asterisk users - Dec 2010 - How to quickly move on to Dahdi channels when SIP provider fails?

Hi Everyone,

There are situations when internet connection is lost, SIP provider fails,
or even authentication to SIP provider fails, and we want to use the backup
Dahdi channels (PSTN). As simple as it may sound but with the
many different situations and error messages it seems like it's not so easy
to predict all the errors. Is there any single parameter value that can be
changed to send the call to Dahdi instead of SIP in-case for any reason the
SIP provider fails? Because I see a server with this need waits for long
time for DNS to resolve when there is no internet even.

Thanks
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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 08, 2010 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to quickly move on to Dahdi channels when
SIPprovider fails?

 

Hi Everyone,

 

There are situations when internet connection is lost, SIP provider fails,
or even authentication to SIP provider fails, and we want to use the backup
Dahdi channels (PSTN). As simple as it may sound but with the many different
situations and error messages it seems like it's not so easy to predict all
the errors. Is there any single parameter value that can be changed to send
the call to Dahdi instead of SIP in-case for any reason the SIP provider
fails? Because I see a server with this need waits for long time for DNS to
resolve when there is no internet even.

 

Thanks

 

Just my .02, but since you're going to (quite possibly) have a long(ish)
timeout if internet connection or SIP provider is down, I would have an AGI
run in front of my dial that did a ping to verify internet and sip provider
connectivity.  Something like this:

Exten => s,1,AGI(verconn.agi)

Exten => s,n,gotoif(${CONNUP} = yes?3:4)

Exten => s,n,Dial(SIP.)

Exten => s,n,Dial(DAHDI.)

 

Verconn.agi could be PERL, C, PHP or even bash.

 

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On 08/12/10 17:53, Danny Nicholas wrote:
> 
> Thanks
> 
>  
> 
> Just my .02, but since you?re going to (quite possibly) have a long(ish)
> timeout if internet connection or SIP provider is down, I would have an
> AGI run in front of my dial that did a ping to verify internet and sip
> provider connectivity.  Something like this:
> 
> Exten => s,1,AGI(verconn.agi)
> 
> Exten => s,n,gotoif(${CONNUP} = yes?3:4)
> 
> Exten => s,n,Dial(SIP?)
> 
> Exten => s,n,Dial(DAHDI?)
I use the following

exten => _9X.,n,Set(TMP=${SIPPEER(sip_outbound,status)})
exten => _9X.,n,GotoIf($["${TMP:0:2}" != "OK"]?isdn)
exten => _9X.,n,Dial(${DIALOUTSIP}/${EXTEN:1})
...
exten => _9X.,n(isdn),Dial($(DIALOUTISDN)/${EXTEN:1})
...

But you mention DNS also look into the possibility of using something
like dnsmasq to give a resiliant dns for when the internet connection fails.

DC

2010/12/8 Bruce B <bruceb444 at gmail.com>
> Hi Everyone,
>
> There are situations when internet connection is lost, SIP provider fails,
>
What do you mean by "SIP provider fails" ?
An outbound call returns "CONGESTION" though internet connection is up
?
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Hi!
> There are situations when internet connection is lost, SIP provider
> fails, or even authentication to SIP provider fails, and we want to use
> the backup Dahdi channels (PSTN). As simple as it may sound but with
> the manydifferentsituations and error messages it seems like it's not
> so easy to predict all the errors. Is there any single parameter value
> that can be changed to send the call to Dahdi instead of SIP 
There is nothing available out-of-the-box. You need to include your own IP &
SIP tests in the
dialplan before dialing out to a SIP channel. Useful for this purpose are 

- ping and host or wget, 
- GROUP() and GROUP_COUNT(), 
- SIPPEER(xxx:status), 
- CHANISAVAIL(), 
- dial timeouts and 
- post-dial error handling (see DIALSTATUS and HANGUPCAUSE as well as Asterisk
1.8
with its ability to act directly upon the SIP response code).  

Philipp

can someone give more eduaction to me about what the asterisk exchange is
all about?

thanks

On Thu, Dec 9, 2010 at 5:43 AM, Sevana Oy <sales at sevana.fi> wrote:
>  Hi,
>
> A couple of months ago we registered our product AQuA at Asterisk Exchange.
> We were told that it collects like 14K visitors per month and knowing
> interest to our product from Asterisk community we have calculated a
certain
> super-mini-minimal % of visitors coming from Asterisk Exchange to our web
> site... Here comes the funny thing - there was no traffic increase since
> then, there were no referral visist from Asterisk Exchange... We are
getting
> something a bit less than 100 product inquiries a month and NONE has ever
> mentioned that learnt about us from Asterisk Exchange...
>
> My question is: have we just wasted $2500 for being listed there? Is
> Asterisk Exchange some kind of bubble? Unfortunately we never got response
> from the people who sold us this service :-)
>
> Thanks and cheers,
> Vallu
> Sevana Oy
>
> --
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