Re: [asterisk-dev] Mailing List Future

e community ? Have A Happy Friday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.411.7266 Fax 1.888.892.7266 Email p...@scom.ca On

Re: [asterisk-dev] Mailing List Future

yes basically all email would come from @asterisk-dev.groups.io which would be more main stream and at the same time be unique to asterisk dev! Have A Happy Friday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca> 00

Re: [asterisk-dev] Mailing List Future

I think we are getting off track here, email can come from anywhere like mine when i send an email it comes from "Paul Kudla " which is perfectly normal, if i sent an email from : "Paul Kudla " that would be wrong because the email address does not exist and would

Re: [asterisk-dev] Mailing List Future

; Win64; x64; rv:109.0) Gecko/20100101 Firefox/115.0) --MyK3xsRbdj9XTGHGzBSm Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable Time to get in on the action ... would appreciate some form of reply to thi= s to see what happens at mail-system level to

Re: [asterisk-dev] Mailing List Future

again just trying to help when i signed up for the new mailing list see headers below, a few things to note, return address & from address needs to match, this is a common spam filter which is enabled on my email server. You have no idea how many emails come in saying from "P

Re: [asterisk-dev] Mailing List Future

53(8.8.8.8) ;; WHEN: Fri Jan 05 02:08:59 EST 2024 ;; MSG SIZE rcvd: 337 Have A Happy Friday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.

Re: [asterisk-dev] Mailing List Future

ok i will post examples if/when this happens then for better clarificastion unless groups.io is uniqe to asterisk ? being an isp mailing lists / open systems are the first to get hacked ! Have A Happy Thursday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca

Re: [asterisk-dev] Mailing List Future

Thursday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.411.7266 Fax 1.888.892.7266 Email p...@scom.ca On 1/4/2024 6:31 AM, Joshua C

Re: [asterisk-dev] Mailing List Future

are doing you best, just letting you know some difficulties before they become a large scale issue Have A Happy Thursday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Cana

Re: [asterisk-dev] Mailing List Future

less disputes. Dispute listing of 66.175.222.12 Have A Happy Tuesday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.41

Re: [asterisk-dev] Mailing List Future

track and update bad ip's on the fly within 24 hours, so to land on this list means a server has been very very bad. let me know if i can help further. Have A Happy Tuesday !!! Thanks - Paul Kudla (Manager SCOM.CA Internet Services Inc.) Scom.ca Internet Services <http://www.scom.ca&g

Re: [asterisk-dev] Mailing List Future

Hi paul from scom.ca I am an isp and would be prepared to host this mailing list for free on a deficated server In all fairness i need the following Prefrered os ( i use freebsd ) Aprox bandwidth needed ( i host out of peer one in Toronto ontario canada ) Full setup info ie apache

Re: [asterisk-dev] Require Help in Re-Compile of chan_sip.so

yep Happy Wednesday !!! Thanks - paul Paul Kudla Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.411.7266 Fax 1.888.892.7266 Email p...@scom.ca On 10/12/2022 9:50 AM, Thomas Ray wrote:

Re: [asterisk-dev] Require Help in Re-Compile of chan_sip.so

as looking at core show settings Happy Wednesday !!! Thanks - paul Paul Kudla Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.411.7266 Fax 1.888.892.7266 Email p...@scom.ca On 10/12/2022 9

Re: [asterisk-dev] Require Help in Re-Compile of chan_sip.so

auth rejection for" with the ip address? Happy Wednesday !!! Thanks - paul Paul Kudla Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.411.7266 Fax 1.888.892.7266 Email p...@scom.ca On 10

[asterisk-dev] Require Help in Re-Compile of chan_sip.so

bly paypal) please advise -- Happy Wednesday !!! Thanks - paul Paul Kudla Scom.ca Internet Services <http://www.scom.ca> 004-1009 Byron Street South Whitby, Ontario - Canada L1N 4S3 Toronto 416.642.7266 Main 1.866.411.7266 Fax 1.888.892.7266 Em

[asterisk-dev] AppKonference 2.7

I have released an updated AppKonference. You can download the latest code from source forge: sourceforge.net/projects/appkonference -- _ -- Bandwidth and Colocation Provided by htt

