[asterisk-users] Asterisk 13.12.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.12.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.12.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: --- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.24.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.24.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.24.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: --- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 14.1.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 14.1.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.1.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: --- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] info about DID.
Hi! I need to make a dialplan by DID. where it gets the asterisk values did? from sip headers or ... ? Thanks! -- KyD GNU/Linux SysAdmin Quanto mais você sabe, mais você percebe que você não sabe nada. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime queue & agent groups
Hello I'm a bit confused on how to group agents (give agents a group number) when using realtime queues. I read on the wiki : * If you include groups in your queue definition the calls get routed in the order of the group regardless of the specified strategy. So I just have a member= line for each agent. member => Agent/@1 ; a group member => Agent/501 ; a single agent member => Agent/:1,1 ; Any agent in group 1, wait for first available, but consider with penalty In my realtime database I have table queue_members : +--++-++-+-++ | uniqueid | membername | queue_name | interface | state_interface | penalty | paused | +--++-++-+-++ | 2916 | testacc77000 | queue7700q4 | testacc77000 | | 0 | NULL | | 2917 | testacc77001 | queue7700q4 | testacc77001 | | 3 | NULL | | 2843 | testacc77000 | queue7700q4 | testacc77000 | | 0 | NULL | | 2905 | testacc7700905 | queue7700q5 | testacc7700905 | | 0 | NULL | | 2888 | testacc77000 | queue7700q5 | testacc77000 | | 0 | NULL | | 2900 | testacc77000 | queue7700q5 | testacc77000 | | 0 | NULL | | 2901 | testacc77001 | queue7700q5 | testacc77001 | | 0 | NULL | How do I define a group to a certain agent/member in this case ? Kind regards J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: recommended helpdesk OSS with Asterisk integration
hi, can you recommend open source helpdesk solution with working Asterisk integration? marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call by DID
On Wednesday 26 Oct 2016, KyD wrote:
> Hi,
>
> My sip provider gave me 2 numbers for the incoming call via pstn.
>
> nro1 = 12341234
> nro2 = 45674567
>
> I have a dialplan for each.
> if i put this on my dialplan:
>
> exten => s,1,Dial(SIP/1001)
> exten => Hangup()
>
> Works!
>
> But if i put one of them:
>
> exten => 12341234,1,Dial(SIP/1001)
> exten => _1234,1,Dial(SIP/1001)
> exten => 45674567,1,Dial(SIP/1001)
> exten => _4567,1,Dial(SIP/1001)
>
> incoming calls do not arrive.
>
> Any ideas?
The incoming call must be arriving with ${EXTEN} containing something that
doesn't match 12341234, _1234, 45674567 or _4567, so it is
not triggering any of the extensions in your dialplan. Maybe it still has the
STD code or even the IDD code prepended. (Been caught this way once before
. our old ISDN-30 provider used to send just the local number, then we
moved to a new ISDN-30 provider who send the number with STD code but no
initial 0. Cue frantic editing of dialplan before rest of staff arrived .)
So try this;
exten => s,1,NoOp(Incoming call for '${EXTEN}')
exten => s,n,Dial(SIP/1001)
exten => s,n,Hangup()
Run `# asterisk -vvvr`, dial one of your DDI numbers from
a mobile phone and watch the messages scrolling past.
Now you will be seeing exactly what ${EXTEN} contains when a call comes in, so
you should be able to work out what is going on, and craft your extension
expressions to suit. If in doubt, post an excerpt from your CLI output.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
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