[asterisk-bugs] [JIRA] (ASTERISK-8247) [bounty] feature request - QSIG call diversion interop with SIP
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Mon Jul 14 14:02:57 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-8247?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Matt Jordan updated ASTERISK-8247:
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Attachment: (was: c036841_ISO_IEC_13873_2003(E).pdf)
> [bounty] feature request - QSIG call diversion interop with SIP
> ---------------------------------------------------------------
>
> Key: ASTERISK-8247
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-8247
> Project: Asterisk
> Issue Type: New Feature
> Components: Core/General
> Reporter: Andrey Kovalenko
>
> We are interested in adding support to Asterisk of the following scenario:
> 1. SETUP message comes to Asterisk over QSIG trunk from PBX
> 2. Asterisk routes INVITE to SIP proxy
> 3. SIP proxy responds with a 302 Moved Temporarily with another number
> 4. Asterisk sends QSIG Call Diversion - CallRerouteing Forwarded Unconditionally - to the PBX. (FACILITY message CallRerouteing encoded per the attached ISO/IEC 13873:2003 document)
> 5. PBX places the call to the new destination and releases the call to Asterisk.
> This behavior is in compliance with the "Signalling Interworking between QSIG
> and SIP ? Call Diversion" (ISO/IEC 23915)
> ****** ADDITIONAL INFORMATION ******
> We can pay bounty to develop this feature. All sources will be GPLed. Please contact akovalen at gmail.com for details.
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