Re: [asterisk-dev] [Code Review] 4379: Example configuration scenario - Super Awesome Company: Phase 1 - Patch 1

racter HEX into their SIP phones. I'd prefer we make this some what simple for people to actually connect phones to and demo out. - Paul Belanger On Jan. 27, 2015, 7:15 p.m., rnewton wrote: > > --

Re: [asterisk-dev] git migration update

of the way OpenStack Ci works around it, basically there is a merge working branch (code in review) into master branch job that fires each time. If successful, the tests continue to run, if it fails, then gerrit gives a -1 right away and no tests are executed. Doing the same would be recommended.

Re: [asterisk-dev] git migration update

er should >> work just fine for that purpose. >> >> > > Another +1 here. The beautiful thing about git is that you're not going to > need to do that. Anyone can use whatever git method they want (local, > github, stash, etc) and just rebase against the origin bran

Re: [asterisk-dev] ARI Extending Existing Feature: bridge control

9683M. > Date: Thu, 18 Dec 2014 07:41:59 GMT. > Cache-Control: no-cache, no-store. > Content-type: application/json. > Content-Length: 61. > . > > > T 178.62.127.227:8088 -> 178.62.127.227:44938 [AP] > { > "message": "Application or extension must be

Re: [asterisk-dev] ARI Extending Existing Feature: bridge control

d to be written from scratch. And I think this is the main reason people are slow to move to ARI or fearful from dropping the dialplan. Because doing: exten => s,1,NoOp() same => n,Dial(SIP/f...@example.org) is a lot easier then origination a channel over ARI, creating bridges and playing any

Re: [asterisk-dev] ARI Extending Existing Feature: bridge control

not be controlled by stasis. And since it is not in stasis, ARI cannot modify it. I think the general idea was to build a new app_dial atop of ARI, then your application would provide that functionality to control the L parameter. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybea

Re: [asterisk-dev] ARI Extending Existing Feature: bridge control

g for each > of the iterated bridges, then output it via the JSON response? > 2. Is there a way to manipulate the configuration of the bridge, via > modifying the associated bridge configuration? > > The floor is now open :-) > I am a little confused what you are asking. You want t

Re: [asterisk-dev] [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 29, 2014, at 4:26 PM, Matthew Jordan wrote: > On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht wrote: >> >> On Oct 28, 2014, at 5:03 PM, Ben Langfeld wrote: >> >> On 28 October 2014 19:47, Derek Andrew wrote: >>> >>> What is the alternativ

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

On Oct 29, 2014, at 2:45 PM, Ben Klang wrote: > >> On 10/28/2014 06:03 PM, Ben Langfeld wrote: >>> On 28 October 2014 19:47, Derek Andrew wrote: >>> What is the alternative to the dial plan? Is everyone talking about getting >>> rid of the statements like: >>> exten => s,1, >>> >>> what is t

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

built for us to move completely away from AMI/AGI.” or this "Paul: take away apps, and whatever is in the core is what we should care about.” > > > -- > _ > -- Bandwidth and Colocation Provided by http://

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

share their vision with the rest of the Asterisk community. On Oct 27, 2014, at 2:32 PM, Jeffrey Ollie wrote: > On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht wrote: >> >> The reason the dial plan can never be deprecated is because Asterisk >> wouldn’t be Asterisk without the

Re: [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)

dude. On Oct 24, 2014, at 11:48 AM, Jeffrey Ollie wrote: > On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht wrote: >> >> When Matt says deprecating the dial plan would be difficult and would take a >> long time it seems to me he’s being evasive and misleading. He doesn’t

[asterisk-dev] AppKonference 2.6

I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonference That said Asterisk 13 doesn’t get that much attention because I use Asterisk 1.4 + some hacks. Here’s a link to my Asterisk 1.4 gi

Re: [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 23, 2014, at 1:58 PM, Kevin Larsen wrote: > > From: Paul Albrecht > > > Seems like now is as good a time as any to raise these issues, in > > fact, sooner is better than later because once developers start down > > a path it’s very difficult to get them c

