems to give the least troubles, but no calling possible,

BC



On 05/09/12 22:10, Richard Mudgett wrote:

Hi Khalid,

my setup is almost identical

except for
loadzone = be
defaultzone = be

(obviously)

and in

chan_dahdi.conf:

[isdn4]
signaling = bri_cpe_ptmp
switchtype = euroisdn
group = 2
context = isdn
dahdichan = 10,11


this results into:

ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable
to receive TEI from network in state 2(Assign awaiting TEI)!

and after trying to call:

May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136
my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of
span 4
[May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms:
Detected alarm on channel 10: Red Alarm
[May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms:
Detected alarm on channel 11: Red Alarm
[May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI
Span: 4 Unable to receive TEI from network in state 3(Establish
awaiting TEI)!
[May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136
my_handle_dchan_exception: PRI got event: No more alarm (5) on
D-channel of span 4
[May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms:
Alarm cleared on channel 10
[May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms:
Alarm cleared on channel 11


An old Asterisk 1.4 installation with mISDN connected to the same
line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more
problems on the console however (every second).

bri_cpe_ptp is not a valid value for the signaling parameter.

 From chan_dahdi.conf.sample:
; bri_cpe:BRI PTP signalling, CPE side
; bri_net:BRI PTP signalling, Network side
; bri_cpe_ptmp:   BRI PTMP signalling, CPE side
; bri_net_ptmp:   BRI PTMP signalling, Network side

Richard

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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Richard Mudgett 
> Hi Khalid,
> 
> my setup is almost identical
> 
> except for
> loadzone = be
> defaultzone = be
> 
> (obviously)
> 
> and in
> 
> chan_dahdi.conf:
> 
> [isdn4]
> signaling = bri_cpe_ptmp
> switchtype = euroisdn
> group = 2
> context = isdn
> dahdichan = 10,11
> 
> 
> this results into:
> 
> ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable
> to receive TEI from network in state 2(Assign awaiting TEI)!
> 
> and after trying to call:
> 
> May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136
> my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of
> span 4
> [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms:
> Detected alarm on channel 10: Red Alarm
> [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms:
> Detected alarm on channel 11: Red Alarm
> [May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI
> Span: 4 Unable to receive TEI from network in state 3(Establish
> awaiting TEI)!
> [May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136
> my_handle_dchan_exception: PRI got event: No more alarm (5) on
> D-channel of span 4
> [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms:
> Alarm cleared on channel 10
> [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms:
> Alarm cleared on channel 11
> 
> 
> An old Asterisk 1.4 installation with mISDN connected to the same
> line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more
> problems on the console however (every second).

bri_cpe_ptp is not a valid value for the signaling parameter.

>From chan_dahdi.conf.sample:
; bri_cpe:BRI PTP signalling, CPE side
; bri_net:BRI PTP signalling, Network side
; bri_cpe_ptmp:   BRI PTMP signalling, CPE side
; bri_net_ptmp:   BRI PTMP signalling, Network side

Richard

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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Bart Coninckx 

Hi Khalid,

my setup is almost identical

except for
loadzone = be
defaultzone = be

(obviously)

and in

chan_dahdi.conf:

[isdn4]
signaling = bri_cpe_ptmp
switchtype = euroisdn
group = 2
context = isdn
dahdichan = 10,11


this results into:

 ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to 
receive TEI from network in state 2(Assign awaiting TEI)!


and after trying to call:

May  9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136 
my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of span 4
[May  9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: 
Detected alarm on channel 10: Red Alarm
[May  9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: 
Detected alarm on channel 11: Red Alarm
[May  9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI 
Span: 4 Unable to receive TEI from network in state 3(Establish awaiting 
TEI)!
[May  9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136 
my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel 
of span 4
[May  9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: 
Alarm cleared on channel 10
[May  9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: 
Alarm cleared on channel 11



An old Asterisk 1.4 installation with mISDN connected to the same line 
used ptp, not ptmp. When I used bri_cpe_ptp, I get even more problems on 
the console however (every second).


