Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Lenz Emilitri]

Just can't wait for the live calorie counter! :)
l.


2010/4/1 Olle E. Johansson 

> FOR IMMEDIATE RELEASE
> Puerto Escondido, Mexico, April 1st, 2010:
>
> Digium launches Asterisk VCC (TM) - a new virtual communication platform
> for enterprises, the public sector and the home.
> ===
>
>  For VoxSwitch customers, VCCnet will mean that every user can monitor
> the movement of coworkers in realtime. By using the new APIs, additional
> data like credit card transactions, fuel consumption in the car, mileage
> in the air and calories eaten can be reported with a 3D graphical display
> using HTML5.
>

-- 
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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Randy R]

We were hoping voicemail would become Tweets and that Tweets from your
bathroom scale could be sent as audio using calls files. I guess that
will be the next minor version?

/r

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Re: [asterisk-users] Necessary hardware

[Ioan Indreias]

Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
other ATA which have FXO ports).

HTH,
Ioan.

On Thu, Apr 1, 2010 at 12:29 AM, Kosa  wrote:

> I have two linksys spa2102 and a sap9000 but as far as I know I need
> something else to connect the asterisk box to the analog phoneline.

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[asterisk-users] OfficeSIP Communications Makes Its VoIP SIP Products Open Source

[Vitali Fomine]

Press Release

For Immediate Release

OfficeSIP Communications
http://www.officesip.com/
i...@officesip.com

OfficeSIP Communications Makes Its VoIP SIP Products Open Source

OfficeSIP Communications makes its two enterprise VoIP SIP clients
officially open-source. OfficeSIP Softphone and OfficeSIP Messenger are now
publicly available, and their source code published under the GPL license.
The two products complete with the source code are available for immediate
download at the company's Web site, officesip.org.

OfficeSIP Communications is committed to continuous development of both SIP
clients, and invites developers to join the project. The company believes
that opening the source code to the community will benefit the development
of the project, and will help it gain trust and popularity among its users.

About OfficeSIP Softphone and OfficeSIP Messenger

The two VoIP applications enable users to communicate via the
industry-standard SIP protocol. OfficeSIP Softphone is a simple,
straightforward SIP client enabling voice and video communications, while
OfficeSIP Messenger offers enterprise customers the ability to communicate
via text, voice and video chats for free. Compatible with Office
Communications Server 2007, OfficeSIP Messenger delivers reliable
performance combined with trouble-free deployment and management. OfficeSIP
Messenger implements ICE, STUN, and TURN protocols to seamlessly traverse
NAT and firewalls, and supports secure communications via the TLS protocol.

OfficeSIP Softphone and OfficeSIP Messenger are written in C# in .NET
framework. The two applications make use of Microsoft Unified Communications
Client API SDK, ensuring the highest quality of audio and video
communications. The use of underlying Microsoft platform ensures the
greatest level of compatibility with a wide range of hardware devices such
as webcams. OfficeSIP Softphone and OfficeSIP Messenger have been
extensively tested, and offer the complete SIP functionality.

About OfficeSIP Communications

Established in 2007, OfficeSIP Communications has been developing
open-source instant messaging and VoIP solutions for enterprises. The
company established solid reputation among its customers, and gained
expertise in meeting the communication needs of its corporate customers.

# # #

OfficeSIP Softphone and OfficeSIP Messenger along with their source code are
available under the GPL license at http://www.officesip.org/



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Re: [asterisk-users] Asterisk load balancing and failover

[Ngo-Vi Hoai-Anh]

I'm not quite sure what do you mean with MSC.

Anyway, I assume your environment is like

[PSTN (Public Switched Telephone Network)]<-->[DTM 
Switch]<---SS7  (PRI line)>[Asterisk 
Box] IP net

If you mean MSC Mobile Switching Center it could look like
[GSM Network]<->[MSC]<- SS7 
>[Asterisk 
Box]<-VoIP-->IP net

Normally, the DTM Switch or MSC should be configurable for 
load-balancing and failover.

Point code is for SS7 networking like IP address for IP networking.

huu giang schrieb:
> Do you mean that SS7 switch is a MSC and do all MSC support load 
> balancing without any hardware between it and my Server.
>
> Sorry for my English, what do you mean two point codes for my servers 
> ?. I have at least two servers.
>
>
> --- On *Wed, 3/31/10, Tobias Wolf //* wrote:
>
>
> From: Tobias Wolf 
> Subject: Re: [asterisk-users] Asterisk load balancing and failover
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Date: Wednesday, March 31, 2010, 4:27 AM
>
> huu giang schrieb:
> > Hi Zeeshan
> >
> > I know a solution using DRBD, Heartbeat and RedFone hardware to
> > provide failover ability to Asterisk.
> >
> > If I have two Asterisk Servers, and each server has a TDM card
> and a
> > PRI line connect to each card, how your solution can provide
> failover
> > ability to Asterisk ? Do you need any other hardware?
> >
> > The calles to my IVR System don't just come from IP network
> (SIP) but
> > can come from SS7 network.
> >
> Well, if that case the SS7 Switch to which you are connected
> should be
> able to load balance the call to both of your servers. I guess you
> have
> two point codes for you servers? If one server goes down, the ss7
> switch
> received the red alarms and
> stops to route calls to it. Once the server is up again it will
> get new
> calls.
>
> So, we only thing you have to worry about is to keep state
> information
> between the two servers consistent if people record messages or
> access
> databases.
>
> Regards,
>
> Tobias
> >
> > Thanks.
> >
> >
> >
> >
> > --- On *Fri, 3/26/10, Zeeshan Zakaria / >/* wrote:
> >
> >
> > From: Zeeshan Zakaria  >
> > Subject: Re: [asterisk-users] Asterisk load balancing and
> failover
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >  >
> > Date: Friday, March 26, 2010, 1:51 AM
> >
> > About two years ago I setup two high availability solutions
> using
> > DRBD and Heartbeat. The worked great and shutting down or
> > unplugging one server stayed transparent for the callers, as
> IVRs
> > stayed available. Having said this, it was not very straight
> > forward to set it up, but not very difficut either. So Heartbeat
> > and DRBD can be a good starting point for you.
> >
> > --
> > Zeeshan A Zakaria
> >
> >> On 2010-03-26 4:40 AM, "huu giang"  
> >>  >> wrote:
> >>
> >> Hi List,
> >>
> >> I'm finding a solution to provide failover and load balancing
> >> features to my IVR system.
> >>
> >> Anyone suggest me what is the best solution please?. what the
> >> hardware I should use ?.
> >>
> >> I heard about RedFone, but someone on the mail list said
> that it
> >> is not good because *TDMoE* module in asterisk is not so
> *stable*
> >> and TDMoE is stale. And It seems that RedFone doesn't not
> support
> >> load balancing ability (I can't find any document about this
> >> feature).
> >>
> >> Best Regards,
> >> Giang Huu.
> >>
> >>
> >>
> >>
> >> --
> >> 
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> >>
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> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > -Inline Attachment Follows-
> >
> > --
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> >
> >

[asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver

[Jorge Churio]

Redfone uses and improved, in house developed TDMoE driver, officially
supported by same Redfone.
Redfone´s support site maintains tdmoe driver updated and "certified" to
operate in every zaptel and dahdi versions.
Txs

Jorge Churio
Redfone Communications


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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Tommy Botten Jensen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512

You know there are 1st of april jokes, and there are evil 1st of april
jokes.