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 23, 2014, at 1:55 AM, Olle E Johansson wrote: > It is critical that a group of developers ask themself questions along > these lines - "what if???" > > - What if we removed AGi and AMI? > - What if we made a pluggable PBX? > - What if we restarted working on a SIP channel? > - What if w

Re: [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 3:27 PM, Kevin Larsen wrote: > > From: Paul Albrecht > > Here’s a link to the minutes: https://wiki.asterisk.org/wiki/ > > display/AST/AstriDevCon+2014 > > > > It has you saying: Leif: we're in a transition, moving from dialplan

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 3:39 PM, Matthew Jordan wrote: > > On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht wrote: > > On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote: > >> >> On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht wrote: >> >> On Oct 22

Re: [asterisk-dev] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)

s to be some more justification for such a profound change to a mature product interface than some vague desire by unknown persons who know best for the entire Asterisk community. > So, to answer your question, yes, and no. > > On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht wrote

Re: [asterisk-dev] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 2:26 PM, Leif Madsen wrote: > > > On 22 October 2014 14:55, Paul Albrecht wrote: > > On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote: >> This is an open source project. Communication is done in an open, >> transparent manner. People shou

Re: [asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang)

On Oct 22, 2014, at 11:47 AM, BJ Weschke wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote: >> >>> Paul Albrecht wrote: >>>> Really? Shouldn’t something this major affecting the entire Asterisk >&

Re: [asterisk-dev] [asterisk-users] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote: > > On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht wrote: > > On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote: > > > Paul Albrecht wrote: > >> Really? Shouldn’t something this major affecting the enti

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Wed, Oct 22, 2014 at 9:14 AM, Paul Albrecht wrote: > The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy > talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the > first place. It’s completely impractical and can never happen. Moreover, Leif

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote: > Paul Albrecht wrote: >> Really? Shouldn’t something this major affecting the entire Asterisk >> community get discussed on the lists? Any idea what Leif is talking >> about when he says the community is in transition, mov

[asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

Really? Shouldn’t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial plan model to external control. -- _

Re: [asterisk-dev] open appliance platform

VoIP > & UC Appliance Developers.” Go to > http://www.patton.com/company/newsrelease.asp?id=2592 > Can I get one for free? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https

Re: [asterisk-dev] Git Migration

On Wed, Sep 24, 2014 at 10:57 AM, Leif Madsen wrote: > I'm not adding much to the conversation, other than to echo both Russell and > Paul that what they've described works very well. At Thinking Phones we > moved to this same model as well from a subversion based system. The

Re: [asterisk-dev] Queue discussion at Astricon

On Fri, Oct 17, 2014 at 9:44 AM, Matthew Jordan wrote: > > > On Fri, Oct 17, 2014 at 4:04 AM, Lenz Emilitri wrote: >> >> Marco and I will be around at Astricon and Astridevcon so would be >> glad to talk about it. >> l. >> >> 2014-10-16 2

[asterisk-dev] Queue discussion at Astricon

cements or even feature requests. Either way, I'll be kicking around Las Vegas is anybody wants to chat. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter

Re: [asterisk-dev] [Code Review] 4035: Dialplan function to get first/head caller channel on queue

) and accept a position argument? - Paul Belanger On Sept. 30, 2014, 8:04 a.m., Kristian Høgh wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.as

Re: [asterisk-dev] Opinions Needed: Case sensitivity in config file section names

On Tue, Sep 23, 2014 at 1:47 PM, George Joseph wrote: > On Tue, Sep 23, 2014 at 11:13 AM, Matthew Jordan wrote: >> >> >> >> On Tue, Sep 23, 2014 at 11:29 AM, Paul Belanger >> wrote: >>> >>> On Tue, Sep 23, 2014 at 11:45 AM, George Joseph >

Re: [asterisk-dev] beta2 compile failure

On Sep 23, 2014, at 12:22 PM, Matthew Jordan wrote: > > > On Tue, Sep 23, 2014 at 11:19 AM, Paul Albrecht wrote: > > On Sep 23, 2014, at 10:24 AM, Matthew Jordan wrote: > >> >> >> On Tue, Sep 23, 2014 at 10:11 AM, Paul Albrecht wrote: >> &