This is why I wondered if DAHDI is supposed to work over here,

cheers,

BC



On 05/09/12 21:14, khalid touati wrote:

Hi Bart,
here is a working configuration in Netherlands:
/etc/dahdi/system.conf:

span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3

span = 2,1,0,ccs,ami
bchan = 4,5
hardhdlc = 6

span = 3,1,0,ccs,ami
bchan = 7,8
hardhdlc = 9

span = 4,1,0,ccs,ami
bchan = 10,11
hardhdlc = 12

loadzone= nl
defaultzone= nl   (of course change those to your country initials)

/etc/asterisk/chan_dahdi.conf:

group = 1
signalling = bri_cpe_ptmp
switchtype = euroisdn
context = mainmenu
echocancel = yes
channel => 1,2,4,5,7,8,10,11

I am not using dahdi-channels, hope it helps!


On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx 
mailto:bart.conin...@telenet.be>> wrote:


Hi,

I'm experiencing difficulties to get a B410P running with Asterisk
10.3.1 and DAHDI 2.6.1.
Am I supposed to use DAHDI for this card and ISDN BRI for my
country (Belgium)?

thx,

BC

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Khalid Touati
Network Administrator at Endosoft, LLC
CCNA




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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread khalid touati 
Hi Bart,
here is a working configuration in Netherlands:
/etc/dahdi/system.conf:

span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3

span = 2,1,0,ccs,ami
bchan = 4,5
hardhdlc = 6

span = 3,1,0,ccs,ami
bchan = 7,8
hardhdlc = 9

span = 4,1,0,ccs,ami
bchan = 10,11
hardhdlc = 12

loadzone= nl
defaultzone= nl   (of course change those to your country initials)

/etc/asterisk/chan_dahdi.conf:

group = 1
signalling = bri_cpe_ptmp
switchtype = euroisdn
context = mainmenu
echocancel = yes
channel => 1,2,4,5,7,8,10,11

I am not using dahdi-channels, hope it helps!


On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx wrote:

> Hi,
>
> I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1
> and DAHDI 2.6.1.
> Am I supposed to use DAHDI for this card and ISDN BRI for my country
> (Belgium)?
>
> thx,
>
> BC
>
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>  
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CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread khalid touati 
Yeah they have a wonderful policy that says "ISDN team are not contactable"
:(   thanks a lot!!

On Wed, May 9, 2012 at 3:06 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09-05-12 20:57, khalid touati wrote:
> > Yeah sorry for that, I realized something is missing after I sent the
> > email, but it is exactly what I have (other than order here, which
> > doesn't really matter: you posted ami,te,term, I have ami,term,te).
> > Actually I had couple technicians from digium look at it and they said
> > BT equipements is not responding to the card within a certain range that
> > the card is looking for (i'm not sure what range but I do believe too
> > it's a BT issue), But I have run all the couple command that Patrick
> > suggested (to double check), tested again and still same kind of errors.
> > But Thank you very much Patrick for the guide, I was looking for that
> > it's been a couple days!!
> > I just hope someone that has the exact same issue or someone with
> > previous BT experience see this and help :) ..we never know :) !
>
> Too bad you could not (yet) make it work. Hope you get somewhere with
> BT. Once you get past the people following those silly scripts you
> should be able to talk to someone who has a clue and resolve this issue.
>
> Good luck!
>
> Regards,
> Patrick
>
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>



-- 
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Kevin P. Fleming 

On 05/09/2012 12:59 PM, Bart Coninckx wrote:

Hi,

I'm experiencing difficulties to get a B410P running with Asterisk
10.3.1 and DAHDI 2.6.1.
Am I supposed to use DAHDI for this card and ISDN BRI for my country
(Belgium)?