... I actually felt a bit nauseous

- - Tommy


Den 01. april 2010 07:53, skrev Olle E. Johansson:
> FOR IMMEDIATE RELEASE
> Puerto Escondido, Mexico, April 1st, 2010:
> 
> Digium launches Asterisk VCC (TM) - a new virtual communication platform
> for enterprises, the public sector and the home.
> ===
> 
> Asterisk 1.8 will contain a stunning new technology for all Asterisk users 
> world-
> wide - virtual communication clouds or VCC (TM).  With this technology, call 
> handling will never be the same. In one move, the Asterisk development team 
> leaves the old world of PBX call switching behind and moves the enterprise 
> telephony server to the cloud. 
> 
> By combining IPv6, the 3G cell network and cloud services with existing 
> Asterisk
> technologies  like Dundi and IAX2, Digium moves into the era of cloud 
> computing. 
> The launch includes end-user applications powered by cloud services 
> - moving Digium technology to the palm of your hand.
> 
> - "Our new platform is built for the new organization in the workplace, the 
> family
> and the community - a truly virtual multimedia communication network for the
> Internet age. By moving our focus away from the traditional PBX, we succeeded
> in changing the  Digium solution from a server centric view to a service 
> centric view." 
> says Sokkie Stevens, product manager for the new platform.
> 
> The first step was to transform Digium into a virtual service provider. Digium
> is one of the first companies to get an IPv6 assignment on a global service
> provider level. After signing peering agreements with major carriers world-
> wide, the next step was to apply the successful Dundi protocol on top of
> IPv6. 
> 
> -"Dundi and IPv6 was a match made in heaven", says Mick Spenser,
> the CTO for Digium, "Dundi had a successful peering and discovery
> infrastructure that is now even stronger with IPv6 multicast and secured
> by using IPsec."
> 
> VCC will be a binary module distributed with Asterisk 1.8. It will connect
> to the Digium VCCnet over native IPv6, IPv6 over IPv4 tunnels and directly
> over layer 2 technologies like Ethernet. All VCC clients will get a native
> IPv6 address assigned. Enterprises may purchase a full IPv6 network range
> in the VCCnet to get full access. VCCnet is a network service managed
> by Digium worldwide.
> 
> VCCnet will enable automatic follow-me functionality. When you turn
> on your VCC-enabled smartphone, the VCCnet client will automatically report 
> your
> location (from 3G cells or GPS) back to the Asterisk service. Your status
> will be automatically updated as you move between networks, from
> WiFi in the office to 3G on the road. One person can have multiple
> VCC clients - one supporting video, another old-fashioned audio
> and a third HD audio and video. The new IAX3 protocol used in VCCnet
> will automatically negiotiate media capabilities and select the right client
> for the right call, depending on privacy settings and personal preferences.
> 
> For VoxSwitch customers, VCCnet will mean that every user can monitor
> the movement of coworkers in realtime. By using the new APIs, additional
> data like credit card transactions, fuel consumption in the car, mileage
> in the air and calories eaten can be reported with a 3D graphical display
> using HTML5.
> 
> As an additional service in the VCCnet cloud, Digium will offer extended
> capacity for your telecommunications platform. When you need more capacity
> for video calls, 5+1 hd voice conferences and other coming services, including
> 3D multimedia conferencing, your existing PBX will be virtually extended by 
> using
> resources available (and unused) in the cloud. For the system manager, it 
> will 
> look like all these services are produced locally, just like before.
> 
> VCC includes clients for all popular platforms, including the soon to be 
> released
> Apple iPAD. "Many people was asking us for the Digium Phone, but it felt very
> wrong to implement an old-fashioned device on top of a modern communication
> network" says Mike Spenser. "The client will be a natural part of the 
> personal computing
> infrastructure that already exists out there. It will be the personal 
> communication
> exchange, the Facebook of the multimedia realtime communications world." 
> 
> Digium will rename the recently launched Asterisk marketplace to 
> VCCstore and  use that infrastructure for distribution of the VCCblocks 
> - applets that enhance your virtual communication cloud. 3rd party developers 
> may apply for development kits and distribution agreements. Digium is 
> currently 
> negotiating the rights to distribute audio books and radio shows for the new 
> culture-on-hold service while not using the VCCclient for two- or multiparty 
> communication.
> 
> While testing, the mo

[asterisk-users] Problem with Sangoma A104 and euroisdn pri

[Jaap Winius]

Hi all,

My problem boils down to these errors:

... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time

This is triggered by lines in extentions.conf such as:

exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W)

The system is CentOS v5.2 with Asterisk 1.4.23  
(druid-asterisk-1.4.23.1-2), a Sangoma A104 4-port card, Wanpipe  
v3.4.4 and Zaptel v1.4.12.1. The system is attached to a single  
EuroISDN PRI and is located in the Netherlands.

Besides the above error, I also noticed this:

CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3

The status needs to be "Provisioned, Up, Active."

Following Sangoma's instructions for debugging an Asterisk PRI span, I  
can confirm that there are only outgoing frames and that the D-channel  
messages in Asterisk are the same as what the Wanpipe drivers are  
seeing. So, assuming that my local telco (KPN Telecom) has activated  
the D-channel, what else could possibly be causing this problem?

Thanks,

Jaap

PS -- Below are my current configuration files and debugging output:

==begin zaptel.conf 

loadzone=us
defaultzone=us
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
hardhdlc=16

==end zaptel.conf ==

==begin wanpipe1.conf ==

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS  = 13
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= NCRC4
FE_LINE= 1
TE_CLOCK = NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE= NO
MTU = 1500
UDPPORT = 9000
TTL= 255
IGNORE_FRONT_END = NO
TDMV_SPAN= 1
TDMV_DCHAN= 16
TDMV_HW_DTMF= NO
TDMV_HW_FAX_DETECT = NO

[w1g1]
ACTIVE_CH= ALL
TDMV_HWEC= NO

==end wanpipe1.conf 

==begin zapata.conf 

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

switchtype=euroisdn
context=default
group=1
signalling=pri_cpe
channel =>1-15,17-31

==end zapata.conf ==

Here's some debugging output:

=== begin debug info ==

# ztcfg -vv

Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: Hardware assisted D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)

31 channels to configure.

# wanrouter status

Devices currently active:
 wanpipe1


Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |  
Baud rate |
wanpipe1| N/A  | A101/1D/A102/2D/4/4D/8| 169 | 4   | 1  
| N/

Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Gordon Henderson]

On Thu, 1 Apr 2010, Tommy Botten Jensen wrote:

> You know there are 1st of april jokes, and there are evil 1st of april
> jokes.

> ... I actually felt a bit nauseous

Thought it was funny myself! Not as funny as the other one just posted 
about the VoIP clients written in C# and .net under Microsoft though. Now 
that was funny!

