Dialout redundancy using this method works perfect. I've been using this
method for some time now. I currently have two IAX2 providers and plan to get
another backup as well (In addition to me getting my Digium cards tomorrow
that'll be another backup.)
That's great for outgoing calls, but... I'm trying to figure out the
best approach to use for incoming calls.
I currently have a VP phone number, it's the only incoming number I have for
the other voip providers I have don't offer local termination (or any at all
for that matter).
We have a POTS line from Verizon and we'd like to continue using that phone
number.
Originally we were just going to forward that phone number to VP. But what
happens if VP goes down? I figure in that case (and we'd have to get in
touch with VP if they will forward to another number if they're done), to
then forward to another voip / pots line that we have.
Is there any other approach we can use to do this?
Possibly, a service that'll offer something like:
Transfer to 1609xxxxxxx but if busy, forward to 1609xxxxxxx, etc. and so on?
In addition does anyone know where I might be able to port my number to that
supports transferring instead of forwarding?
I currently have Verizon and they said we need a CustoFlex plan which will only
support 6 "forwards" so if 7 callers call in, the 7th will get a busy
signal.
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Brent Franks
Sent: Sunday, February 08, 2004 3:15 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] dialout redunancy.
You will need to set priorities for each one.
For example:
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Playback(pstnallbusy)
exten =>
_91NXXNXXXXXX,3,Dial,IAX2/[PROTECTED]@voicepulse/${EXTEN:${TRUNKMSD}}
exten => _91NXXNXXXXXX,4,Congestion
Basically what happens here, is I try to put it out on the Verizon POTS
lines first, then if that doesn't work, I play a message saying all the
lines are busy, hold if the call is important (it's now billable), the
user holds, and it goes to voicepulse.
You could get rid of the All Busy message if you wanted, I just like to
know that the call is going to be billed (since I have unlimited LD on
my POTS lines). If that fails, It plays a fast busy.
You can also do a qualify in your iax.conf and sip entries to know
whether they are reachable before trying the call. Read up on qualify to
find out how to do it for your needs.
Brent
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of John Bittner
Sent: Sunday, February 08, 2004 2:37 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] dialout redunancy.
Hi,
?
How do I get asterisk to use an alternate outbound provider in the event
my primary IAX provider goes down. I currently have an IAX provider that
is having issues, so I signed up with a sip provider for a backup. I
added the sip provider info into the extensions.conf file as the second
outbound entry, but asterisk still tries to call the iax provider
1st?and since the call is incomplete the end-user hangs up.?Any ideas
would be?helpful.
?
Thanks
?
John Bittner
Simlab.net
?
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I got it working by configuring qualify in my iax.conf. I guess asterisk didn't think the IAX provider was down until I added that line. As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on voicepulse and 1 on voiceglo. This way if voicepulse is down it will route the call to voiceglo. Hopefully as the voip providers get better they will offer a forwarding feature. Vonage does. John Bittner Simlab.net> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Matthew B Marlowe > Sent: Sunday, February 08, 2004 5:45 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] dialout redundancy. > > Dialout redundancy using this method works perfect. I've > been using this method for some time now. I currently have > two IAX2 providers and plan to get another backup as well (In > addition to me getting my Digium cards tomorrow that'll be > another backup.) > > That's great for outgoing calls, but... I'm trying to figure > out the best approach to use for incoming calls. > > I currently have a VP phone number, it's the only incoming > number I have for the other voip providers I have don't offer > local termination (or any at all for that matter). > > We have a POTS line from Verizon and we'd like to continue > using that phone number. > > Originally we were just going to forward that phone number to > VP. But what happens if VP goes down? I figure in that case > (and we'd have to get in touch with VP if they will forward > to another number if they're done), to then forward to > another voip / pots line that we have. > > Is there any other approach we can use to do this? > > Possibly, a service that'll offer something like: > > Transfer to 1609xxxxxxx but if busy, forward to 1609xxxxxxx, > etc. and so on? > > In addition does anyone know where I might be able to port my > number to that supports transferring instead of forwarding? > > I currently have Verizon and they said we need a CustoFlex > plan which will only support 6 "forwards" so if 7 callers > call in, the 7th will get a busy signal. > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Brent Franks > Sent: Sunday, February 08, 2004 3:15 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] dialout redunancy. > > You will need to set priorities for each one. > > For example: > > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91NXXNXXXXXX,2,Playback(pstnallbusy) > exten => > _91NXXNXXXXXX,3,Dial,IAX2/[PROTECTED]@voicepulse/${EXTEN:${TRUNKMSD}} > exten => _91NXXNXXXXXX,4,Congestion > > Basically what happens here, is I try to put it out on the > Verizon POTS > lines first, then if that doesn't work, I play a message > saying all the > lines are busy, hold if the call is important (it's now billable), the > user holds, and it goes to voicepulse. > > You could get rid of the All Busy message if you wanted, I > just like to > know that the call is going to be billed (since I have unlimited LD on > my POTS lines). If that fails, It plays a fast busy. > > You can also do a qualify in your iax.conf and sip entries to know > whether they are reachable before trying the call. Read up on > qualify to > find out how to do it for your needs. > > Brent > > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > John Bittner > Sent: Sunday, February 08, 2004 2:37 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] dialout redunancy. > > Hi, > ? > How do I get asterisk to use an alternate outbound provider > in the event > my primary IAX provider goes down. I currently have an IAX > provider that > is having issues, so I signed up with a sip provider for a backup. I > added the sip provider info into the extensions.conf file as > the second > outbound entry, but asterisk still tries to call the iax provider > 1st?and since the call is incomplete the end-user hangs up.?Any ideas > would be?helpful. > ? > Thanks > ? > John Bittner > Simlab.net > ? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
You're toll-free number automatically forwards to the next number if one is busy? Cool. I wasn't sure if it would do that. I know VP reports a fast busy. Don't know what Voiceglo reports. What toll-free provider do you have out of curiosity? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of John Bittner Sent: Sunday, February 08, 2004 6:12 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] dialout redunancy. I got it working by configuring qualify in my iax.conf. I guess asterisk didn't think the IAX provider was down until I added that line. As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on voicepulse and 1 on voiceglo. This way if voicepulse is down it will route the call to voiceglo. Hopefully as the voip providers get better they will offer a forwarding feature. Vonage does. John Bittner Simlab.net> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Matthew B Marlowe > Sent: Sunday, February 08, 2004 5:45 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] dialout redundancy. > > Dialout redundancy using this method works perfect. I've > been using this method for some time now. I currently have > two IAX2 providers and plan to get another backup as well (In > addition to me getting my Digium cards tomorrow that'll be > another backup.) > > That's great for outgoing calls, but... I'm trying to figure > out the best approach to use for incoming calls. > > I currently have a VP phone number, it's the only incoming > number I have for the other voip providers I have don't offer > local termination (or any at all for that matter). > > We have a POTS line from Verizon and we'd like to continue > using that phone number. > > Originally we were just going to forward that phone number to > VP. But what happens if VP goes down? I figure in that case > (and we'd have to get in touch with VP if they will forward > to another number if they're done), to then forward to > another voip / pots line that we have. > > Is there any other approach we can use to do this? > > Possibly, a service that'll offer something like: > > Transfer to 1609xxxxxxx but if busy, forward to 1609xxxxxxx, > etc. and so on? > > In addition does anyone know where I might be able to port my > number to that supports transferring instead of forwarding? > > I currently have Verizon and they said we need a CustoFlex > plan which will only support 6 "forwards" so if 7 callers > call in, the 7th will get a busy signal. > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Brent Franks > Sent: Sunday, February 08, 2004 3:15 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] dialout redunancy. > > You will need to set priorities for each one. > > For example: > > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91NXXNXXXXXX,2,Playback(pstnallbusy) > exten => > _91NXXNXXXXXX,3,Dial,IAX2/[PROTECTED]@voicepulse/${EXTEN:${TRUNKMSD}} > exten => _91NXXNXXXXXX,4,Congestion > > Basically what happens here, is I try to put it out on the > Verizon POTS > lines first, then if that doesn't work, I play a message > saying all the > lines are busy, hold if the call is important (it's now billable), the > user holds, and it goes to voicepulse. > > You could get rid of the All Busy message if you wanted, I > just like to > know that the call is going to be billed (since I have unlimited LD on > my POTS lines). If that fails, It plays a fast busy. > > You can also do a qualify in your iax.conf and sip entries to know > whether they are reachable before trying the call. Read up on > qualify to > find out how to do it for your needs. > > Brent > > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > John Bittner > Sent: Sunday, February 08, 2004 2:37 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] dialout redunancy. > > Hi, > ? > How do I get asterisk to use an alternate outbound provider > in the event > my primary IAX provider goes down. I currently have an IAX > provider that > is having issues, so I signed up with a sip provider for a backup. I > added the sip provider info into the extensions.conf file as > the second > outbound entry, but asterisk still tries to call the iax provider > 1st?and since the call is incomplete the end-user hangs up.?Any ideas > would be?helpful. > ? > Thanks > ? > John Bittner > Simlab.net > ? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users