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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Zeeshan Zakaria
2010-Mar-26 08:51 UTC
[asterisk-users] Asterisk load balancing and failover
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>
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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Zeeshan Zakaria
2010-Mar-26 12:20 UTC
[asterisk-users] Asterisk load balancing and failover
Unfortunately not. DRBD and Heartbeat solution is good for pure VoIP. On 2010-03-26 6:33 AM, "huu giang" <huugiang104 at yahoo.com> wrote: 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. 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 andHeartbeat. The worke... -----Inline Attachment Follows----- --> _____________________________________________________________________ > -- Bandwidth and Colocatio...-- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/1d3b004f/attachment.htm
I have 1 PRI and 1 E&M Wink Circuit. If I call a non working number and route it through the PRI, I get the following: <Ringing>...."You have reached a non-working number....." If I call a non working number and route it through the E&M Wink Circuit, I get the following: <NO RINGING... DEAD AIR> A core show channels shows the state as ringing... Is there something I have to do to get the "You have reached a non-working number....." message to play back? Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/0a77c16f/attachment.htm
Zeeshan Zakaria
2010-Mar-26 14:25 UTC
[asterisk-users] Not hearing Telco Operator messages
Can you post your /etc/zaptel.conf and /etc/asterisk/zapata.conf.
--
Zeeshan
On 2010-03-26 9:06 AM, "Robert Grignon" <rgrignon at
fleetone.com> wrote:
I have 1 PRI and 1 E&M Wink Circuit.
If I call a non working number and route it through the PRI, I get the
following:
<Ringing>...."You have reached a non-working number....."
If I call a non working number and route it through the E&M Wink Circuit, I
get the following:
<NO RINGING... DEAD AIR>
A core show channels shows the state as ringing...
Is there something I have to do to get the "You have reached a non-working
number....." message to play back?
Thanks,
Robert
--
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>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?Have a look at this article and how they shared a single T1 line across two servers for failover: http://www.linuxjournal.com/article/7661 (sorry about the missing in-reply-to header. I'm not sure how to get Evolution to add in-reply-to manually and I'm receiving messages in digest form.) -- Eric Wheeler President Portland Linux Support www.PortlandLinuxSupport.com 503-330-4277 PO Box 86710 Portland, OR 97286
chan_dahdi.conf: ================================================;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2009-12-04 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [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=no relaxdtmf=yes rxgain=3.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A108 port 1 [slot:4 bus:8 span:1] <wanpipe1> switchtype=national context=from-xo-nvfleetone3 group=2 echocancel=yes signalling=pri_cpe channel =>1-23 ;Sangoma A108 port 2 [slot:4 bus:8 span:2] <wanpipe2> context=from-xo-transp154760dl group=1 echocancel=yes signalling=em_w channel => 25-48 /etc/dahdi/system.conf ====================================================#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2009-12-04 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A108 port 1 [slot:4 bus:8 span:1] <wanpipe1> span=1,0,0,esf,b8zs bchan=1-23 echocanceller=mg2,1-23 hardhdlc=24 #Sangoma A108 port 2 [slot:4 bus:8 span:2] <wanpipe2> span=2,0,0,esf,b8zs e&m=25-48 echocanceller=mg2,25-48 ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Friday, March 26, 2010 9:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not hearing Telco Operator messages Can you post your /etc/zaptel.conf and /etc/asterisk/zapata.conf. -- Zeeshan On 2010-03-26 9:06 AM, "Robert Grignon" <rgrignon at fleetone.com> wrote: I have 1 PRI and 1 E&M Wink Circuit. If I call a non working number and route it through the PRI, I get the following: <Ringing>...."You have reached a non-working number....." If I call a non working number and route it through the E&M Wink Circuit, I get the following: <NO RINGING... DEAD AIR> A core show channels shows the state as ringing... Is there something I have to do to get the "You have reached a non-working number....." message to play back? Thanks, Robert -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/c31ee1a0/attachment.htm
Zeeshan Zakaria
2010-Mar-31 11:37 UTC
[asterisk-users] Asterisk load balancing and failover
Hi, Good to know this but I am not the poster of this question and not doing any load balancing. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-03-31 7:33 AM, "Tobias Wolf" <tobias.wolf at evision.de> wrote: huu giang schrieb:> Hi Zeeshan > > I know a solution using DRBD, Heartbeat and RedFone hardware to > provide failover...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: > > > ...>> </mc/compose?to=huugiang104 at yahoo.com>> wrote: >> >> Hi List, >> >> I'm finding a sol...-- _____________________________________________________________________ -- Bandwidth and Colocation Pr... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100331/894f572a/attachment.htm