Gordon

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[asterisk-users] RPID on called party

[Ondrej Valousek]

Hello,

Did anyone manage to force asterisk to put Remote-party-ID attribute on 
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of 
B displayed on his phone.
Note that name of A gets displayed on the B's phone fine, but this is 
not what I want.
This works with Cisco Call manager fine - the RPID is sent as a part of 
the response to the SIP INVITE this way:


SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport
From: "Ondrej Valousek"   
;tag=as4786d518
To:   
;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104
Date: Tue, 30 Mar 2010 13:53:15 GMT
Call-ID: 465a9c200587260d164f451409489...@192.168.60.20
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
Allow-Events: presence
*Remote-Party-ID: "Paul Ryan"   
;party=called;screen=yes;privacy=off*
Contact:   
Content-Length: 0


But I can not make it working with Asterisk. Does anyone have any glue 
how to achieve this WITHOUT patching asterisk? I am happy to upgrade to 
the latest/greatest version, I just do not want to patch.
Many thanks,

Ondrej

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Re: [asterisk-users] RPID on called party

[Juan E. Rodríguez]

Try using sendrpid=yes on sip.conf

Regards,
Juan

Ondrej Valousek wrote:
> Hello,
>
> Did anyone manage to force asterisk to put Remote-party-ID attribute on 
> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of 
> B displayed on his phone.
> Note that name of A gets displayed on the B's phone fine, but this is 
> not what I want.
> This works with Cisco Call manager fine - the RPID is sent as a part of 
> the response to the SIP INVITE this way:
>
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport
> From: "Ondrej Valousek"   
> ;tag=as4786d518
> To:   
> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104
> Date: Tue, 30 Mar 2010 13:53:15 GMT
> Call-ID: 465a9c200587260d164f451409489...@192.168.60.20
> CSeq: 102 INVITE
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
> SUBSCRIBE, NOTIFY
> Allow-Events: presence
> *Remote-Party-ID: "Paul Ryan"  
>  ;party=called;screen=yes;privacy=off*
> Contact:   
> Content-Length: 0
>
>
> But I can not make it working with Asterisk. Does anyone have any glue 
> how to achieve this WITHOUT patching asterisk? I am happy to upgrade to 
> the latest/greatest version, I just do not want to patch.
> Many thanks,
>
> Ondrej
>
>   

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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Vitali Fomine]

> Thought it was funny myself! Not as funny as the other one just posted
> about the VoIP clients written in C# and .net under Microsoft though. Now
> that was funny!

It was not a joke. What is wrong with c#,.net?

Best regards,
Vitali Fomine 


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Re: [asterisk-users] RPID on called party

[Mark Michelson]

Ondrej Valousek wrote:
> Hello,
> 
> Did anyone manage to force asterisk to put Remote-party-ID attribute on 
> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of 
> B displayed on his phone.
> Note that name of A gets displayed on the B's phone fine, but this is 
> not what I want.
> This works with Cisco Call manager fine - the RPID is sent as a part of 
> the response to the SIP INVITE this way:
> 
> 
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport
> From: "Ondrej Valousek"   
> ;tag=as4786d518
> To:   
> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104
> Date: Tue, 30 Mar 2010 13:53:15 GMT
> Call-ID: 465a9c200587260d164f451409489...@192.168.60.20
> CSeq: 102 INVITE
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
> SUBSCRIBE, NOTIFY
> Allow-Events: presence
> *Remote-Party-ID: "Paul Ryan"  
>  ;party=called;screen=yes;privacy=off*
> Contact:   
> Content-Length: 0
> 
> 
> But I can not make it working with Asterisk. Does anyone have any glue 
> how to achieve this WITHOUT patching asterisk? I am happy to upgrade to 
> the latest/greatest version, I just do not want to patch.
> Many thanks,
> 
> Ondrej
> 

This feature is in Asterisk trunk and will be present in the upcoming 1.8 
release. By setting sendrpid=yes on A's phone, Asterisk will send a 
Remote-Party-ID header that corresponds to what Asterisk received from B. Also, 
there is a CONNECTEDLINE() dialplan function that can be used to send this 
information prior to a call. I actually gave a presentation on this topic at 
Astricon last year, but for some reason the Astricon '09 archive does not seem 
to have my presentation video available.

Mark Michelson

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Re: [asterisk-users] Necessary hardware

[Kosa]

I found a sap400, wich is FXO and seems to work fine with asterisk. It
offers the posibility to plug 4 analog phonelines.

Thanks again.

Kosa

- Un mundo mejor es posible -

Ioan Indreias escribió:
> Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
> other ATA which have FXO ports).
> 
> HTH,
> Ioan.
> 
> On Thu, Apr 1, 2010 at 12:29 AM, Kosa  wrote:
> 
>> I have two linksys spa2102 and a sap9000 but as far as I know I need
>> something else to connect the asterisk box to the analog phoneline.
> 


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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Tzafrir Cohen]

On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA512
> 
> You know there are 1st of april jokes, and there are evil 1st of april
> jokes.

Huh? Have you actually bothered reading the site it points to?

The date there clearly states that it is:

  Press Release (5 April 2010)

Surely that's not a April Fool's day joke.

(Earlier this morning the date there was 10 April)

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Re: [asterisk-users] Polycom not updating the directory list

[hin lee]

Figured out my issue.  My contacts are in  -directory.cfg when it 
should be in -directory.xml.  

When did Polycom switched from CFG to XML?





From: hin lee 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Fri, March 19, 2010 12:00:14 PM
Subject: Re: [asterisk-users] Polycom not updating the directory list


The -directory.cfg  permission is 777 with a symoblic link pointing 
to -directory.xml with permission of 644.  

I would manually edit the -directory.xml and make the changes 
needed.  Upon rebooting the phone, the directory is still show the old 
contacts.  Somehow the phone did not pull the new contacts directory.  If I 
format the phone file system, then it will reflect the new contacts.





From: "Lee, John (Sydney)" 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Wed, March 17, 2010 11:05:09 PM
Subject: Re: [asterisk-users] Polycom not updating the directory list

The very obvious thing to check is the permission of the 
-directory.cfg.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee
Sent: Thursday, 18 March 2010 4:56 PM
To: Asterisk Users
Subject: Re: [asterisk-users] Polycom not updating the directory list

anyone?


From: hin lee 
To: Asterisk Users 
Sent: Fri, March 12, 2010 10:08:53 AM
Subject: Polycom not updating the directory list
Hi,

I have a strange problem with all of our Polycom 550 & 650 phones.  I am 
running a TFTP server on my Asterisk server and option 66 Boot Host pointing to 
Asterisk on my DHCP server.  The auto-provisioning is working because the 
phones are registering correctly with their extension.  If I change the MAC.cfg 
file to another extension and reboot the phone, it will reflect the new ext.  

The part that doesn't work is the -directory.cfg.  If I make an update to 
this file and reboot the phones, they do not reflect the new directory list.  
The only way I was able to get the phone to see the new directory list was to 
"Format" the phone.  Of course this is not the ideal way.  Also to add, the 
-directory.cfg files point to 0-directory.xml.  This way I 
only have one file to maintain.

Anyone knows why it's not pull the new -directory.cfg file.


Thank you!



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Re: [asterisk-users] Reset personal voicemail settings

[Felix Tiefenthaler]

Hi,

thank you very much. this solved my problem.

greets

felix


Am 31.03.2010 um 22:52 schrieb Mark Michelson:

> Felix Tiefenthaler wrote:
>> Hi list,
>>
>> can anyone tell me how to reset/delete all modifications (personal
>> greeting message, personal name, ...) I made in my voicemail?
>> I just want to get the default automatic computer messages back.
>>
>> thank you!
>>
>> greets
>> felix
>>
>
> If you are storing voicemail on the file system, then you can just  
> go to
> /var/spool/asterisk/voicemail/// and delete the  
> items in there
> that you want to. The "INBOX" and "Old" folders contain new and old  
> messages.
> Anything else in there will be greetings and other similar recordings.
>
> Mark Michelson
>
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[asterisk-users] Exceptionally long voice queue length errors...

[James Lamanna]

Hi,
I'm seeing a lot of "Exceptionally long voice queue length" errors in
my logs, and then I seem to have a problem
where Asterisk will drop the registration for a significant number of
phones (they go UNREACHABLE), but then they
come back approximately a minute later.
Is this some sort of load problem? Or something else?

Thank you.

-- James

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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Tilghman Lesher]

On Thursday 01 April 2010 10:36:23 Vitali Fomine wrote:
> > Thought it was funny myself! Not as funny as the other one just posted
> > about the VoIP clients written in C# and .net under Microsoft though. Now
> > that was funny!
>
> It was not a joke. What is wrong with c#,.net?