That is the supported method to use in standard Asterisk, yes. DAHDI, 
libpri and Asterisk working together.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread Patrick Lists 
On 09-05-12 20:57, khalid touati wrote:
> Yeah sorry for that, I realized something is missing after I sent the
> email, but it is exactly what I have (other than order here, which
> doesn't really matter: you posted ami,te,term, I have ami,term,te).
> Actually I had couple technicians from digium look at it and they said
> BT equipements is not responding to the card within a certain range that
> the card is looking for (i'm not sure what range but I do believe too
> it's a BT issue), But I have run all the couple command that Patrick
> suggested (to double check), tested again and still same kind of errors.
> But Thank you very much Patrick for the guide, I was looking for that
> it's been a couple days!!
> I just hope someone that has the exact same issue or someone with
> previous BT experience see this and help :) ..we never know :) !

Too bad you could not (yet) make it work. Hope you get somewhere with
BT. Once you get past the people following those silly scripts you
should be able to talk to someone who has a clue and resolve this issue.

Good luck!

Regards,
Patrick

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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread khalid touati 
Yeah sorry for that, I realized something is missing after I sent the
email, but it is exactly what I have (other than order here, which doesn't
really matter: you posted ami,te,term, I have ami,term,te).
Actually I had couple technicians from digium look at it and they said BT
equipements is not responding to the card within a certain range that the
card is looking for (i'm not sure what range but I do believe too it's a BT
issue), But I have run all the couple command that Patrick suggested (to
double check), tested again and still same kind of errors.
But Thank you very much Patrick for the guide, I was looking for that it's
been a couple days!!
I just hope someone that has the exact same issue or someone with previous
BT experience see this and help :) ..we never know :) !

On Wed, May 9, 2012 at 2:40 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09-05-12 19:54, khalid touati wrote:
> > Thank you for your answer,  I think I posted dhadi version and so but
> > let me add more details and recap them below:
> >
> > We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
> > digium card 1HA8-0400BLF
> >
> > output of dahdi_hardware: pci::04:08.0 wctdm24xxp+  d161:8008
> > HB8-
> >
> > From BT side: it is called by BT a "system access" ISDN2 BRI (per BT NT
> > mode and signaling as PTP)
> >
> > chan_dahdi.conf
> >
> > ; Span 1: WCBRI/0/0 "HB8-" (MASTER) AMI/CCS
> > group=1,11
> > context= mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 1-2
> > context = default
> > group = 63
> >
> > ; Span 2: WCBRI/0/1 "HB8-" AMI/CCS
> > group=1,12
> > context=mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 4-5
> > context = default
> > group = 63
> >
> > ; Span 3: WCBRI/0/2 "HB8-" AMI/CCS
> > group=1,13
> > context=mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 7-8
> > context = default
> > group = 63
> >
> > ; Span 4: WCBRI/0/3 "HB8-" AMI/CCS
> > group=1,14
> > context=mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 10-11
> > context = default
> > group = 63
> >
> > the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error
> > in state 7(Multi-frame established)
> > *
> > Thank you!!
>
> You did not provide system.conf. Do you have something like this (may
> have errors, I did not check):
>
> loadzone = uk
> defaultzone = uk
> span => 1,1,0,ccs,ami,te,term
> bchan = 1,2
> hardhdlc = 3
>
> span => 2,2,0,ccs,ami,te,term
> bchan = 4,5
> hardhdlc = 6
>
> span => 3,3,0,ccs,ami,te,term
> bchan = 7,8
> hardhdlc = 9
>
> span => 4,4,0,ccs,ami,te,term
> bchan = 10,11
> hardhdlc = 12
>
>
> Then as root:
> modprobe wctdm23xxp
>
> And as root:
> dahdi_cfg -vvv
>
> And check if all is well (green leds, happy messages in
> /var/log/messages, etc.).
>
>
> Then in chan_dahdi.conf use something like:
>
> ;BRI Module
> group = 1
> signalling = bri_cpe
> context = incoming
> channel => 1,2,4,5,7,8,10,11
>
> Your chan_dahdi.conf has "group" and "context" multiple times and that
> does not seem right (admittedly it's been ages since I setup a Digium
> card).
>
> Hope this helps. If not follow the installation manual step for step or
> call Digium support.
>
> http://docs.digium.com/H8/hx8_series_manual.pdf
>
> Regards,
> Patrick
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread Patrick Lists 
On 09-05-12 19:54, khalid touati wrote:
> Thank you for your answer,  I think I posted dhadi version and so but
> let me add more details and recap them below:
> 
> We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
> digium card 1HA8-0400BLF
> 
> output of dahdi_hardware: pci::04:08.0 wctdm24xxp+  d161:8008
> HB8-
> 
> From BT side: it is called by BT a "system access" ISDN2 BRI (per BT NT
> mode and signaling as PTP)
> 
> chan_dahdi.conf
> 
> ; Span 1: WCBRI/0/0 "HB8-" (MASTER) AMI/CCS
> group=1,11
> context= mainmenu
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 1-2
> context = default
> group = 63
> 
> ; Span 2: WCBRI/0/1 "HB8-" AMI/CCS
> group=1,12
> context=mainmenu
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 4-5
> context = default
> group = 63
> 
> ; Span 3: WCBRI/0/2 "HB8-" AMI/CCS
> group=1,13
> context=mainmenu
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 7-8
> context = default
> group = 63
> 
> ; Span 4: WCBRI/0/3 "HB8-" AMI/CCS
> group=1,14
> context=mainmenu
> switchtype = euroisdn
> signalling = bri_cpe
> channel => 10-11
> context = default
> group = 63
> 
> the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error
> in state 7(Multi-frame established)
> *
> Thank you!!