Re: [asterisk-dev] Opinions Needed: Case sensitivity in config file section names

e > match case sensitive might cause unexpected problems. > > Thoughts? > For me, case sensitive. Because I config files that do have: [Foo] [foo] [fOO] don't ask, long story. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode

Re: [asterisk-dev] beta2 compile failure

On Sep 23, 2014, at 10:24 AM, Matthew Jordan wrote: > > > On Tue, Sep 23, 2014 at 10:11 AM, Paul Albrecht wrote: > > On Sep 23, 2014, at 9:25 AM, Joshua Colp wrote: > > > Paul Albrecht wrote: > >> > >> On Sep 22, 2014, at 3:47 PM, Joshua

Re: [asterisk-dev] beta2 compile failure

On Sep 23, 2014, at 9:25 AM, Joshua Colp wrote: > Paul Albrecht wrote: >> >> On Sep 22, 2014, at 3:47 PM, Joshua Colp > <mailto:jc...@digium.com>> wrote: >> >>> Paul Albrecht wrote: >>>> >>>> Asterisk 13 beta2 compile fails:

Re: [asterisk-dev] beta2 compile failure

On Sep 22, 2014, at 3:47 PM, Joshua Colp wrote: > Paul Albrecht wrote: >> >> Asterisk 13 beta2 compile fails: >> >> . >> . >> . >> [CC] chan_pjsip.c -> chan_pjsip.o >> [CC] pjsip/dialplan_functions.c -> pjsip/dialplan_fu

[asterisk-dev] beta2 compile failure

Asterisk 13 beta2 compile fails: . . . [CC] chan_pjsip.c -> chan_pjsip.o [CC] pjsip/dialplan_functions.c -> pjsip/dialplan_functions.o [LD] chan_pjsip.o pjsip/dialplan_functions.o -> chan_pjsip.so /usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../x86_64-pc-linux-gnu/bin/ld: /usr/lib/gcc/x86_64-

Re: [asterisk-dev] Git Migration

ter > no matter what specific infrastructure you go with. > > -- > Russell Bryant > > -- > _ > -- Bandwidth and Colocation Provided by http://w

Re: [asterisk-dev] Git Migration

ile complicated and full of moving parts, its extremely good at what it does. I have a system running both public and private for different projects I am doing. One thing that is great about it, developers can develop faster without knowing or understanding the release processes of a project. I

Re: [asterisk-dev] [Code Review] 3981: chan_rtp: Add unicast RTP support to chan_multicast_rtp.

On Sep 7, 2014 2:28 PM, "Joshua Colp" wrote: > > Johann Steinwendtner wrote: >> >> On 2014-09-07 17:07, Joshua Colp wrote: >>> >>> This is an automatically generated e-mail. To reply, visit: >>> https://reviewboard.asterisk.org/r/3981/ >>> >>> >> >>> Testing >>> >>> Originated a call to a UnicastR

Re: [asterisk-dev] AstriCon Hackathon

On Thu, Sep 4, 2014 at 4:14 PM, Paul Belanger wrote: > On Thu, Sep 4, 2014 at 3:54 PM, Matthew Jordan wrote: >> Join a worldwide community of designers, developers, and communications >> technologists to to create, code, and design apps built on Asterisk and >> other commu

Re: [asterisk-dev] AstriCon Hackathon

gt; or at http://www.asterisk.org/community/astricon-user-conference/hackathon > > See everyone in Las Vegas! > > Matt > Is there any prices associated with hackathon? Both sites seem to be missing this information. I'd consider them a good motivator for people to join. -- Paul Be

Re: [asterisk-dev] [Code Review] 3952: Add 'rtpbindaddr' setting for chan_sip

rtp (media) on eth1. Diffs - trunk/configs/samples/sip.conf.sample 422198 trunk/channels/chan_sip.c 422198 trunk/CHANGES 422198 Diff: https://reviewboard.asterisk.org/r/3952/diff/ Testing --- kamailio proxy with rtpengine. Multihomed asterisk system. Thanks, Paul

[asterisk-dev] [Code Review] 3952: Add 'rtpbindaddr' setting for chan_sip

Diff: https://reviewboard.asterisk.org/r/3952/diff/ Testing --- kamailio proxy with rtpengine. Multihomed asterisk system. Thanks, Paul Belanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-dev] [Code Review] 3925: PJSIP: Use IP address in favor of hostname in SDP origin line.