You probably should have thought twice about announcing it on this list (as
opposed to the -biz list) on April 1st, then.  We specifically have avoided
announcing new releases or security advisories on this date for that exact
reason.

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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[--[ UxBoD ]--]

- "Tzafrir Cohen"  wrote:

> On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA512
> > 
> > You know there are 1st of april jokes, and there are evil 1st of
> april
> > jokes.
> 
> Huh? Have you actually bothered reading the site it points to?
> 
> The date there clearly states that it is:
> 
>   Press Release (5 April 2010)
> 
> Surely that's not a April Fool's day joke.
> 
> (Earlier this morning the date there was 10 April)
> 
> -- 
>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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>http://www.asterisk.org/hello
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

Please send emails about this, as instructed, to loofli...@digium.com or 
perhaps just reverse the letters before the domain name ;)
-- 
Thanks, Phil

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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Tzafrir Cohen]

On Thu, Apr 01, 2010 at 07:05:07PM +0100, --[ UxBoD ]-- wrote:
> - "Tzafrir Cohen"  wrote:
> 
> > On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> > > -BEGIN PGP SIGNED MESSAGE-
> > > Hash: SHA512
> > > 
> > > You know there are 1st of april jokes, and there are evil 1st of
> > april
> > > jokes.
> > 
> > Huh? Have you actually bothered reading the site it points to?
> > 
> > The date there clearly states that it is:
> > 
> >   Press Release (5 April 2010)
> > 
> > Surely that's not a April Fool's day joke.
> > 
> > (Earlier this morning the date there was 10 April)


> Please send emails about this, as instructed, to loofli...@digium.com or 
> perhaps just reverse the letters before the domain name ;)

I was referring to http://officesip.org/

Now, gro.piseci...@ofni does sound odd :-)

(As with Olle: I'm not related to that company in any way)

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Re: [asterisk-users] Press release: Virtual Communication Clouds:: New feature in Asterisk 1.8

[Danny Nicholas]

Back to your original point, this is NOT the day to post a real news item;
we all know that's Friday after 4:30 EST.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Thursday, April 01, 2010 1:28 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Press release: Virtual Communication Clouds::
New feature in Asterisk 1.8

On Thu, Apr 01, 2010 at 07:05:07PM +0100, --[ UxBoD ]-- wrote:
> - "Tzafrir Cohen"  wrote:
> 
> > On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> > > -BEGIN PGP SIGNED MESSAGE-
> > > Hash: SHA512
> > > 
> > > You know there are 1st of april jokes, and there are evil 1st of
> > april
> > > jokes.
> > 
> > Huh? Have you actually bothered reading the site it points to?
> > 
> > The date there clearly states that it is:
> > 
> >   Press Release (5 April 2010)
> > 
> > Surely that's not a April Fool's day joke.
> > 
> > (Earlier this morning the date there was 10 April)


> Please send emails about this, as instructed, to loofli...@digium.com or
perhaps just reverse the letters before the domain name ;)

I was referring to http://officesip.org/

Now, gro.piseci...@ofni does sound odd :-)

(As with Olle: I'm not related to that company in any way)

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[asterisk-users] problem compiling asterisk with cdr_odbc

[Nathan Pryor]

"make menuconfig" does not show cdr_odbc as a selectable compile option. I
have compiled and installed both unixODBC and freetds from source and have
verified both successfully connect to my sql server. Both were installed to
standard locations (/usr/lib). I had no problem compiling cdr_odbc on my
test server(CentOS 4.6), however following the same steps on my production
server (CentOS 5.4) gives no joy.

asterisk-1.6.2.0
unixODBC-2.2.14-p2
freetds-0.82

Any help is greatly appreciated.
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Re: [asterisk-users] problem compiling asterisk with cdr_odbc

[Tilghman Lesher]

On Thursday 01 April 2010 13:36:00 Nathan Pryor wrote:
> "make menuconfig" does not show cdr_odbc as a selectable compile option. I
> have compiled and installed both unixODBC and freetds from source and have
> verified both successfully connect to my sql server. Both were installed to
> standard locations (/usr/lib). I had no problem compiling cdr_odbc on my
> test server(CentOS 4.6), however following the same steps on my production
> server (CentOS 5.4) gives no joy.
>
> asterisk-1.6.2.0
> unixODBC-2.2.14-p2
> freetds-0.82
>
> Any help is greatly appreciated.

Did you remember to run ./configure AFTER installing the above packages?

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[asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM

[Chris Miller]

It seems that asterisk-addons and one or more of Digium's licensed 
modules such as res_fax_digium have a conflict that doesn't seem to 
be documented anywhere I can find.

In a nutshell, asterisk14-addons-core has a fake provide for 
asterisk-gplonly :

#
#  core subpackage
#
%package core
Summary: Asterisk-addons core package.
Group: Utilities/System
Provides: asterisk-gplonly
Provides: asterisk-addons-core
Obsoletes: asterisk-addons-core
Requires: asterisk14-core


The Digium licensed packages look for this package and prevent 
installation :

---> Package asterisk14-res_fax.i386 1:1.4_1.0.14-1_centos5 set to 
be updated
---> Package asterisk14-res_fax_digium.i386 1:1.4_1.0.11-1_centos5 
set to be updated
--> Processing Conflict: asterisk14-res_fax conflicts asterisk-gplonly
--> Processing Conflict: asterisk14-res_fax_digium conflicts 
asterisk-gplonly
--> Finished Dependency Resolution
1:asterisk14-res_fax_digium-1.4_1.0.11-1_centos5.i386 from 
digium-current has depsolving problems
   --> asterisk14-res_fax_digium conflicts with asterisk14-addons-core
1:asterisk14-res_fax-1.4_1.0.14-1_centos5.i386 from digium-current 
has depsolving problems
   --> asterisk14-res_fax conflicts with asterisk14-addons-core
Error: asterisk14-res_fax conflicts with asterisk14-addons-core
Error: asterisk14-res_fax_digium conflicts with asterisk14-addons-core


A comment in the spec file would have been nice... Does anyone know 
if this a real technical issue, or simply a licensing conflict 
between GPL and Digium?


Chris

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Re: [asterisk-users] problem compiling asterisk with cdr_odbc

[Nathan Pryor]

Yes, after installation of the drivers I've tried:

./configure
./configure --with-unixodbc=/usr
./configure --with-unixodbc=/usr/lib

None turn on cdr_odbc. I'm at a loss here. Is there anything in the
configure output I should be looking for?


On Thu, Apr 1, 2010 at 2:10 PM, Tilghman Lesher  wrote:

> On Thursday 01 April 2010 13:36:00 Nathan Pryor wrote:
> > "make menuconfig" does not show cdr_odbc as a selectable compile option.
> I
> > have compiled and installed both unixODBC and freetds from source and
> have
> > verified both successfully connect to my sql server. Both were installed
> to
> > standard locations (/usr/lib). I had no problem compiling cdr_odbc on my
> > test server(CentOS 4.6), however following the same steps on my
> production
> > server (CentOS 5.4) gives no joy.
> >
> > asterisk-1.6.2.0
> > unixODBC-2.2.14-p2
> > freetds-0.82
> >
> > Any help is greatly appreciated.
>
> Did you remember to run ./configure AFTER installing the above packages?
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] problem compiling asterisk with cdr_odbc

[Jared Smith]

- "Nathan Pryor"  wrote: 


"make menuconfig" does not show cdr_odbc as a selectable compile option. I have 
compiled and installed both unixODBC and freetds from source and have verified 
both successfully connect to my sql server. Both were installed to standard 
locations (/usr/lib). I had no problem compiling cdr_odbc on my test 
server(CentOS 4.6), however following the same steps on my production server 
(CentOS 5.4) gives no joy. 
Install the 'libtool-ltdl' and 'libtool-ltdl-devel' packages, and then re-run 
./configure. 

