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Douglas Garstang

2005-Dec-08 21:03 UTC

[Asterisk-Users] Asterisk Dial Failover

All,
 
I have an Asterisk system that sends PSTN calls to an OpenSER system to be
routed. I have a command like this in my extensions.conf:
 
exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
 
There's actually two OpenSER systems for redundancy. I'm trying to find
a way to have Asterisk attempt to route the call to one OpenSER system, and if
it's down, fallback to another.
 
Any first thoughts on how to achieve this?
 
I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are
broken. If I issue a series of Dial commands, such as this:
 
exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
 
... what seems to happen is that when proxy1 is down, Asterisk waits the full 20
seconds before returning control. Also, This 20s includes the time is takes for
the other end to answer, so if I put a small value of say 5s in there, the dial
command will probably give up before someone answers at the other end. Neither
is workable.
 
Asterisk SHOULD be able to distinguish between a TRYING and no response. In the
event it gets no TRYING response to a dial command within a specified timeout it
should return control and flag an error. If on the other hand it does get a
TRYING response (and maybe a RINGING too) it should continue to wait until the
20s has expired.
 
I can't use dynamic DNS (ie putting two A records for a hostname in DNS)
because Asterisk reads the extensions.conf on startup and also seems to cache
what the host maps to on startup. Subsequent calls to the host always return the
same IP address.
 
But... in general... how have people implemented this? 
 
Help appreciated!
Doug
 
 
 
 
 
 
 
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Philipp von Klitzing

2005-Dec-09 04:51 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Hi!
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
> lookups are broken. If I issue a series of Dial commands, such as
> this: 
>  
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
How about you use ChanIsAvail() before each dial statement? Or try to 
Dial with 1 sec timeout or no timeout? Also check ${DIALSTATUS}.

Cheers, Philipp

burke@tailorhosting.com

2005-Dec-09 05:51 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes. One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system to be
> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a
> way to have Asterisk attempt to route the call to one OpenSER system, and
> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups
> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits the
> full 20 seconds before returning control. Also, This 20s includes the time
> is takes for the other end to answer, so if I put a small value of say 5s
> in there, the dial command will probably give up before someone answers at
> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no response.
> In the event it gets no TRYING response to a dial command within a
> specified timeout it should return control and flag an error. If on the
> other hand it does get a TRYING response (and maybe a RINGING too) it
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
> because Asterisk reads the extensions.conf on startup and also seems to
> cache what the host maps to on startup. Subsequent calls to the host
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Jonathan k. Creasy

2005-Dec-09 06:44 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Adam Robins

2005-Dec-09 07:58 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a 
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it 
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host 
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally privileged.
This transmission is sent in trust, for the sole purpose of delivery to the
intended recipient. If you have received this transmission in error, any use,
reproduction or dissemination of this transmission is strictly prohibited. If
you are not the intended recipient, please immediately notify the sender by
reply email and delete this message and its attachments, if any.

John Cianfarani

2005-Dec-09 08:09 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Ryan/Jonathan,

Couple quick questions regarding your setup?
Do you operate this in a strictly master/slave setup? 
Do you have anything(mon/ha's internal status/monitor options) that
actually monitors the asterisk process (to determine if it is hung). Or
is it only with total box failure to you fail over?
Do you use something to sync config/vm/cdr? Rsync/unison?

Thanks
John

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Douglas Garstang

2005-Dec-09 08:12 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-----Original Message-----
From: Adam Robins [mailto:arobins@PharmaCentra.com]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a 
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it 
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host 
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally privileged.
This transmission is sent in trust, for the sole purpose of delivery to the
intended recipient. If you have received this transmission in error, any use,
reproduction or dissemination of this transmission is strictly prohibited. If
you are not the intended recipient, please immediately notify the sender by
reply email and delete this message and its attachments, if any.


_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Douglas Garstang

2005-Dec-09 08:19 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Yes, that's a great question. I'm wondering the same thing. Can these
heartbeat apps monitor applications as well as network connectivity? The
heartbeat utility at www.linux-ha.org talks about monitoring some standard apps
like web servers and such but isn't clear about other apps... like Asterisk
or SER.

