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huu giang | 1 Apr 2010 04:22
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Re: Asterisk load balancing and failover

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 <tobias.wolf <at> evision.de> wrote:

From: Tobias Wolf <tobias.wolf <at> evision.de>
Subject: Re: [asterisk-users] Asterisk load balancing and failover
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users <at> lists.digium.com>
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 /<zishanov <at> gmail.com>/* wrote:
>
>
>     From: Zeeshan Zakaria <zishanov <at> gmail.com>
>     Subject: Re: [asterisk-users] Asterisk load balancing and failover
>     To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>     <asterisk-users <at> lists.digium.com>
>     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" <huugiang104 <at> yahoo.com
>>     </mc/compose?to=huugiang104 <at> yahoo.com>> 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.
>>
>>
>>
>>
>>     --
>>     _____________________________________________________________________
>>     -- 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
>>
>>     asterisk-users mailing list
>>     To UNSUBSCRIBE or update options visit:
>>       http://lists.digium.com/mailman/listinfo/asterisk-users
>
>     -----Inline Attachment Follows-----
>
>     --
>     _____________________________________________________________________
>     -- 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
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>        http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


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Olle E. Johansson | 1 Apr 2010 07:53
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Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

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
calls between timezones. The TimeShiftBlock includes ten popular synthetic voices,
including the Asterisk Diva Allison Smiths' attractive voice. In addition to Allison
Digium added the voice of the southern gentleman Danny Wyndham and the 
Swenglish dialect of Asterisk developer and guru Olle E. Johansson, one that
was recognized with a strange smile by all Asterisk developers testing VCC.

VCCnet technology includes scalability and security components  licensed by
Edvina AB in Sweden. Edvina's experience of large scale Unified Communication
networks was necessary to build a world-wide network-centric platform for 
this new service. 

- "We find it exciting to contribute to this new service. Realizing the perfect
match between the open IPv6 protocol and the proprietary Dundi technology
was an eye-opener. No NAT issues and the possibility to build a worldwide
network with service discovery, security and managed QoS will make this
a success story. We're proud to contribute to this solution." says Olle E. Johansson,
founder of Edvina. "The new IAX3 protocol is also really interesting, as it
not only combines media and signalling over one port, but now also adds
presence, instant messaging, file transfer, printing, database queries, directory
services and network management  over the same port. It's a one-size-fits-all 
protocol that will handle all services a user want."

The VCCnet network is already in operation, The VCCnet PBX interface will be part
of Asterisk 1.8 to be launched later this year and part of the VoxSwitch update
Q2 2010. The VCCstore opens June 1st. Development kits are available to
Digium authorized VCC development partners today. The VCC technology
is patented by Digium and will be operated as a private virtual network on top
of the Internet and the ISDN network.

For questions and further information, please contact the Digium marketing department at
looflirpa <at> digium.com today. A press conference will be held April 1st, 15:00 GSM+1 in 
VCCconference room 142857 for media representatives. It will be available for one 
week on vcc://digium.com::conference:142857 for later viewing.

VCC, VCCnet, VCCblock, VCCstore, Digium, IAX3, Dundi and Asterisk are
trademarks registered by Digium Inc.

--

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Lenz Emilitri | 1 Apr 2010 09:38
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Re: Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

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


2010/4/1 Olle E. Johansson <oej <at> edvina.net>
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.

--
Loway - home of QueueMetrics - http://queuemetrics.com

--

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Randy R | 1 Apr 2010 09:52
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Re: Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

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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Ioan Indreias | 1 Apr 2010 11:32
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Re: Necessary hardware

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 <kosa <at> piradio.org> 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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Vitali Fomine | 1 Apr 2010 11:46

OfficeSIP Communications Makes Its VoIP SIP Products Open Source

Press Release

For Immediate Release

OfficeSIP Communications
http://www.officesip.com/
info <at> 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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Ngo-Vi Hoai-Anh | 1 Apr 2010 12:01
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Re: Asterisk load balancing and failover