; Mark Michelson wrote: > In all seriousness though, a better idea would probably be to set the SDP > origin information per session based on the transport in use, similar to how > the connection line is set. > > Paul Belanger wrote: > +1 I'd actually like the ability

Re: [asterisk-dev] [Code Review] 3925: PJSIP: Use IP address in favor of hostname in SDP origin line.

; Mark Michelson wrote: > In all seriousness though, a better idea would probably be to set the SDP > origin information per session based on the transport in use, similar to how > the connection line is set. +1 I'd actually like the ability to assign a specific IP address

Re: [asterisk-dev] chan_sip marked as Extended support in Asterisk 13+

Joking aside, I don't see an issue with this. Even though I have done zero testing with chan_pjsip at the moment, I am happy to see it being promoted. Obviously there is going to be some bumps in the road however I'd rather see development efforts focused on chan_pjsip then split with ch

Re: [asterisk-dev] [Code Review] 3823: Stasis: Allow configuration of message types to disallow

find anything on Jira / wiki that explained the need. I assume it is because of performance issues? - Paul Belanger On July 17, 2014, 12:49 p.m., opticron wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 3807: xmldoc: Add support for an tag in the Asterisk XML documentation

tps://reviewboard.asterisk.org/r/3807/#comment22975> Your priority values for each extension is not consistent. 5 are exten => 1,1 and 1 is exten => 1,n. - Paul Belanger On July 16, 2014, 4:58 p.m., Matt Jordan wrote: > > --- > Thi

Re: [asterisk-dev] Asterisk PJSIP stack PUBLISH support

wiki/display/AST/Resource+List+Subscriptions > > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://as

Re: [asterisk-dev] [Code Review] 3572: cli_aliases: Add 'pjsip reload' alias because it's nice. Plus, another for 11 syntax compatability.

anges in the template seem unnecessary and potentially break systems that depend on the default config files. - Paul Belanger On May 28, 2014, 8:14 p.m., rnewton wrote: > > --- > This is an automatically generated e-mail. T

Re: [asterisk-dev] [Code Review] 3549: Replace __ast_answer with ast_raw_answer in app_control_answer

/res_stasis_answer.c 414194 Diff: https://reviewboard.asterisk.org/r/3549/diff/ Testing --- Local ARI application (payload-voice[1]) with SIPp (sipp -sn uac 127.0.0.1) to confirm HTTP 500 is no longer returned on answer. [1] https://github.com/kickstandproject/payload-voice Thanks, Paul Belanger

[asterisk-dev] [Code Review] 3549: Replace __ast_answer with ast_raw_answer in app_control_answer

-sn uac 127.0.0.1) to confirm HTTP 500 is no longer returned on answer. [1] https://github.com/kickstandproject/payload-voice Thanks, Paul Belanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-dev] [svn-commits] rmudgett: branch 1.8 r411715 - in /branches/1.8: ./ channels/ configs/ includ...

ssive change mid-release. Removing a command-line option is certainly going to break some peoples boxes. Why not a deprecated warning and then removal from trunk to give people time to react? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Free

Re: [asterisk-dev] Asterisk and external SIP subscribtions

nf.sample > > ; Configuration file for SIP presence integration > ; > ; This is where you configure > ; - presence servers to use > ; - which extensions to publish > ; - which devices to publish > > [pinana-one] > type=presenceserver > ; Domain for presence stat

Re: [asterisk-dev] Asterisk and external SIP subscribtions

On Wed, Mar 19, 2014 at 9:56 AM, Olle E. Johansson wrote: > > On 19 Mar 2014, at 14:50, Paul Belanger wrote: > >> Greetings, >> >> I wanted to ask if there is any sort of design document / work started >> on having Asterisk 13 be able to subscribe to external S

[asterisk-dev] Asterisk and external SIP subscribtions

related is outside asterisk. EG: [example.org] server = 192.168.1.1 users = * So, thoughts? Interested in helping? Want to also fund this venture? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twit

Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11236 --- Ship it! Ship It! - Paul Belanger On March 15, 2014, 10:02

Re: [asterisk-dev] [Code Review] 3365: res_pjsip_session: decline SDP offers with no audio streams

488ing? According to http://tools.ietf.org/html/rfc6337#section-2.3 it's preferred, but I'm unsure if pjproject exposes the ability. - Paul Belanger On March 16, 2014, 8:36 p.m., Matt Jordan wrote: > > --- > This is an auto

Re: [asterisk-dev] [Code Review] 3362: func_beep: New function for periodic beeps.

implement something like a tick option on a channel. I _know_ you can do it, but not currently within Asterisk. For me, I can see using it for a conference user before I'm about to kick them. But ya, func_tick is where it is at! - Paul --

Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE

org/r/3349/#comment20860> space between t and + since you are here. /trunk/channels/sip/reqresp_parser.c <https://reviewboard.asterisk.org/r/3349/#comment20861> might be worth adding a unit test for this code path. Something like tel:911 - Paul Belanger On March 15, 201

Re: [asterisk-dev] [Code Review] 3343: res_pjsip: Enable DNS support.

rt maintaining it, because of difficulties getting code merged. I cannot speak to pjsip or the DNS patch, since I have not used them yet. But, if pjproject supports it, why not use it? If we can expose DNS for other channel drivers, great, but that should be a discussion point. Either way,

Re: [asterisk-dev] [Code Review] 3343: res_pjsip: Enable DNS support.

, we'd ideally work with upstream to get them merged up. I'm pretty happy we finally got all the patches to pjproject merged by teluu, but some what concerned about the troubles other people have had getting code merged (I think ajprojects). I'm not sure if Digium had to get some su

Re: [asterisk-dev] magic number 128- for concurrent meetme monitoring calls.

On Fri, Mar 14, 2014 at 10:02 AM, Shaun Ruffell wrote: > On Fri, Mar 14, 2014 at 02:40:22PM +0100, Olle E. Johansson wrote: >> >> On 14 Mar 2014, at 14:13, Paul Belanger wrote: >> >> > On Fri, Mar 14, 2014 at 3:02 AM, Olle E. Johansson wrote: >> >> >

Re: [asterisk-dev] [Code Review] 3343: res_pjsip: Enable DNS support.

On Fri, Mar 14, 2014 at 9:51 AM, Sean Bright wrote: > On 3/14/2014 2:41 AM, Olle E. Johansson wrote: > > > On 13 Mar 2014, at 22:13, Sean Bright wrote: > > On 3/13/2014 4:42 PM, Paul Belanger wrote: > > +1 with Dan. Comments aside on DNS functionality (I have opinions bu

Re: [asterisk-dev] magic number 128- for concurrent meetme monitoring calls.

ile handles that > will > expire at some point and give you strange problems. We had a discussion > about that > a year or two ago on the list. > Interesting, I missed that discussion, can you sum it up to a few lines while I look for the thread? -- Paul Belanger | PolyBeacon

Re: [asterisk-dev] magic number 128- for concurrent meetme monitoring calls.

On Fri, Mar 14, 2014 at 7:32 AM, Tony Mountifield wrote: > In article > , > Paul Belanger wrote: >> > >> > Sounds like the ulimit is at the default 1024. You need to increase it >> > because >> > Asterisk needs a lot of file descriptors. >>

Re: [asterisk-dev] magic number 128- for concurrent meetme monitoring calls.

e default 1024. You need to increase it > because > Asterisk needs a lot of file descriptors. > > This kind of question is better asked on the asterisk-users list. > Yup, next limit you'll hit is dahdi pseudo channels, which is 512. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.b

Re: [asterisk-dev] [Code Review] 3343: res_pjsip: Enable DNS support.

way that > all channels would > > benefit. It is kind of amusing to see that turned around to not apply to > the new kid on the block. > +1 with Dan. Comments aside on DNS functionality (I have opinions but sitting this one out). Any functionality should be channel agnostic. I

Re: [asterisk-dev] PJSIP: allow/disallow or codecs?