-- 
Jared Smith 
Digium, Inc. 
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Re: [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM

[Kevin P. Fleming]

Chris Miller wrote:

> A comment in the spec file would have been nice... Does anyone know 
> if this a real technical issue, or simply a licensing conflict 
> between GPL and Digium?

It is not a technical issue; it is an issue because some of the modules
in -addons have licenses that are pure GPLv2 only, and in addition the
license for MySQL-based components restricts their usage to *only*
GPLv2-licensed applications unless a commercial license for MySQL is
obtained. Since loading one of Digium's binary modules into an Asterisk
process changes it to no longer be pure GPLv2, such usage restrictions
should be taken into account.

The purpose of that conflict is to ensure that the person installing the
packages is made aware of the issue and that they must take explicit
action to override it (thus ensuring that we don't facilitate accidental
violation of third-party license agreements).

If you can suggest a method to provide this information to people in
some automatic way when they are made aware of the conflict by RPM, feel
free to do so and we'll try to get it incorporated into the RPMs themselves.

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[asterisk-users] Question about MaxRetries in the Asterisk Outgoing folder

[John Timms]

I'm doing some automated calling by putting .call files in the Outgoing
folder of Asterisk. I'm concerned this might be a stupid question, but I'm
pretty sure I've done my research well and I'm unable to come up with an
answer on my own.

I want to know: what happens to the .call files after the "MaxRetries"
number has been reached?

In my experience, they stay in the Outgoing folder, but are never deleted.
Instead, Asterisk keeps processing them, but never actually making a call.
In my mind, once the MaxRetries number has been met, Asterisk should do
something to get rid of the files, whether moving them to another "failed"
folder or just deleting them. You can see an example of my problem below.
The Yellow Highlighted remarks are my own for clarification and are not in
the actual .call file.

--
Channel: SIP/8644161...@vitel-outbound
MaxRetries: 9  <= Set to retry 9 times
RetryTime: 120 <= Retrys after 120 seconds
Context: autodial
Extension: s
Priority: 1
CallerID: 8645553190
Set: USERNUMBER=8644161809-JohnTimms
Set: DIGITS=8644161809
   <=
I've tried the "Archive: Yes" option here, but had no change in behavior
StartRetry: 2397 1 (1270149233)<= This line &
the following are all added by Asterisk

EndRetry: 2397 1 (1270149158)

StartRetry: 2397 2 (1270149399)

EndRetry: 2397 2 (1270149324)

StartRetry: 2397 3 (1270149565)

EndRetry: 2397 3 (1270149490)

StartRetry: 2397 4 (1270149731)

EndRetry: 2397 4 (1270149656)

StartRetry: 2397 5 (1270149897)

EndRetry: 2397 5 (1270149822)

StartRetry: 2397 6 (1270150063)

EndRetry: 2397 6 (1270149988)

StartRetry: 2397 7 (1270150229)

EndRetry: 2397 7 (1270150154)

StartRetry: 2397 8 (1270150395)

EndRetry: 2397 8 (1270150320)

StartRetry: 2397 9 (1270150561)

DelayedRetry: 2397 8 (1270151821)

DelayedRetry: 2397 8 (1270151942)

DelayedRetry: 2397 8 (1270152063)

DelayedRetry: 2397 8 (1270152184)

DelayedRetry: 2397 8 (1270152305)

DelayedRetry: 2397 8 (1270152426)

DelayedRetry: 2397 8 (1270152547)

DelayedRetry: 2397 8 (1270152668)

DelayedRetry: 2397 8 (1270152789)

DelayedRetry: 2397 8 (1270152910)

DelayedRetry: 2397 8 (1270153031)

DelayedRetry: 2397 8 (1270153152)

DelayedRetry: 2397 8 (1270153273)

DelayedRetry: 2397 8 (1270153394)

DelayedRetry: 2397 8 (1270153515)

DelayedRetry: 2397 8 (1270153636)

--

If anyone can help me out, that would be much appreciated.

--
John Timms
(864) 416-1809
johngti...@gmail.com
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Re: [asterisk-users] problem compiling asterisk with cdr_odbc

[Nathan Pryor]

I was missing libtool-ltdl-devel. Thanks!

On Thu, Apr 1, 2010 at 2:48 PM, Jared Smith  wrote:

> - "Nathan Pryor"  wrote:
>
> "make menuconfig" does not show cdr_odbc as a selectable compile option. I
> have compiled and installed both unixODBC and freetds from source and have
> verified both successfully connect to my sql server. Both were installed to
> standard locations (/usr/lib). I had no problem compiling cdr_odbc on my
> test server(CentOS 4.6), however following the same steps on my production
> server (CentOS 5.4) gives no joy.
>
> Install the 'libtool-ltdl' and 'libtool-ltdl-devel' packages, and then
> re-run ./configure.
>
> --
> Jared Smith
> Digium, Inc.
>
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[asterisk-users] canary_thread

[Marcus Vinicius]

People,

Anybody knows what mean this message in my CLI:


[Apr  1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary is 
no more.  He has ceased to be!  He's expired and gone to meet his maker!  He's 
a stiff!  Bereft of life, he rests in peace.  His metabolic processes are now 
history!  He's off the twig!  He's kicked the bucket.  He's shuffled off his 
mortal coil, run down the curtain, and joined the bleeding choir invisible!!  
THIS is an EX-CANARY.  (Reducing priority)
mediagw*CLI>

Asterisk: 1.6.2.6

tks


  

Veja quais são os assuntos do momento no Yahoo! +Buscados
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Re: [asterisk-users] canary_thread

[Andrew Latham]

Hint Apr  1


~
Andrew "lathama" Latham
lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Thu, Apr 1, 2010 at 6:05 PM, Marcus Vinicius  wrote:
> People,
>
> Anybody knows what mean this message in my CLI:
>
>
> [Apr  1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary
> is no more.  He has ceased to be!  He's expired and gone to meet his maker!
> He's a stiff!  Bereft of life, he rests in peace.  His metabolic processes
> are now history!  He's off the twig!  He's kicked the bucket.  He's shuffled
> off his mortal coil, run down the curtain, and joined the bleeding choir
> invisible!!  THIS is an EX-CANARY.  (Reducing priority)
> mediagw*CLI>
>
> Asterisk: 1.6.2.6
>
> tks
>
>
>
>
>
>
> 
> Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 -
> Celebridades - Música - Esportes
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Re: [asterisk-users] canary_thread

[Danny Nicholas]

You’d think that this is/was some kind of April fool message, but it is a
real 1.6 warning

http://lists.digium.com/pipermail/asterisk-commits/2008-May/022745.html

 

Since 1.6 has more multi-thread capabilities, the good folks at
Digium/Asterisk made this warning program to keep runaway threads from
crippling Asterisk.  When you get this message, the mine is about to
collapse (potentially) on your Asterisk instance.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marcus
Vinicius
Sent: Thursday, April 01, 2010 4:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] canary_thread

 

People,

Anybody knows what mean this message in my CLI:


[Apr  1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary
is no more.  He has ceased to be!  He's expired and gone to meet his maker!
He's a stiff!  Bereft of life, he rests in peace.  His metabolic processes
are now history!  He's off the twig!  He's kicked the bucket.  He's shuffled
off his mortal coil, run down the curtain, and joined the bleeding choir
invisible!!  THIS is an EX-CANARY.  (Reducing priority)
mediagw*CLI>

Asterisk: 1.6.2.6

tks






 

  _  

Veja quais são os assuntos do momento no Yahoo! + Buscados: Top
  10 - Celebridades
  - Música
  - Esportes
 

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Re: [asterisk-users] Updating Asterisk and its use with MySQL

[Daniel Bareiro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi, Alyed.