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]<----------------VoIP (SIP/IAX etc...)---------> 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 /<tobias.wolf <at> evision.de>/* wrote:
>
>
>     From: Tobias Wolf <tobias.wolf <at> evision.de>
>     Subject: Re: [asterisk-users] Asterisk load balancing and failover
>     To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>     <asterisk-users <at> lists.digium.com>
>     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 /<zishanov <at> gmail.com
>     </mc/compose?to=zishanov <at> gmail.com>>/* wrote:
>     >
>     >
>     >     From: Zeeshan Zakaria <zishanov <at> gmail.com
>     </mc/compose?to=zishanov <at> gmail.com>>
>     >     Subject: Re: [asterisk-users] Asterisk load balancing and
>     failover
>     >     To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>     >     <asterisk-users <at> lists.digium.com
>     </mc/compose?to=asterisk-users <at> lists.digium.com>>
>     >     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" <huugiang104 <at> yahoo.com
>     </mc/compose?to=huugiang104 <at> yahoo.com>
>     >>     </mc/compose?to=huugiang104 <at> yahoo.com
>     </mc/compose?to=huugiang104 <at> yahoo.com>>> 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.
>     >>
>     >>
>     >>
>     >>
>     >>     --
>     >> 
>        _____________________________________________________________________
>     >>     -- 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
>     >>
>     >>     asterisk-users mailing list
>     >>     To UNSUBSCRIBE or update options visit:
>     >>       http://lists.digium.com/mailman/listinfo/asterisk-users
>     >
>     >     -----Inline Attachment Follows-----
>     >
>     >     --
>     > 
>        _____________________________________________________________________
>     >     -- 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
>     >
>     >     asterisk-users mailing list
>     >     To UNSUBSCRIBE or update options visit:
>     >        http://lists.digium.com/mailman/listinfo/asterisk-users
>     >
>     >
>
>
>     -- 
>     _____________________________________________________________________
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>     New to Asterisk? Join us for a live introductory webinar every Thurs:
>                    http://www.asterisk.org/hello
>
>     asterisk-users mailing list
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>        http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

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Jorge Churio | 1 Apr 2010 14:28
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Asterisk Load Balancing with Redfone/TDMoE driver

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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Tommy Botten Jensen | 1 Apr 2010 14:41
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Re: Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8


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 most popular VCCblock was the TimeShiftBlock that includes
> the former voicemail service, now enhanced with virtual timeshifting for realtime
> calls between timezones. The TimeShiftBlock includes ten popular synthetic voices,
> including the Asterisk Diva Allison Smiths' attractive voice. In addition to Allison
> Digium added the voice of the southern gentleman Danny Wyndham and the 
> Swenglish dialect of Asterisk developer and guru Olle E. Johansson, one that
> was recognized with a strange smile by all Asterisk developers testing VCC.
> 
> VCCnet technology includes scalability and security components  licensed by
> Edvina AB in Sweden. Edvina's experience of large scale Unified Communication
> networks was necessary to build a world-wide network-centric platform for 
> this new service. 
> 
> - "We find it exciting to contribute to this new service. Realizing the perfect
> match between the open IPv6 protocol and the proprietary Dundi technology
> was an eye-opener. No NAT issues and the possibility to build a worldwide
> network with service discovery, security and managed QoS will make this
> a success story. We're proud to contribute to this solution." says Olle E. Johansson,
> founder of Edvina. "The new IAX3 protocol is also really interesting, as it
> not only combines media and signalling over one port, but now also adds
> presence, instant messaging, file transfer, printing, database queries, directory
> services and network management  over the same port. It's a one-size-fits-all 
> protocol that will handle all services a user want."
> 
> The VCCnet network is already in operation, The VCCnet PBX interface will be part
> of Asterisk 1.8 to be launched later this year and part of the VoxSwitch update
> Q2 2010. The VCCstore opens June 1st. Development kits are available to
> Digium authorized VCC development partners today. The VCC technology
> is patented by Digium and will be operated as a private virtual network on top
> of the Internet and the ISDN network.
> 
> For questions and further information, please contact the Digium marketing department at
> looflirpa <at> digium.com today. A press conference will be held April 1st, 15:00 GSM+1 in 
> VCCconference room 142857 for media representatives. It will be available for one 
> week on vcc://digium.com::conference:142857 for later viewing.
> 
> VCC, VCCnet, VCCblock, VCCstore, Digium, IAX3, Dundi and Asterisk are
> trademarks registered by Digium Inc.
> 
> 
>
Permalink | Reply | 7 comments |
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Jaap Winius | 1 Apr 2010 15:15
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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: 10000
    T305 Timer: 30000
    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/A | 0
   |

Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |

# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
           UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
           RX packets:5281234 errors:0 dropped:0 overruns:0 frame:0
           TX packets:5281234 errors:0 dropped:0 overruns:0 carrier:4
           collisions:0 txqueuelen:100
           RX bytes:0 (0.0 b)  TX bytes:0 (0.0 b)
           Interrupt:169 Memory:f8b40000-f8b41fff

#

=== end debug info ====================================================

--

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