You did not provide system.conf. Do you have something like this (may
have errors, I did not check):

loadzone = uk
defaultzone = uk
span => 1,1,0,ccs,ami,te,term
bchan = 1,2
hardhdlc = 3

span => 2,2,0,ccs,ami,te,term
bchan = 4,5
hardhdlc = 6

span => 3,3,0,ccs,ami,te,term
bchan = 7,8
hardhdlc = 9

span => 4,4,0,ccs,ami,te,term
bchan = 10,11
hardhdlc = 12


Then as root:
modprobe wctdm23xxp

And as root:
dahdi_cfg -vvv

And check if all is well (green leds, happy messages in
/var/log/messages, etc.).


Then in chan_dahdi.conf use something like:

;BRI Module
group = 1
signalling = bri_cpe
context = incoming
channel => 1,2,4,5,7,8,10,11

Your chan_dahdi.conf has "group" and "context" multiple times and that
does not seem right (admittedly it's been ages since I setup a Digium card).

Hope this helps. If not follow the installation manual step for step or
call Digium support.

http://docs.digium.com/H8/hx8_series_manual.pdf

Regards,
Patrick

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[asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Bart Coninckx 

Hi,

I'm experiencing difficulties to get a B410P running with Asterisk 
10.3.1 and DAHDI 2.6.1.
Am I supposed to use DAHDI for this card and ISDN BRI for my country 
(Belgium)?


thx,

BC

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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread khalid touati 
Thank you for your answer,  I think I posted dhadi version and so but let
me add more details and recap them below:

We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
digium card 1HA8-0400BLF

output of dahdi_hardware: pci::04:08.0 wctdm24xxp+  d161:8008
HB8-

>From BT side: it is called by BT a "system access" ISDN2 BRI (per BT NT
mode and signaling as PTP)

chan_dahdi.conf
; Span 1: WCBRI/0/0 "HB8-" (MASTER) AMI/CCS
group=1,11
context= mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 1-2
context = default
group = 63

; Span 2: WCBRI/0/1 "HB8-" AMI/CCS
group=1,12
context=mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 4-5
context = default
group = 63

; Span 3: WCBRI/0/2 "HB8-" AMI/CCS
group=1,13
context=mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 7-8
context = default
group = 63

; Span 4: WCBRI/0/3 "HB8-" AMI/CCS
group=1,14
context=mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 10-11
context = default
group = 63

the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error in
state 7(Multi-frame established)
*
Thank you!!

