[Asterisk-Users] Fail over to Pri on VoIP connection failure
All, Thanks for the help. Checking on and changing the route based on dialstatus is the way to go. Thanks, ___ --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
Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
; call $ARG1 through nufone, failing over to the PRI.
[macro-nufone-dial]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],,go)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Congestion
exten = dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
It really is as simple as that. :-)
-A.
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RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure
Andrew,
Thanks for this - I have also been looking for a way to fail over
calls to a second SIP path, but;
In the event that the first attempt DOES NOT RESPOND (is down) there has
to be a timeout value to go to the next priority, correct? Otherwise the
channels just sits silent waiting for a response.
I think your macro assumes that you got a response from nufone, but what
if they were dead in the water?
Have I missed something?
Is there a way to modify the relevant SIP timer so if the INVITE is not
ack'd in a specific period of time then the next priority is executed?
Damon
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Friday, January 27, 2006 1:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection
failure
On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas
calls(busy
or just a hang-up). Is there a way in the dial plan to
automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
; call $ARG1 through nufone, failing over to the PRI.
[macro-nufone-dial]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],,go)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS
is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Congestion
exten = dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
It really is as simple as that. :-)
-A.
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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
On Friday 27 January 2006 16:00, Damon Estep wrote: In the event that the first attempt DOES NOT RESPOND (is down) there has to be a timeout value to go to the next priority, correct? Otherwise the channels just sits silent waiting for a response. That's what the qualify parameter in sip/iax.conf is for. Never terminate calls without it. :-) It won't *guarantee* that you'll never get dead air, but it sure goes a long way to ensuring that it happens so infrequently you'll think you misdialed. I think your macro assumes that you got a response from nufone, but what if they were dead in the water? Then qualify would have failed and Dial() would have immediately returned CHANUNAVAIL. -A. ___ --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
RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, January 27, 2006 2:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure On Friday 27 January 2006 16:00, Damon Estep wrote: In the event that the first attempt DOES NOT RESPOND (is down) there has to be a timeout value to go to the next priority, correct? Otherwise the channels just sits silent waiting for a response. That's what the qualify parameter in sip/iax.conf is for. Never terminate calls without it. :-) It won't *guarantee* that you'll never get dead air, but it sure goes a long way to ensuring that it happens so infrequently you'll think you misdialed. I think your macro assumes that you got a response from nufone, but what if they were dead in the water? Then qualify would have failed and Dial() would have immediately returned CHANUNAVAIL. -A. OK - starting to make sense now Qualify=yes for the peer in sip.conf If you have qualify=yes I assume that triggers a sip query to get channel capabilities from the peer? What is the qualify timeout? Can it be manipulated? If the goal was strictly to try one provider, and if the channel fails qualify, then try the next, is the macro you posted needed? Couldn't you just; Exten = ,1,Dial(SIP/[EMAIL PROTECTED] Exten = ,2,Dial(SIP/[EMAIL PROTECTED] Exten = ,3,Congestion(15) Exnte = ,4,Hangup ___ --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
Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
On Friday 27 January 2006 16:24, Damon Estep wrote: If you have qualify=yes I assume that triggers a sip query to get channel capabilities from the peer? What is the qualify timeout? Can it be manipulated? qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for a response. If it does not receive one within 1000ms (by default) and qualifysmoothing is not enabled, it will flag the peer as UNREACHABLE which means that any attempts to Dial() the peer will fail immediately with CHANUNAVAIL. Asterisk continues to send these pings until it receives a response within the accepted timeframe and once it gets responses again it will flag the peer as being available once again. There are some other tuning parameters which can be used to modify this behaviour slightly but this is what qualify does in a nutshell. If the goal was strictly to try one provider, and if the channel fails qualify, then try the next, is the macro you posted needed? Correct. Couldn't you just; Exten = ,1,Dial(SIP/[EMAIL PROTECTED] Exten = ,2,Dial(SIP/[EMAIL PROTECTED] Exten = ,3,Congestion(15) Exnte = ,4,Hangup Well I've never been a fan of just letting things fall off the edge and expecting them to work reliably. I use the 'g' Dial() option so that I can handle failover and call completion correctly or properly -- instead of just letting it do whatever svn trunk deems right at this point I specifically do things based on how the call terminated. It's just a nicer way of doing what you've provided, and ends up being more robust to code policy changes. -A. ___ --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
RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, January 27, 2006 2:45 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure On Friday 27 January 2006 16:24, Damon Estep wrote: If you have qualify=yes I assume that triggers a sip query to get channel capabilities from the peer? What is the qualify timeout? Can it be manipulated? qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for a response. If it does not receive one within 1000ms (by default) and qualifysmoothing is not enabled, it will flag the peer as UNREACHABLE which means that any attempts to Dial() the peer will fail immediately with CHANUNAVAIL. Asterisk continues to send these pings until it receives a response within the accepted timeframe and once it gets responses again it will flag the peer as being available once again. There are some other tuning parameters which can be used to modify this behaviour slightly but this is what qualify does in a nutshell. Since your original hint on qualify=yes have been hunting for the parameter tuning capabilities of this feature - to no avail. Are you aware of any reference anywhere on tuning the qualify frequency and timeout? I assume this (tuning) does not require code changes. Correct? If the goal was strictly to try one provider, and if the channel fails qualify, then try the next, is the macro you posted needed? Correct. Couldn't you just; Exten = ,1,Dial(SIP/[EMAIL PROTECTED] Exten = ,2,Dial(SIP/[EMAIL PROTECTED] Exten = ,3,Congestion(15) Exnte = ,4,Hangup Well I've never been a fan of just letting things fall off the edge and expecting them to work reliably. I use the 'g' Dial() option so that I can handle failover and call completion correctly or properly -- instead of just letting it do whatever svn trunk deems right at this point I specifically do things based on how the call terminated. It's just a nicer way of doing what you've provided, and ends up being more robust to code policy changes. Sounds like words of wisdom to me :) Thanks a million D ___ --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
[Asterisk-Users] Fail over to Pri on VoIP connection failure
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the innocent)
-- Executing Dial(Zap/47-1,
IAX2/VoIPServicePrividerOUT/011) in new stack
-- Called VoIPServicePrividerOUT/011
-- Call accepted by 72.34.43.5 (format g729)
-- Format for call is g729
-- Channel 0/23, span 2 got hangup request here I get a busy
signal
-- Hungup 'IAX2/ VoIPServicePrividerOUT-1'
[Outbound context]
exten = _9011.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9011.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9011.,3,Macro(outisbusy) ; No available circuits
exten = _918.,1,Macro(dialout-trunk,2,${EXTEN:1},); 800 numbers to the
PRI
exten = _918.,2,Macro(outisbusy) ; No available circuits
exten = _9Z.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9Z.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9Z.,3,Macro(outisbusy); No available circuits
Richard
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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
I know this may be a backwards way but for several
reasons I have asterisk send all calls thru astcc.
With astcc you specify multiple routes with prioroty
settings. If it cant complete a call with one route it
will roll over and use the next one.
Regards,
Dovid
--- Cavanna, Richard [EMAIL PROTECTED] wrote:
I am trying to tweak my dial plan and I am running
into a problem.
Sometimes my VoIP out bound calls do not complete on
overseas calls(busy
or just a hang-up). Is there a way in the dial plan
to automatically
dial out of my PRI when something like this happens.
Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the innocent)
-- Executing Dial(Zap/47-1,
IAX2/VoIPServicePrividerOUT/011) in
new stack
-- Called VoIPServicePrividerOUT/011
-- Call accepted by 72.34.43.5 (format g729)
-- Format for call is g729
-- Channel 0/23, span 2 got hangup request
here I get a busy
signal
-- Hungup 'IAX2/ VoIPServicePrividerOUT-1'
[Outbound context]
exten = _9011.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9011.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9011.,3,Macro(outisbusy); No available
circuits
exten = _918.,1,Macro(dialout-trunk,2,${EXTEN:1},);
800 numbers to the
PRI
exten = _918.,2,Macro(outisbusy) ; No available
circuits
exten = _9Z.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9Z.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9Z.,3,Macro(outisbusy) ; No available
circuits
Richard
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