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<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=elguero" class="activity-item-user activity-item-author">Michael L. Young</a> commented on <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24221">ASTERISK-24221 -...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

What is the "port" column set to in the table for this realtime peer voxout?

I just saw the INSERT up above in your prior comment... I think I see the problem...

The reason why an A lookup is performed instead of the SRV is commented in the code:

Read more chan_sip: SRV lookup is not performed when using a realtime peerhttp://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=rnewton" class="activity-item-user activity-item-author">Rusty Newton</a> changed the status to Waiting for Feedback on <a href="https://issues.asterisk.org/jira/browse...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

I just tested 11 svn 423210 vs 12 svn 423172. The behavior appears identical.

Note I had to make some changes to your configuration to attempt reproduction.

In queues.conf:

joinempty=yes leavewhenempty=no

and the dialplan needed an answer:

[callmenow] exten = Read more Queue variables not passed to local channelhttp://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=elguero" class="activity-item-user activity-item-author">Michael L. Young</a> commented on <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24332">ASTERISK-24332 -...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

The "fromstring" is a global setting, not an individual mailbox setting. Your patch changes the expected behavior. Also, I think that the patch would break things for those who using a standard configuration file.

Where is this "fromstring" being

Read more Realtime ODBC Voicemail is not using the "fromstring" parameterhttp://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=rnewton" class="activity-item-user activity-item-author">Rusty Newton</a> updated the Description of <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24334">ASTERISK...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

Essentially the same configuration as ASTERISK-24205, but attempting to call from a local Digium phone configured as a SIP peer to the SIPML5 client in Chrome (which resides on another machine on local network).

The crash!:

D Phone -> Box1(Asterisk) -

Read more Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5)http://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=rnewton" class="activity-item-user activity-item-author">Rusty Newton</a> commented on <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24205">ASTERISK-24205 - DTLS...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

Aleksei Kulakov I tested with your config and got the same behavior as I did previously. There likely is something different about the environment or topology. Can you post more details on your SIPML5 configuration and network topology concerning the

Read more DTLS-SRTP fails on SIP over WebSockets call from SIPML5(chrome) to Asteriskhttp://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=rnewton" class="activity-item-user activity-item-author">Rusty Newton</a> updated the Description of <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24334">ASTERISK...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

Essentially the same configuration as ASTERISK-24205, but attempting to call from a local Digium phone configured as a SIP peer to the SIPML5 client in Chrome (which resides on another machine on local network).

The crash!:

D Phone -> Asterisk -> Chrome(SIPML5)

Read more Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5)http://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=rnewton" class="activity-item-user activity-item-author">Rusty Newton</a> updated the Description of <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24334">ASTERISK...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

Essentially the same configuration as ASTERISK-24205, but attempting to call from a local Digium phone configured as a SIP peer to the SIPML5 client in Chrome (which resides on another machine on local network).

The crash!:

D Phone -> Asterisk -> Chrome(SIPML5)

Read more Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5)http://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

<a href="https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=rnewton" class="activity-item-user activity-item-author">Rusty Newton</a> created <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24334">ASTERISK-24334 - Crash with...

Asterisk Issue Tracker - Tue, 2014-09-16 21:34

Essentially the same configuration as ASTERISK-24205, but attempting to call from a local Digium phone configured as a SIP peer to the SIPML5 client in Chrome (which resides on another machine on local network).

The crash!:

D Phone -> Asterisk -> Chrome(SIPML5)

Read more Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5)http://streams.atlassian.com/syndication/types/issue-0500
Categories: Asterisk Project

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