On Thu, Mar 6, 2014 at 5:32 PM, Damien Wedhorn wrote: > On 07/03/14 08:21, Matthew Jordan wrote: > > On Thu, Mar 6, 2014 at 3:42 PM, Damien Wedhorn wrote: >> >> On 07/03/14 07:29, Matthew Jordan wrote: >> >> On Thu, Mar 6, 2014 at 3:22 PM, Paul Belanger >>

Re: [asterisk-dev] PJSIP: allow/disallow or codecs?

low in 12, discontinue > allow/disallow in 13. > >> >> Note that even if codecs is chosen, allow and disallow continue to work so >> no existing pjsip.conf is broken. >> > For me to be on-board with the change, we'd have to apply it to all channel drives

Re: [asterisk-dev] [Code Review] 3191: object uniqueid phases 1-3: ami/ari origination & ami bridge, playback, and snoop

dn't this be false? /branches/12/rest-api/api-docs/channels.json <https://reviewboard.asterisk.org/r/3191/#comment20680> tabbing seems out of place - Paul Belanger On March 3, 2014, 11:51 p.m., Scott Griepentrog wrote: > > -

Re: [asterisk-dev] [Code Review] 3242: starpy: add UniqueId parameter to ami.Originate()

ou'll need to do some version checks on the AMI for the API breakage. I'm not sure what will happen when you send asterisk a parameter it doesn't know about. But we still want starpy to support 1.8, 11 as long as possible. - Paul Belanger On Feb. 25, 2014, 9:42 p.m., Sco

Re: [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section

? I'll be trying to use the new config framework and see if I can get it going. - Paul Belanger On Feb. 28, 2014, 4:34 p.m., Paul Belanger wrote: > > --- > This is an automatically generated e-mail. To reply,

Re: [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id

> On March 1, 2014, 11:41 p.m., Paul Belanger wrote: > > /branches/12/rest-api/api-docs/bridges.json, lines 50-56 > > <https://reviewboard.asterisk.org/r/3278/diff/1/?file=54950#file54950line50> > > > > After reading Matts comments, this still doesn'

Re: [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id

name space already, we should be just creating the bridge with the ID passed. POST /bridges should be use when you want asterisk to generate the UUID for channel, and POST /bridges/{bridgeId} when you want to override it. - Paul Belanger On Feb. 28, 2014, 5:39 p.m., Scott Griepe

Re: [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id

> On Feb. 28, 2014, 9:09 p.m., Paul Belanger wrote: > > Woah, nice. What about the ability to do PUT /bridges/a1b2c3, creating the > > bridge if it doesn't exists. Actually, the more I think about this. The more I can see having PUT and dropping POST, like we do in device

Re: [asterisk-dev] [Code Review] 3278: ari: create bridges with specified unique id

, creating the bridge if it doesn't exists. - Paul Belanger On Feb. 28, 2014, 5:39 p.m., Scott Griepentrog wrote: > > --- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.as

Re: [asterisk-dev] CentOS packaging

fork die and moving directly to the 2.2 release (Please say yes, please say yes). -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- __

Re: [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section

/r/3279/diff/ Testing --- local development. Setup [general] queue_log = no queue_log = yes Queue logfiles were created. Thanks, Paul Belanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section

Diff: https://reviewboard.asterisk.org/r/3279/diff/ Testing --- local development. Setup [general] queue_log = no queue_log = yes Queue logfiles were created. Thanks, Paul Belanger -- _ -- Bandwidth and Colocation

Re: [asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section

nice, but this is at least a step in > a better direction. @Matt: Reason for not using the new configuration options was because we are back porting this patch into 1.8, which we've tested with. If committed, I can try my hand at using the new config_options framework and try upgradi

[asterisk-dev] [Code Review] 3279: Iterate through logger.conf [general] section

] queue_log = no queue_log = yes Queue logfiles were created. Thanks, Paul Belanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-dev] [Code Review] 2904: More consistent ARI error messages

. Diffs - trunk/res/res_ari.c 400880 Diff: https://reviewboard.asterisk.org/r/2904/diff/ Testing --- Local development box, with python-ari! Thanks, Paul Belanger -- _ -- Bandwidth and Colocation Provided by

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