-----Original Message-----
From: John Cianfarani [mailto:jcianfarani@rogers.com]
Sent: Friday, December 09, 2005 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


Ryan/Jonathan,

Couple quick questions regarding your setup?
Do you operate this in a strictly master/slave setup? 
Do you have anything(mon/ha's internal status/monitor options) that
actually monitors the asterisk process (to determine if it is hung). Or
is it only with total box failure to you fail over?
Do you use something to sync config/vm/cdr? Rsync/unison?

Thanks
John

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second
failover. I think there is a article on voip-info.org about this, but
don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
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Adam Robins

2005-Dec-09 09:00 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Doug, 

We currently are using Digium TE410P boards directly into each Asterisk
server.  I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers.  I've looked at Cisco AS5400 ($$$$), Lucent APX 1000 ($$$), and
Quintum Tenor CMS ($$).

Thanks,
Adam

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Friday, December 09, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-----Original Message-----
From: Adam Robins [mailto:arobins@PharmaCentra.com]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a 
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it 
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host 
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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   http://lists.digium.com/mailman/listinfo/asterisk-users
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   http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential
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legally privileged. This transmission is sent in trust, for the sole
purpose of delivery to the intended recipient. If you have received this
transmission in error, any use, reproduction or dissemination of this
transmission is strictly prohibited. If you are not the intended
recipient, please immediately notify the sender by reply email and
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Ashley Wright

2005-Dec-09 09:04 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

Hi I use a allied telesyn at-vp730
Works quite well

ash

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam
Robins
Sent: 09 December 2005 16:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Doug, 

We currently are using Digium TE410P boards directly into each Asterisk
server.  I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers.  I've looked at Cisco AS5400 ($$$$), Lucent APX 1000 ($$$), and
Quintum Tenor CMS ($$).

Thanks,
Adam

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Friday, December 09, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-----Original Message-----
From: Adam Robins [mailto:arobins@PharmaCentra.com]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a 
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it 
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host 
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential
and are intended solely for addressee. The information may also be
legally privileged. This transmission is sent in trust, for the sole
purpose of delivery to the intended recipient. If you have received this
transmission in error, any use, reproduction or dissemination of this
transmission is strictly prohibited. If you are not the intended
recipient, please immediately notify the sender by reply email and
delete this message and its attachments, if any.


_______________________________________________
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Douglas Garstang

2005-Dec-09 09:06 UTC

head link

[Asterisk-Users] Asterisk Dial Failover

I think the Audiocodes boxes run at about $19,000 each.

-----Original Message-----
From: Adam Robins [mailto:arobins@PharmaCentra.com]
Sent: Friday, December 09, 2005 9:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


Doug, 

We currently are using Digium TE410P boards directly into each Asterisk
server.  I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers.  I've looked at Cisco AS5400 ($$$$), Lucent APX 1000 ($$$), and
Quintum Tenor CMS ($$).

Thanks,
Adam

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Friday, December 09, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-----Original Message-----
From: Adam Robins [mailto:arobins@PharmaCentra.com]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
burke@tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan
> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a> way to have Asterisk attempt to route the call to one OpenSER system,
and> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the> full 20 seconds before returning control. Also, This 20s includes the
time> is takes for the other end to answer, so if I put a small value of say
5s> in there, the dial command will probably give up before someone
answers at> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.> In the event it gets no TRYING response to a dial command within a 
> specified timeout it should return control and flag an error. If on
the> other hand it does get a TRYING response (and maybe a RINGING too) it 
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)> because Asterisk reads the extensions.conf on startup and also seems
to> cache what the host maps to on startup. Subsequent calls to the host 
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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_______________________________________________
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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential
and are intended solely for addressee. The information may also be
legally privileged. This transmission is sent in trust, for the sole
purpose of delivery to the intended recipient. If you have received this
transmission in error, any use, reproduction or dissemination of this
transmission is strictly prohibited. If you are not the intended
recipient, please immediately notify the sender by reply email and
delete this message and its attachments, if any.


_______________________________________________
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Dakota

2005-Dec-09 17:48 UTC

head link

[Asterisk-Users] Synthesized Voice for Asterisk

Are there any cool free software I can use to create automated voice message 
greetings for my PBX?

I want to customized some of my messages, however prefer to use a standard 
voice.

asterisk users - Dec 2005 - Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Asterisk Dial Failover

[Asterisk-Users] Synthesized Voice for Asterisk