Re: [asterisk-dev] pjsip vs cel
On Wed, Jun 10, 2015 at 11:30 PM, James Cloos cl...@jhcloos.com wrote: RM == Richard Mudgett rmudg...@digium.com writes: RM You can specify the dialplan context incoming calls go to when defining RM the endpoint. Obviously I've done that. The issue is that the CHAN_START cel event does not reflect the specified context and the INVITE's ruri like chan_sip's CHAN_START event does, even though the subsequent events do. I thought I had explained that clearly; apologies for missing any ambiguity in my note. It is a bug. When we allocate a channel in chan_pjsip, we are passing empty strings into ast_channel_alloc_with_endpoint for the extension and context parameters. The act of creating the channel in ast_channel_alloc will, when the routine is finished, publish the existence of the channel to the 'world' via Stasis, which will create the CHAN_START event in CEL. We set the context/extension later on in chan_pjsip_new after the channel has been created; it should be trivial to refactor that to pass that information into ast_channel_alloc_with_endpoint. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] pjsip vs cel
On Wed, Jun 10, 2015 at 6:22 PM, James Cloos cl...@jhcloos.com wrote: I've finally gotten a box setup with pjsip. I see that the CHAN_START cel event logs exten='s', context='default' instead of the INVITEd extension and the endpoint's context. All of the rest of the events log the expected data. This is unlike chan_sip. Is this a bug? Or an expected behavorial difference beteen chan_sip and res_pjsip? For reference, this particular box does not have a context named default. And I do not see any way to tell res_pjsip what the default context should be, like sip.conf's [general] section. You can specify the dialplan context incoming calls go to when defining the endpoint. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] pjsip vs cel
I've finally gotten a box setup with pjsip. I see that the CHAN_START cel event logs exten='s', context='default' instead of the INVITEd extension and the endpoint's context. All of the rest of the events log the expected data. This is unlike chan_sip. Is this a bug? Or an expected behavorial difference beteen chan_sip and res_pjsip? For reference, this particular box does not have a context named default. And I do not see any way to tell res_pjsip what the default context should be, like sip.conf's [general] section. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] pjsip vs cel
RM == Richard Mudgett rmudg...@digium.com writes: RM You can specify the dialplan context incoming calls go to when defining RM the endpoint. Obviously I've done that. The issue is that the CHAN_START cel event does not reflect the specified context and the INVITE's ruri like chan_sip's CHAN_START event does, even though the subsequent events do. I thought I had explained that clearly; apologies for missing any ambiguity in my note. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
