Re: [asterisk-users] PRI failover to SIP trunk
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] PRI failover to SIP trunk
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
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Re: [asterisk-users] PRI failover to SIP trunk
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] PRI failover to SIP trunk
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
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Re: [asterisk-users] PRI failover to SIP trunk
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
True I guess except that if the call fails as the OP posted, because the PRI
is down, it should work then right?
Another thing. For outbound calls, I do not have a timeout. So the user
hangs up when they are ready, or when the other side hangs up or gets
congestion, which amounts to the h exten, or am I not correct.
Why have a timeout on outbound dialing (unless you are a dialer app?) It is
not like voicemail where you want it to ring for so many seconds and then
roll to VM.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] PRI failover to SIP trunk
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
True I guess except that if the call fails as the OP posted, because the PRI
is down, it should work then right?
I believe so, I haven't tried it. I imagine DIALSTATUS would be either
CHANUNAVAIL or CONGESTION.
Another thing. For outbound calls, I do not have a timeout. So the user
hangs up when they are ready, or when the other side hangs up or gets
congestion, which amounts to the h exten, or am I not correct.
I can't answer to the use of the h exten, I've never used it.
Why have a timeout on outbound dialing (unless you are a dialer app?) It is
not like voicemail where you want it to ring for so many seconds and then
roll to VM.
You usually wouldn't use a timeout for outbound PSTN calls. I only
mentioned it to try to be as complete as possible.
-Dave
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[asterisk-users] PRI failover to SIP trunk
Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service, however we have been experiencing some issues where one or both of them can fail randomly. We are working with the telephone company to have this resolved. In the meantime, we want to have a good failover solution where if both PRIs fail, asterisk will dial out through the SIP trunk. I've found solutions as simple as two Dial commands one after the other, and others where the failover Dial is in a jump to CONGESTION. Unfortunately we don't have a testing environment, so the solution really has to work. Does anyone else on the list have a PRI to VoIP failover setup that's worked for them in a high volume environment? Thanks! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI failover to SIP trunk
On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin jmar...@metrixmatrix.comwrote:
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2
PRIs from the local telephone company in one group (g1) and a SIP
trunk from bandwidth.com. The PRIs are the primary outgoing service,
however we have been experiencing some issues where one or both of
them can fail randomly. We are working with the telephone company to
have this resolved.
In the meantime, we want to have a good failover solution where if
both PRIs fail, asterisk will dial out through the SIP trunk. I've
found solutions as simple as two Dial commands one after the other,
and others where the failover Dial is in a jump to CONGESTION.
Unfortunately we don't have a testing environment, so the solution
really has to work.
Does anyone else on the list have a PRI to VoIP failover setup that's
worked for them in a high volume environment?
Thanks!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400
Simple enough,
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
That is if Zap/DAHDI completely craps out. If the dialplan/Asterisk thinks
it is working it will hang.
If totally out of commission, then the second priority gets called.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] PRI failover to SIP trunk
On Thu, Jul 9, 2009 at 5:31 PM, Steve Totaro stot...@first-notification.com
wrote:
On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin jmar...@metrixmatrix.comwrote:
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2
PRIs from the local telephone company in one group (g1) and a SIP
trunk from bandwidth.com. The PRIs are the primary outgoing service,
however we have been experiencing some issues where one or both of
them can fail randomly. We are working with the telephone company to
have this resolved.
In the meantime, we want to have a good failover solution where if
both PRIs fail, asterisk will dial out through the SIP trunk. I've
found solutions as simple as two Dial commands one after the other,
and others where the failover Dial is in a jump to CONGESTION.
Unfortunately we don't have a testing environment, so the solution
really has to work.
Does anyone else on the list have a PRI to VoIP failover setup that's
worked for them in a high volume environment?
Thanks!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400
Simple enough,
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
That is if Zap/DAHDI completely craps out. If the dialplan/Asterisk thinks
it is working it will hang.
If totally out of commission, then the second priority gets called.
Let me clarify that I think that is how it works. Been a long time.
Maybe it was the old N+101 trick? Not sure why that was ever deprecated.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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