On Sun, 28 Mar 2010, Alyed wrote:

>> I didn't know that there was Digium's GUI. It is FLOSS? I was looking
>> for in the site of Digium in the download section, but the unique
>> thing that I saw that it speaks of a GUI is AsteriskNow, that in fact
>> it is a complete distribution of GNU/Linux. You talked about to the
>> GUI provided by AsteriskNow? Because if is this case, I don't believe
>> that it is very practical. When I spoke of GUI was referring to a
>> separated component to install over which already one had running.
>>
>> As far as the use of Asterisk with a DBMS (MySQL, for example), do
>> you know some document or reference where indicate the steps to
>> follow to migrate from config files?

> Yes I'm talking about Asterisk Now's GUI and yes, you can just install
> this component.
> google for Asterisk Gui 2.0 and you'll find plenty of info.

Perfect. I will consider it. Thanks for the reference. In the tests that
you said to me that you were doing, did you find this GUI as extensible
as FreePBX?

> Regarding the DB I can't help you here, maybe someone else can.

Well. If somebody can add something on this subject, will be welcome.

Thanks for your reply.

Regards,
Daniel

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v6oAn2YfB3s9RvqbqFt9/WSvX+TV4eqx
=9PWK
-END PGP SIGNATURE-


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Re: [asterisk-users] Updating Asterisk and its use with MySQL

[Daniel Bareiro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi, Jim.

On Sun, 28 Mar 2010, Jim Dickenson wrote:

> I think if you are installing dahdi complete from source you do
> "make all" and "make install" and "make config"

Something that I forgot to ask previously is if the update of Asterisk
or DAHDI is independent or the update of a component requires to also
update the other.

Thanks in advance for your reply.

Regards,
Daniel

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (GNU/Linux)

iEYEARECAAYFAku1IFAACgkQZpa/GxTmHTdZaACdGP8CAFLaGP2ek4pvdC2eHLOF
3noAnAhyWdDVboeGWzfP3Hw45s3jMPip
=w4HG
-END PGP SIGNATURE-


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[asterisk-users] SIP Connection Question

[Kenneth Noisewater]

Hi All,

I have a question about how a particular situation would work between two
PBX systems:

If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same
rack, same network), and then pass a call from the Mitel to Asterisk to
perform some functions (lookups, maybe routing), and then pass the call back
to the Mitel to be routed to it's endpoint, would Asterisk stay in that loop
after the call was passed back to the Mitel? Or, does the call leave
Asterisk completely when passed back?

If it does leave/stay in the loop, is there a way to force it to leave/stay
based on what my needs are?

Thanks,

Kenny
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Re: [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM

[Chris Miller]

On 4/1/2010 1:52 PM, Kevin P. Fleming wrote:
> Chris Miller wrote:
>
>> A comment in the spec file would have been nice... Does anyone know
>> if this a real technical issue, or simply a licensing conflict
>> between GPL and Digium?
>
> It is not a technical issue; it is an issue because some of the modules
> in -addons have licenses that are pure GPLv2 only, and in addition the
> license for MySQL-based components restricts their usage to *only*
> GPLv2-licensed applications unless a commercial license for MySQL is
> obtained. Since loading one of Digium's binary modules into an Asterisk
> process changes it to no longer be pure GPLv2, such usage restrictions
> should be taken into account.
>
> The purpose of that conflict is to ensure that the person installing the
> packages is made aware of the issue and that they must take explicit
> action to override it (thus ensuring that we don't facilitate accidental
> violation of third-party license agreements).


Understood, I figured it was something like that. Do you have some 
mechanism in the source install that causes similar enforcement 
behavior?


> If you can suggest a method to provide this information to people in
> some automatic way when they are made aware of the conflict by RPM, feel
> free to do so and we'll try to get it incorporated into the RPMs themselves.


A method of providing the GPL license conflict information at 
install time, or the reason for (and resolution of) the RPM install 
conflict?

It seems to me that the GPL information could be displayed in the 
"register" binary since no end user can use a Digium supplied 
commercial module without registration, right? It could also be 
displayed on the Digium website where end users have to purchase 
their Digium licenses.

This begs the question of when the actual violation occurs. In other 
words, is this really a "usage" issue, or does the violation occur 
at install time even though the non-GPL component is not "usable"?

It sounds to me that many users are violating the GPL by installing 
the non-GPL modules. Rather than simply making it difficult to 
install, why not be proactive in encouraging compliance by detailing 
the steps openly. When I Googled for this issue, I turned up no 
useful information. Seems like a page explaining the above somewhere 
on the Asterisk and/or Digium site would be helpful.

The only workaround at this point is to force install the RPMs. This 
encourages lesser skilled sysadmins to use this practice regularly 
(on all Linux dependency issues) without fully understanding what 
they are doing. I took the time to download the SRPM and saw this 
was an arbitrary dependency, but most sysadmins won't burn the time. 
What also concerned me was a few posts about a system stability 
issue with the SkypeForAsterisk module after force installing the 
RPM. This contributed to my being uneasy about proceeding with this 
route without full knowledge of the situation.

Alternatively I need to maintain my own version of 
asterisk-addons-core without the gplonly provide. Kinda defeats the 
purpose of using a third party repository for convenience.

I understand the reasons why this was done, but unless I've 
overlooked some resource on the interwebs, it looks like the other 
shoe never dropped and zero documentation was provided to work with 
this issue. I can't think of a clean way off the top of my head to 
address this in RPM, so I'd argue that RPM is simply not the 
appropriate choke point to enforce compliance. Feel free to send me 
a PM if you want to discuss further.

Regards,
Chris

Chris Miller
President - Rocket Scientist
ScratchSpace Inc.
(831) 621-7928
http://www.scratchspace.com

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Re: [asterisk-users] SIP Connection Question

[Juan E. Rodríguez]

Depends on the configuration you make. For example, if you want to route the 
call giving the Mitel a new desrination or prefix, you can use Transfer 
dialplan app. Transfer before answering the call will be redirected with SIP 
302.

If the call is to be anwered on *, then canreinvite set to yes or directrtp set 
to yes can help you.


Saludos,
Juan E. Rodríguez


-Original Message-
From: Kenneth Noisewater 
Date: Thu, 1 Apr 2010 16:50:47 
To: 
Subject: [asterisk-users] SIP Connection Question

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Re: [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM

[Richard Kenner]

> This begs the question of when the actual violation occurs. In other 
> words, is this really a "usage" issue, or does the violation occur 
> at install time even though the non-GPL component is not "usable"?

It's hard to see how the violation could occur unless and until the
resulting program were distributed.  Yes, executing a program is
technically making a copy, but the GPL, along with almost all licenses,
don't make any restrictions on that copy. 

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Re: [asterisk-users] SIP Connection Question

[Dr. Kenneth Noisewater]

On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote:
> Depends on the configuration you make. For example, if you want to route the 
> call giving the Mitel a new desrination or prefix, you can use Transfer 
> dialplan app. Transfer before answering the call will be redirected with SIP 
> 302.
>
> If the call is to be anwered on *, then canreinvite set to yes or directrtp 
> set to yes can help you.
>
>
> Saludos,
> Juan E. Rodríguez
>
>
> -Original Message-
> From: Kenneth Noisewater
> Date: Thu, 1 Apr 2010 16:50:47
> To:
> Subject: [asterisk-users] SIP Connection Question
>
>
OK, so for instance if I passed a call to Asterisk and grabbed CID info 
and did some lookups and then transferred it back to mitel to route to a 
user, then * would be out of the call path (loop, whatever). But, if I 
were to answer that call in * with an IVR to collect caller input to use 
and then transferred the call back to the Mitel to route to the 
endpoint, * would remain in the call. Is that a correct understanding?

