Re: [asterisk-users] PSTN failover
On Tuesday 16 October 2007 06:21:45 Andrew Kohlsmith wrote:
GotoIf($[${DIALSTATUS} = BUSY]?busy)
GotoIf($[${DIALSTATUS} = NOANSWER]?noanswer)
GotoIf($[${DIALSTATUS} = ANSWERED]?answered)
Dial(Zap/...)
Of course, I do this inside a macro, and I emit correct CDR and correct
hangupcauses for those who use my system.
Dialing twice like that without checking your return value is an invitation
for future problems.
Well, as far as i have tried - i never get ANSWERED in DIALSTATUS. Only thing
that continues is h extension.
Regards,
Atis
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Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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Re: [asterisk-users] PSTN failover
- Original Message - From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 16, 2007 1:20 AM Subject: Re: [asterisk-users] PSTN failover Dovid B wrote: Chanisavail does not work well for this. I would use priority jumping (n+101). Using ChanIsAvail with the 's' option is supposed to assume a SIP channel is occupied if it's in use ANYWHERE under asterisk's wing. For clarification, Dovid, have your poor experiences occurred with the 's' or without? I tell myself I've been meaning to switch away from priority jumping into using chanisavail, but always just assumed that the 's' options was all I would need to use when it came time It was not with s. I was with Exten = _X.,ChanisAvail(.. ___ --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] PSTN failover
- Original Message - From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 16, 2007 1:12 AM Subject: Re: [asterisk-users] PSTN failover On Tue, 16 Oct 2007, Dovid B wrote: Chanisavail does not work well for this. I would use priority jumping (n+101). Why not? I have seen where it will go to n+101 when the trunk was in use as well as the carrier may be up but it may reject the call. First see if they take the call if not we can always send it else where. ___ --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] PSTN failover
On Tuesday 16 October 2007 03:49:37 Atis Lezdins wrote: Well, as far as i have tried - i never get ANSWERED in DIALSTATUS. Only thing that continues is h extension. You must of course use 'g' in the Dial flags so that it continues on in the dialplan after hangup... -A. ___ --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
[asterisk-users] PSTN failover
Hi, Does anyone have any advice in how to implement PSTN failover should an internet connection for IAX trunking go down? to route outbound to analog lines Can this be written into the dialplan using a GotoIf statement by testing the whether the internet connection is up, or from a IAX/SIP response? Thanks Robert McNaught ___ --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] PSTN failover
On Mon, 15 Oct 2007, Robert McNaught wrote: Does anyone have any advice in how to implement PSTN failover should an internet connection for IAX trunking go down? to route outbound to analog lines Can this be written into the dialplan using a GotoIf statement by testing the whether the internet connection is up, or from a IAX/SIP response? In the dial plan, you can simply have a second instruction with another Dial() command that will only be hit if the first Dial() command fails. This is automatic behaviour. Or, if you prefer, you can use ChanAvail() to check if the SIP / IAX trunk is up and that reachability through it has been verified through 'qualify=' (if you have this turned on), and in the event that it is not, use GotoIf() to route out to the PSTN via analog lines, like you said. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] PSTN failover
Chanisavail does not work well for this. I would use priority jumping (n+101). - Original Message - From: Alex Balashov [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 15, 2007 6:47 PM Subject: Re: [asterisk-users] PSTN failover On Mon, 15 Oct 2007, Robert McNaught wrote: Does anyone have any advice in how to implement PSTN failover should an internet connection for IAX trunking go down? to route outbound to analog lines Can this be written into the dialplan using a GotoIf statement by testing the whether the internet connection is up, or from a IAX/SIP response? In the dial plan, you can simply have a second instruction with another Dial() command that will only be hit if the first Dial() command fails. This is automatic behaviour. Or, if you prefer, you can use ChanAvail() to check if the SIP / IAX trunk is up and that reachability through it has been verified through 'qualify=' (if you have this turned on), and in the event that it is not, use GotoIf() to route out to the PSTN via analog lines, like you said. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 ___ --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] PSTN failover
On Tue, 16 Oct 2007, Dovid B wrote: Chanisavail does not work well for this. I would use priority jumping (n+101). Why not? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] PSTN failover
Dovid B wrote: Chanisavail does not work well for this. I would use priority jumping (n+101). Using ChanIsAvail with the 's' option is supposed to assume a SIP channel is occupied if it's in use ANYWHERE under asterisk's wing. For clarification, Dovid, have your poor experiences occurred with the 's' or without? I tell myself I've been meaning to switch away from priority jumping into using chanisavail, but always just assumed that the 's' options was all I would need to use when it came time Thanks! ___ --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] PSTN failover
Alex Balashov wrote: On Tue, 16 Oct 2007, Dovid B wrote: Chanisavail does not work well for this. I would use priority jumping (n+101). Why not? Priority jumping is no solution to failover, it's just an ugly hack. ;) I'd basically just Dial() 2 times: Dial(SIP/...); Dial(Zap/...); No need for priority jumping. And not need to check if the ChanIsAvail(). Just Dial(). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] PSTN failover
I don't really understand how ChanIsAvail() can be of any use. Even if it tells you that the channel is available there's no guarantee that the call will go through. And moreover between the ChanIsAvail() check and the Dial() command someone else could have taken the channel. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] PSTN failover
On Monday 15 October 2007 19:50:03 Philipp Kempgen wrote:
I'd basically just Dial() 2 times:
Dial(SIP/...);
Dial(Zap/...);
No need for priority jumping. And not need to check if
the ChanIsAvail(). Just Dial().
Why not just do it the correct way?
Dial(SIP/,,g)
GotoIf($[${DIALSTATUS} = BUSY]?busy)
GotoIf($[${DIALSTATUS} = NOANSWER]?noanswer)
GotoIf($[${DIALSTATUS} = ANSWERED]?answered)
Dial(Zap/...)
Of course, I do this inside a macro, and I emit correct CDR and correct
hangupcauses for those who use my system.
Dialing twice like that without checking your return value is an invitation
for future problems.
-A.
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