On Wed, May 9, 2012 at 1:22 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09-05-12 18:46, khalid touati wrote:
> > Should I understand that no Asterisk user has issues with ISDN "system
> > access" configuration from UK? or maybe no one is using Asterisk In UK
> :) ?
>
> I have no idea. But other than the error you have given very little
> information to go on. Which card are you using, what type of ISDN line
> (PTP?), what's the DAHDI or Sangoma (or...) and Asterisk configuration,
> what do the log files say, etc?
>
> Have you tried calling the vendor of the ISDN card for support?
>
> Regards,
> Patrick
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread Patrick Lists 
On 09-05-12 18:46, khalid touati wrote:
> Should I understand that no Asterisk user has issues with ISDN "system
> access" configuration from UK? or maybe no one is using Asterisk In UK :) ?

I have no idea. But other than the error you have given very little
information to go on. Which card are you using, what type of ISDN line
(PTP?), what's the DAHDI or Sangoma (or...) and Asterisk configuration,
what do the log files say, etc?

Have you tried calling the vendor of the ISDN card for support?

Regards,
Patrick

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Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Ricardo Carvalho 
Problem SOLVED.

You'r right, this is a problem of codec mismatching. Activating sip debug i
can see it:

Capabilities: us - 0x802 (gsm|g726), peer - audio=0x10d
(g723|ulaw|alaw|g729)
[May 9 17:16:37] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
codecs, not accepting this offer!

I solved the problem thanks to your help! Since that SIP trunk isn't
authenticated, i just receive calls in the default context that is set in
sip.conf, and so, I don't set the codecs to be used. I discovered that the
problem was that i had one other peer defined in sip.conf that had the same
IP address set, so it was shuffling asterisk some how. Funny that the same
configuration wasn't a problem in asterisk 1.4, but in this 1.8 it caused
this problem.

Thank you onde again,

Regards,
Ricardo.



On Wed, May 9, 2012 at 5:10 PM, Andres  wrote:

> On 5/9/2012 11:56 AM, Ricardo Carvalho wrote:
>
>> That's weird, because it's negotiated with success the codec ulaw for
>> outbound calls through the same SIP trunk.
>>
>>  My guess is the incoming call is not being matched with the peer you are
> expecting.  Do a sip debug and watch the output to see what peer is being
> selected.
>
> Andres
>
>  Besides, ulaw and alaw shows up when i do "core show codecs audio" in the
>> asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules
>> under the path /usr/lib/asterisk/modules/
>>
>> I don't get it!...
>>
>> More ideas?
>>
>> Thanks,
>> Ricardo.
>>
>>
>>
>> On Wed, May 9, 2012 at 3:32 PM, A J Stiles 
>> > asterisk_l...@earthshod.co.uk>> wrote:
>>
>>On Wednesday 09 May 2012, Ricardo Carvalho wrote:
>>
>>> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No
>>compatible
>>> codecs, not accepting this offer!
>>>
>>> Any help?
>>
>>Are you sure you compiled all the codecs you need?
>>
>>What happens if you run `make menuselect` in both the 1.4 source
>>tree and in
>>the 1.8 source tree, "side-by-side" in tabs of the same terminal
>>window?  You
>>need at least GSM, A-law and micro-law.
>>
>>(The above is my preferred method of building a configuration like
>>an existing
>>one.  No doubt someone will weigh in with a better way of doing it.)
>>
>>--
>>AJS
>>
>>Answers come *after* questions.
>>
>>--
>>_
>>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
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Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-09 Thread khalid touati 
Should I understand that no Asterisk user has issues with ISDN "system
access" configuration from UK? or maybe no one is using Asterisk In UK :) ?