Also one more question, and please excuse my ignorance (I'm just a 
developer with pretty limited knowledge on the telephony side of things):

When I talk about connecting the Mitel box and the Asterisk box together 
via a SIP trunk, is that trunk equal to 1 analog line, or channel or 
whatever, or can I make as many connections as I want on that trunk? 
Again, my knowledge is a bit limited, and thusfar people have been using 
a lot of terms interchangably with me to add to my confusion :). This 
only concerns me because I'm pretty sure we have to buy a license for 
each SIP trunk with Mitel.

It would be really great if I could work out a solution like this, it 
will allow me to prove Asterisk's worth to my management, and open up a 
lot of doors for us and our internal apps. The Mitel SDK is 
unfortunately rather limited, but management is not in any way 
interested in jumping ship from Mitel to Asterisk. Personally, I say 
jump, I've had great experience with Asterisk, even in fairly heavy use 
situations. Anyone have any input on selling Asterisk to higher up's? I 
know there is the whole "enterprise support" aspect, but my team manages 
the Mitel stuff as it is anyway, and I think we'd all much rather be 
dealing with Asterisk/SER as the core solution.

Thanks everyone for your input!

Kenny

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Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Jeff LaCoursiere]

I finally got it at "Calories Consumed".  Geesh.  Good one!  :)

j

On Thu, 1 Apr 2010, Olle E. Johansson wrote:

> FOR IMMEDIATE RELEASE
> Puerto Escondido, Mexico, April 1st, 2010:
>
> Digium launches Asterisk VCC (TM) - a new virtual communication platform
> for enterprises, the public sector and the home.
> ===
>
> Asterisk 1.8 will contain a stunning new technology for all Asterisk users 
> world-
> wide - virtual communication clouds or VCC (TM).  With this technology, call
> handling will never be the same. In one move, the Asterisk development team
> leaves the old world of PBX call switching behind and moves the enterprise
> telephony server to the cloud.
>
> By combining IPv6, the 3G cell network and cloud services with existing 
> Asterisk
> technologies  like Dundi and IAX2, Digium moves into the era of cloud 
> computing.
> The launch includes end-user applications powered by cloud services
> - moving Digium technology to the palm of your hand.
>
> - "Our new platform is built for the new organization in the workplace, the 
> family
> and the community - a truly virtual multimedia communication network for the
> Internet age. By moving our focus away from the traditional PBX, we succeeded
> in changing the  Digium solution from a server centric view to a service 
> centric view."
> says Sokkie Stevens, product manager for the new platform.
>
> The first step was to transform Digium into a virtual service provider. Digium
> is one of the first companies to get an IPv6 assignment on a global service
> provider level. After signing peering agreements with major carriers world-
> wide, the next step was to apply the successful Dundi protocol on top of
> IPv6.
>
> -"Dundi and IPv6 was a match made in heaven", says Mick Spenser,
> the CTO for Digium, "Dundi had a successful peering and discovery
> infrastructure that is now even stronger with IPv6 multicast and secured
> by using IPsec."
>
> VCC will be a binary module distributed with Asterisk 1.8. It will connect
> to the Digium VCCnet over native IPv6, IPv6 over IPv4 tunnels and directly
> over layer 2 technologies like Ethernet. All VCC clients will get a native
> IPv6 address assigned. Enterprises may purchase a full IPv6 network range
> in the VCCnet to get full access. VCCnet is a network service managed
> by Digium worldwide.
>
> VCCnet will enable automatic follow-me functionality. When you turn
> on your VCC-enabled smartphone, the VCCnet client will automatically report 
> your
> location (from 3G cells or GPS) back to the Asterisk service. Your status
> will be automatically updated as you move between networks, from
> WiFi in the office to 3G on the road. One person can have multiple
> VCC clients - one supporting video, another old-fashioned audio
> and a third HD audio and video. The new IAX3 protocol used in VCCnet
> will automatically negiotiate media capabilities and select the right client
> for the right call, depending on privacy settings and personal preferences.
>
> For VoxSwitch customers, VCCnet will mean that every user can monitor
> the movement of coworkers in realtime. By using the new APIs, additional
> data like credit card transactions, fuel consumption in the car, mileage
> in the air and calories eaten can be reported with a 3D graphical display
> using HTML5.
>
> As an additional service in the VCCnet cloud, Digium will offer extended
> capacity for your telecommunications platform. When you need more capacity
> for video calls, 5+1 hd voice conferences and other coming services, including
> 3D multimedia conferencing, your existing PBX will be virtually extended by 
> using
> resources available (and unused) in the cloud. For the system manager, it will
> look like all these services are produced locally, just like before.
>
> VCC includes clients for all popular platforms, including the soon to be 
> released
> Apple iPAD. "Many people was asking us for the Digium Phone, but it felt very
> wrong to implement an old-fashioned device on top of a modern communication
> network" says Mike Spenser. "The client will be a natural part of the 
> personal computing
> infrastructure that already exists out there. It will be the personal 
> communication
> exchange, the Facebook of the multimedia realtime communications world."
>
> Digium will rename the recently launched Asterisk marketplace to
> VCCstore and  use that infrastructure for distribution of the VCCblocks
> - applets that enhance your virtual communication cloud. 3rd party developers
> may apply for development kits and distribution agreements. Digium is 
> currently
> negotiating the rights to distribute audio books and radio shows for the new
> culture-on-hold service while not using the VCCclient for two- or multiparty 
> communication.
>
> While testing, the most popular VCCblock was the TimeShiftBlock that includes
> the former voicemail service, now enhanced with virtual timeshifting for 
> realtime

Re: [asterisk-users] SIP Connection Question

[Juan E. Rodríguez]

If * answers the call, it will be on the "loop" but with canreinvite or 
directrtp the media can be out of * and redirected to the final end point even 
if signaling goes through *.

For the trunk, you can have multiple simultaneous calls. I do not know about 
Mitel's licensing but with only one trunk you can have as much calls as * 
supports.

Saludos,
Juan E. Rodríguez


-Original Message-
From: "Dr. Kenneth Noisewater" 
Date: Thu, 01 Apr 2010 19:35:54 
To: ; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] SIP Connection Question

On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote:
> Depends on the configuration you make. For example, if you want to route the 
> call giving the Mitel a new desrination or prefix, you can use Transfer 
> dialplan app. Transfer before answering the call will be redirected with SIP 
> 302.
>
> If the call is to be anwered on *, then canreinvite set to yes or directrtp 
> set to yes can help you.
>
>
> Saludos,
> Juan E. Rodríguez
>
>
> -Original Message-
> From: Kenneth Noisewater
> Date: Thu, 1 Apr 2010 16:50:47
> To:
> Subject: [asterisk-users] SIP Connection Question
>
>
OK, so for instance if I passed a call to Asterisk and grabbed CID info 
and did some lookups and then transferred it back to mitel to route to a 
user, then * would be out of the call path (loop, whatever). But, if I 
were to answer that call in * with an IVR to collect caller input to use 
and then transferred the call back to the Mitel to route to the 
endpoint, * would remain in the call. Is that a correct understanding?

