Re: [asterisk-dev] [Code Review] 3575: config: Fix config files not reloading when only an included file changes.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3575/#review12022 --- I didn't read the code, but: > Made sip.conf include several s

[asterisk-dev] dial url with sip

Looking at app_dial.c and chan_sip.c, I get the impression that the url in a dial string cannot get sent as part of the sip INVITE, yes? (I base that on sip_sendhtml().) Am I reading chan_sip correctly? Will I need to change sip_sendhtml() to send the url as part of the INVITE? A test call show

Re: [asterisk-dev] dial url with sip

If you are in control of the SIP-Phone, you could pass additional information via SIPAddHeader in your dialplan. On Jun 2, 2014 10:33 AM, "James Cloos" wrote: > Looking at app_dial.c and chan_sip.c, I get the impression that the url > in a dial string cannot get sent as part of the sip INVITE, ye

[asterisk-dev] [Code Review] 3579: odbc: Fix for remove fixed size buffers fix (r414968).

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3579/ --- Review request for Asterisk Developers and Joshua Colp. Repository: Asteri

Re: [asterisk-dev] [Code Review] 3579: odbc: Fix for remove fixed size buffers fix (r414968).

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3579/#review12023 --- Ship it! Ship It! - Joshua Colp On June 2, 2014, 10:23 a.m.

Re: [asterisk-dev] [Code Review] 3579: odbc: Fix for remove fixed size buffers fix (r414968).

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3579/ --- (Updated June 2, 2014, noon) Review request for Asterisk Developers and Jo

Re: [asterisk-dev] dial url with sip

On Mon, Jun 2, 2014 at 3:39 AM, Dennis Guse wrote: > If you are in control of the SIP-Phone, you could pass additional > information via SIPAddHeader in your dialplan. > > On Jun 2, 2014 10:33 AM, "James Cloos" wrote: >> >> Looking at app_dial.c and chan_sip.c, I get the impression that the url >

Re: [asterisk-dev] [Code Review] 3579: odbc: Fix for remove fixed size buffers fix (r414968).

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3579/#review12024 --- Ship it! Just checked the differences between this patch and t

Re: [asterisk-dev] [Code Review] 3579: odbc: Fix for remove fixed size buffers fix (r414968).

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3579/#review12025 --- /branches/1.8/funcs/func_odbc.c

[asterisk-dev] [Code Review] 3580: Handle 183 without SDP - don't always convert to ringing

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3580/ --- Review request for Asterisk Developers. Repository: Asterisk Description

Re: [asterisk-dev] [Code Review] 3579: odbc: Fix for remove fixed size buffers fix (r414968).

> On June 2, 2014, 1:47 p.m., Tilghman Lesher wrote: > > /branches/1.8/funcs/func_odbc.c, line 931 > > > > > > Red blob, while you're in the neighborhood. I'll remember to touch that when committing. > On June 2,

Re: [asterisk-dev] [Code Review] 3575: config: Fix config files not reloading when only an included file changes.

> On June 2, 2014, 2:04 a.m., wdoekes wrote: > > I didn't read the code, but: > > > > > Made sip.conf include several specific files. The config was reloaded > > > when any file was touched. > > > > What about includes in included files? > > > > It's legal to do this right? > > > > sip.co

Re: [asterisk-dev] [Code Review] 3541: res_http_websocket: Create a websocket client

> On May 31, 2014, 4:06 a.m., wdoekes wrote: > > trunk/main/uri.c, lines 183-187 > > > > > > I'm not seeing that with the git version. And what happens when someone > > *does* want X in the URL? > > > > >

Re: [asterisk-dev] [Code Review] 3541: res_http_websocket: Create a websocket client

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3541/ --- (Updated June 2, 2014, 10:57 a.m.) Review request for Asterisk Developers

Re: [asterisk-dev] dial url with sip

> "MJ" == Matthew Jordan writes: MJ> That is incorrect. The sip_sendhtml callback will update the url MJ> stringfield on the SIP pvt. It then transmits a re-INVITE via MJ> transmit_reinvite_with_sdp. There was no re-INVITE, just the initial INVITE. And it did not have an Access-URL header.

Re: [asterisk-dev] dial url with sip

> "DG" == Dennis Guse writes: DG> If you are in control of the SIP-Phone, you could pass additional DG> information via SIPAddHeader in your dialplan. Thanks. That turns out to be exactly what I wanted. I've spent the last few years at a lower level, including when working with voip softwa

Re: [asterisk-dev] dial url with sip

On Mon, Jun 2, 2014 at 4:14 PM, James Cloos wrote: >> "MJ" == Matthew Jordan writes: > > MJ> That is incorrect. The sip_sendhtml callback will update the url > MJ> stringfield on the SIP pvt. It then transmits a re-INVITE via > MJ> transmit_reinvite_with_sdp. > > There was no re-INVITE, just

Re: [asterisk-dev] [Code Review] 3541: res_http_websocket: Create a websocket client

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3541/#review12031 --- Ship it! Other than the minor nit below, LGTM. trunk/tests/t