On Tue, May 8, 2012 at 12:46 PM, khalid touati wrote:

> Hi All,
> I am posting this thread with the hope that someone in UK (or elsewhere)
> had a similar issue:
> Our issue is simple, we cannnot use our ISDN line, when watching asterisk
> console it gives a bunch of ISDN errors where the following is probably the
> most relevant:
>
> Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established)
>
> We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. I can post
> configuration and/or further information if needed.
>
> --
> Khalid Touati
> Network Administrator
> CCNA
>
>
>


-- 
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CCNA
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Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Andres 

On 5/9/2012 11:56 AM, Ricardo Carvalho wrote:
That's weird, because it's negotiated with success the codec ulaw for 
outbound calls through the same SIP trunk.


My guess is the incoming call is not being matched with the peer you are 
expecting.  Do a sip debug and watch the output to see what peer is 
being selected.


Andres

Besides, ulaw and alaw shows up when i do "core show codecs audio" in 
the asterisk CLI, and there exists both codec_ulaw.so and 
codec_alaw.so modules under the path /usr/lib/asterisk/modules/


I don't get it!...

More ideas?

Thanks,
Ricardo.



On Wed, May 9, 2012 at 3:32 PM, A J Stiles 
mailto:asterisk_l...@earthshod.co.uk>> 
wrote:


On Wednesday 09 May 2012, Ricardo Carvalho wrote:

> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No
compatible
> codecs, not accepting this offer!
>
> Any help?

Are you sure you compiled all the codecs you need?

What happens if you run `make menuselect` in both the 1.4 source
tree and in
the 1.8 source tree, "side-by-side" in tabs of the same terminal
window?  You
need at least GSM, A-law and micro-law.

(The above is my preferred method of building a configuration like
an existing
one.  No doubt someone will weigh in with a better way of doing it.)

--
AJS

Answers come *after* questions.

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Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Eric Wieling 
Do a "sip show peer PEERNAME" and check the codecs allowed for that specific 
peer.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho
Sent: Wednesday, May 09, 2012 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No compatible codecs, not accepting this offer! - 
after upgrading to 1.8.11

That's weird, because it's negotiated with success the codec ulaw for outbound 
calls through the same SIP trunk.


Besides, ulaw and alaw shows up when i do "core show codecs audio" in the 
asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules 
under the path /usr/lib/asterisk/modules/

I don't get it!...

More ideas?

Thanks,
Ricardo.



On Wed, May 9, 2012 at 3:32 PM, A J Stiles  
wrote:


On Wednesday 09 May 2012, Ricardo Carvalho wrote:

> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No 
compatible
> codecs, not accepting this offer!
>
> Any help?


Are you sure you compiled all the codecs you need?

What happens if you run `make menuselect` in both the 1.4 source tree 
and in
the 1.8 source tree, "side-by-side" in tabs of the same terminal 
window?  You
need at least GSM, A-law and micro-law.

(The above is my preferred method of building a configuration like an 
existing
one.  No doubt someone will weigh in with a better way of doing it.)

--
AJS

Answers come *after* questions.

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Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Ricardo Carvalho 
That's weird, because it's negotiated with success the codec ulaw for
outbound calls through the same SIP trunk.

Besides, ulaw and alaw shows up when i do "core show codecs audio" in the
asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules
under the path /usr/lib/asterisk/modules/

I don't get it!...

More ideas?

Thanks,
Ricardo.



On Wed, May 9, 2012 at 3:32 PM, A J Stiles wrote:

> On Wednesday 09 May 2012, Ricardo Carvalho wrote:
>
> > [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
> > codecs, not accepting this offer!
> >
> > Any help?
>
> Are you sure you compiled all the codecs you need?
>
> What happens if you run `make menuselect` in both the 1.4 source tree and
> in
> the 1.8 source tree, "side-by-side" in tabs of the same terminal window?
>  You
> need at least GSM, A-law and micro-law.
>
> (The above is my preferred method of building a configuration like an
> existing
> one.  No doubt someone will weigh in with a better way of doing it.)
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread A J Stiles 
On Wednesday 09 May 2012, Ricardo Carvalho wrote:

> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
> codecs, not accepting this offer!
> 
> Any help?