Also one more question, and please excuse my ignorance (I'm just a 
developer with pretty limited knowledge on the telephony side of things):

When I talk about connecting the Mitel box and the Asterisk box together 
via a SIP trunk, is that trunk equal to 1 analog line, or channel or 
whatever, or can I make as many connections as I want on that trunk? 
Again, my knowledge is a bit limited, and thusfar people have been using 
a lot of terms interchangably with me to add to my confusion :). This 
only concerns me because I'm pretty sure we have to buy a license for 
each SIP trunk with Mitel.

It would be really great if I could work out a solution like this, it 
will allow me to prove Asterisk's worth to my management, and open up a 
lot of doors for us and our internal apps. The Mitel SDK is 
unfortunately rather limited, but management is not in any way 
interested in jumping ship from Mitel to Asterisk. Personally, I say 
jump, I've had great experience with Asterisk, even in fairly heavy use 
situations. Anyone have any input on selling Asterisk to higher up's? I 
know there is the whole "enterprise support" aspect, but my team manages 
the Mitel stuff as it is anyway, and I think we'd all much rather be 
dealing with Asterisk/SER as the core solution.

Thanks everyone for your input!

Kenny
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Re: [asterisk-users] canary_thread

[Darrick Hartman]

Danny,

I haven't been able to test further, but I've seen the same issue since 
I upgraded to 1.6.2.6.  The astcanary does not appear to stay up more 
than a few seconds when asterisk is initially started.  On this same 
system using 1.6.2.2 (the previous version I was running prior to the 
upgrade) it was running fine.

I never did like birds anyway...

Darrick

On 04/01/2010 04:22 PM, Danny Nicholas wrote:
> You’d think that this is/was some kind of April fool message, but it is
> a real 1.6 warning
>
> http://lists.digium.com/pipermail/asterisk-commits/2008-May/022745.html
>
> Since 1.6 has more multi-thread capabilities, the good folks at
> Digium/Asterisk made this warning program to keep runaway threads from
> crippling Asterisk.  When you get this message, the mine is about to
> collapse (potentially) on your Asterisk instance.
>
> 
>
> *From:* asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marcus
> Vinicius
> *Sent:* Thursday, April 01, 2010 4:06 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] canary_thread
>
> People,
>
> Anybody knows what mean this message in my CLI:
>
>
> [Apr 1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The
> canary is no more. He has ceased to be! He's expired and gone to meet
> his maker! He's a stiff! Bereft of life, he rests in peace. His
> metabolic processes are now history! He's off the twig! He's kicked the
> bucket. He's shuffled off his mortal coil, run down the curtain, and
> joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing
> priority)
> mediagw*CLI>
>
> Asterisk: 1.6.2.6


-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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Re: [asterisk-users] canary_thread

[Stefan Schmidt]

Hello,

i´ve got this when i asterisk has died / killed and was restarted but i 
dont have seen that it will collapse then.

i also got this after restarting asterisk from the CLI with restart now.

so dont worry ;)

best regards

steve smith

Danny Nicholas schrieb:
>
> You’d think that this is/was some kind of April fool message, but it 
> is a real 1.6 warning
>
> http://lists.digium.com/pipermail/asterisk-commits/2008-May/022745.html
>
> Since 1.6 has more multi-thread capabilities, the good folks at 
> Digium/Asterisk made this warning program to keep runaway threads from 
> crippling Asterisk. When you get this message, the mine is about to 
> collapse (potentially) on your Asterisk instance.
>
> 
>
> *From:* asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marcus 
> Vinicius
> *Sent:* Thursday, April 01, 2010 4:06 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] canary_thread
>
> People,
>
> Anybody knows what mean this message in my CLI:
>
>
> [Apr 1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The 
> canary is no more. He has ceased to be! He's expired and gone to meet 
> his maker! He's a stiff! Bereft of life, he rests in peace. His 
> metabolic processes are now history! He's off the twig! He's kicked 
> the bucket. He's shuffled off his mortal coil, run down the curtain, 
> and joined the bleeding choir invisible!! THIS is an EX-CANARY. 
> (Reducing priority)
> mediagw*CLI>
>
> Asterisk: 1.6.2.6
>
> tks
>
>
>
>
> 
>
> Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 
>  
> - Celebridades 
> 
>  
> - Música 
> 
>  
> - Esportes 
> 
>


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[asterisk-users] there have any one run asterisk on ubuntu enterprise cloud ?

[Jeffery]

-- 
Jeffery
___
/\__\ "What is the world coming to?"
\/__/
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Re: [asterisk-users] Problem with Sangoma A104 and euroisdn pri

[RESEARCH]

Can you post outputs for the following commands;

#asterisk -rx 'pri show spans'
#asterisk -rx 'zap show channels' 
#wanpipemon -i w1g1 -c Ta

Sam

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent: Thursday, April 01, 2010 4:15 PM
To: Asterisk
Subject: [asterisk-users] Problem with Sangoma A104 and euroisdn pri

Hi all,

My problem boils down to these errors:

... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time

This is triggered by lines in extentions.conf such as:

exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W)

The system is CentOS v5.2 with Asterisk 1.4.23  
(druid-asterisk-1.4.23.1-2), a Sangoma A104 4-port card, Wanpipe  
v3.4.4 and Zaptel v1.4.12.1. The system is attached to a single  
EuroISDN PRI and is located in the Netherlands.

Besides the above error, I also noticed this:

CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3

The status needs to be "Provisioned, Up, Active."

Following Sangoma's instructions for debugging an Asterisk PRI span, I  
can confirm that there are only outgoing frames and that the D-channel  
messages in Asterisk are the same as what the Wanpipe drivers are  
seeing. So, assuming that my local telco (KPN Telecom) has activated  
the D-channel, what else could possibly be causing this problem?

Thanks,

Jaap

PS -- Below are my current configuration files and debugging output:

==begin zaptel.conf 

loadzone=us
defaultzone=us
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
hardhdlc=16

==end zaptel.conf ==

==begin wanpipe1.conf ==

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS  = 13
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= NCRC4
FE_LINE= 1
TE_CLOCK = NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE= NO
MTU = 1500
UDPPORT = 9000
TTL= 255
IGNORE_FRONT_END = NO
TDMV_SPAN= 1
TDMV_DCHAN= 16
TDMV_HW_DTMF= NO
TDMV_HW_FAX_DETECT = NO

[w1g1]
ACTIVE_CH= ALL
TDMV_HWEC= NO

==end wanpipe1.conf 

==begin zapata.conf 

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

switchtype=euroisdn
context=default
group=1
signalling=pri_cpe
channel =>1-15,17-31

==end zapata.conf ==

Here's some debugging output:

=== begin debug info ==

# ztcfg -vv

Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: Hardware assisted D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Chann

[asterisk-users] Strange Centos Problem with Dahdi installation

[ABBAS SHAKEEL]

Hello Community,

I have installed Dahdi on Centos on many system and succesfully used that..
But today i have bad luck...
This is the error that i am facing


[r...@localhost dahdi-linux-complete-2.2.1+2.2.1]# make all
make -C linux all
make[1]: Entering directory
`/usr/src/dahdi-linux-complete-2.2.1+2.2.1/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
`/usr/src/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory
`/usr/src/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 2.6.18-128.el5PAE kernel
installed.
make[1]: *** [modules] Error 1
make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.2.1+2.2.1/linux'
make: *** [all] Error 2


When i try to upgrade kernel . Kernel Conflicts occur and system can not
boot properly.


I have tried to installed rpms and alot of brute force tries.. but in
vain. in the mean time i have reinstalled Centos Four times.

Please some one shed some light on this issue..

Kind Regards
Shakeel Abbas
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