Are you sure you compiled all the codecs you need?

What happens if you run `make menuselect` in both the 1.4 source tree and in 
the 1.8 source tree, "side-by-side" in tabs of the same terminal window?  You 
need at least GSM, A-law and micro-law.

(The above is my preferred method of building a configuration like an existing 
one.  No doubt someone will weigh in with a better way of doing it.)

-- 
AJS

Answers come *after* questions.

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[asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

2012-05-09 Thread Ricardo Carvalho 
Hi,

I've upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don't work anymore. I
don't know why!...

This is the SDP portion that comes in the INVITE messages of calls through
that trunk (let's say, whose endpoint has the IP x.x.x.x, purposely
omitted). Nothing seems to be wrong with that to me:
v=0
o=CSM 0 1 IN IP4 x.x.x.x
s=Acme
c=IN IP4 x.x.x.x
t=0 0
m=audio 22152 RTP/AVP 8 0 18 4 101
a=rtpmap:101 telephone-event/8000

And here's the debugging:
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP
to Off
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP o=CSM 0 1 IN IP4 x.x.x.x... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP s=Acme... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting 'x.x.x.x' into...
[May 8 17:45:30] DEBUG[6444]: netsock2.c:188 ast_sockaddr_split_hostport:
...host 'x.x.x.x' and port ''.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP c=IN IP4 x.x.x.x... OK.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP t=0 0... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:9110 process_sdp: Processing
media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x416e25b0
[May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
codecs, not accepting this offer!


Any help?

Thanks,
Ricardo.
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Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

2012-05-09 Thread Bryant Zimmerman 
I agree with Chad on this one. FreePBX is for a different kind of installs 
from what you described. I believe if you pitch it as your solution; as you 
have stated you will be disappointing. Before you start offering it up to 
the bosses as the silver bullet. I would do a test install on some spare 
hardware (You can use VM Ware or Hyper-V if you know what you are doing). 
Other wise you might be the one being shot with the silver bullet. We have 
used free PBX and it works great as long as your install needs are 
standard, but if you go beyond the standard you need something more. We are 
a sip trunk provider and hosted PBX provider, and we have wholesalers that 
try to use FreePBX to deliver customer sip trunks and they have issues.. It 
is the square peg round hole issue. Get the right shape peg  for your 
communications hole..

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 


 From: "Chad Wallace" 
Sent: Tuesday, May 08, 2012 8:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on 
my new architecture.

On Mon, 07 May 2012 06:40:46 -0600
Nunya Biznatch  wrote:

> >> I need to plan to use FreePBX on all Asterisk Servers, but I don't
> >> intend to install it until I'm in regular MAC maintenance mode.
> >>
> > It is ashame you are going this far with your setup to rely on 
> > FreePBX.  For something this complex, you are setting your self up
> > for some heartache.
> >
> It is my intention to do everything from the command line. However, 
> there will be times when I'll have Interns coming in and doing some
> of the MAC activities, and I thought this might be an easier way for
> the day to day to get done. I've never seen it myself either, so am
> curious. Finally, there's the "glitter" factor. When my bosses come
> in and want a dog and pony show on the new phone system, they want to
> see fluffy bunnies and kittens, not the Ox that's doing the pulling.
> CLI = "old" in the minds of those that don't comprehend.

I just installed FreePBX, and I'm pretty sure that "do everything from
the command line" and "use FreePBX" are mutually exclusive situations.

When you install FreePBX, it replaces (overwrites!) your config, and
then you manually enter everything (devices, queues, IVR, etc.) in the
GUI--or use a bulk import/export tool if there is one.  It takes over
your Asterisk install, and you have to adapt to it, not the other way
around.

If you want to set it up with standard Asterisk configs and your own
dialplan, you'll need to find another way to do the GUI for the
noobs/interns.

For the bosses, I would suggest sending them to an actual dog and pony
show instead.  But that's just me. ;-)

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0

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