Re: [asterisk-dev] ISDN Userfield UUI has a length constraint?

On Mon, Jun 16, 2014 at 3:03 PM, Yves A. wrote: > Hello List, > > I made the changes in the q931.c sourcefile and got rid of the 35 > character constraint. > thanks for your attention. > yves > > Am 16.06.2014 15:23, schrieb Yves A.: > > seems, I found the right place by myself... no wonder I co

Re: [asterisk-dev] Looking for definitive answer for Busy() vs. Hangup() when used on a non-answered PRI channel

On Thu, Jul 10, 2014 at 10:57 AM, Justin Killen < jkil...@allamericanasphalt.com> wrote: > Hi, > > > > I’m using freepbx, and I’ve gotten myself into a bit of an argument ( > http://issues.freepbx.org/browse/FREEPBX-7706) about when to use Busy() > vs. Hangup(17). The conversation boils down to

Re: [asterisk-dev] Looking for definitive answer for Busy() vs. Hangup() when used on a non-answered PRI channel

On Thu, Jul 10, 2014 at 12:36 PM, Justin Killen < jkil...@allamericanasphalt.com> wrote: > Okay, that makes sense. So then, how does the answer()’d status affect > that? From my (admittedly limited) experience, it seems that playing > in-band tones only works if the call has been answered? > >

Re: [asterisk-dev] Looking for definitive answer for Busy() vs. Hangup() when used on a non-answered PRI channel

On Thu, Jul 10, 2014 at 2:33 PM, Justin Killen < jkil...@allamericanasphalt.com> wrote: >On Thu, Jul 10, 2014 at 12:36 PM, Justin Killen < > jkil...@allamericanasphalt.com> wrote: > > Okay, that makes sense. So then, how does the answer()’d status affect > that? From my (admittedly limited)

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

On Tue, Sep 23, 2014 at 10:51 AM, George Joseph wrote: > On Tue, Sep 23, 2014 at 9:45 AM, George Joseph < > george.jos...@fairview5.com> wrote: > >> I've been working on some changes for config.c and in the process I've >> found 5 instances of someone attempting to do "cat->name == category_name"

Re: [asterisk-dev] Asterisk 12 - Security Fix Only! (aka: update repotools)

On Wed, Dec 10, 2014 at 1:25 PM, Matthew Jordan wrote: > tl;dr: Update repotools, run 'make'/'make install' > > Since the final bug fix release of Asterisk 12 has now been made as a > release candidate, we need to start merging bug fixes from the > Asterisk 11 branch directly into Asterisk 13. As

Re: [asterisk-dev] DAHDI / indications: US tone as stutter on India, Mexico, and the Philippines

On Sun, Jan 26, 2014 at 7:44 AM, Tzafrir Cohen wrote: > Hi, > > A while ago we noticed while testing that if you set the tonezone of a > DAHDI phone to Mexican ("mx"), a phone with a message waiting eventually > gives a US dialtone after the stutter tone is over. This is because in > zonedata.c: >

Re: [asterisk-dev] DAHDI / indications: US tone as stutter on India, Mexico, and the Philippines

On Mon, Jan 27, 2014 at 4:52 AM, Tzafrir Cohen wrote: > On Sun, Jan 26, 2014 at 04:21:35PM -0600, Richard Mudgett wrote: > > On Sun, Jan 26, 2014 at 7:44 AM, Tzafrir Cohen >wrote: > > > > > Hi, > > > > > > A while ago we noticed while testing that i

Re: [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame

On Thu, Feb 6, 2014 at 10:23 AM, Dave WOOLLEY wrote: > We have a situation where about 1% of incoming 302 responses result in the > call failing as busy, rather than redirecting. We are using a heavily > patched 1.6.1.0, however we have a theory for the mechanism that still > seems to be valid on

Re: [asterisk-dev] 302 redirects ocassionally ignored; hypothesis: later queued busy preferred to earlier early media frame

On Thu, Feb 6, 2014 at 11:43 AM, Dave WOOLLEY wrote: > One point of detail. It is actually the write to the alertpipe, within > ast_queue_control that does the wakeup. > Yes, but there needs to be at least one write to the alertpipe for every frame in the queue. Richard --

Re: [asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements

On Thu, Feb 6, 2014 at 6:40 PM, Jonathan White wrote: > Good afternoon. Thanks for adding this feature. I have been testing it > today and notice some unexpected behaviour. > > When multiple users call in and set different languages they will only hear > the language set by the first caller t

Re: [asterisk-dev] [Code Review] 3192: chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel

On Fri, Feb 7, 2014 at 7:20 AM, Tzafrir Cohen wrote: > On Thu, Feb 06, 2014 at 07:04:39PM -, rmudgett wrote: > > > > --- > > This is an automatically generated e-mail. To reply, visit: > > https://reviewboard.asterisk.org/r/3192/#review10

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

On Thu, Mar 13, 2014 at 5:07 PM, beiyan jin wrote: > In my load test calls, > 1. each call has two parties connected by meetme conference. > 2. Each call is recorded by monitor. > > For every load test, before the number of concurrent calls reach 128, > everything is fine. But after 128, newly st

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

On Sat, Apr 5, 2014 at 1:58 AM, Olle E. Johansson wrote: > > On 04 Apr 2014, at 20:32, SVN commits to the Digium repositories < > svn-comm...@lists.digium.com> wrote: > > > - case 'I': > > - ast_set_flag(&ast_options, > AST_OPT_FLAG_INTERNAL_TIMING); > > -

Re: [asterisk-dev] ISDN UDI Call with Dial command

On Thu, May 8, 2014 at 5:29 PM, Pawel Pastuszak wrote: > Hello, > > > I have notice that if i uses Dial(DAHDI/g1d/1234) the d option doesn't get to > chan_dahi but if i uses originate with d option i see the UDI call but not > with dial can some help why is this? > > > Example of the originate >

Re: [asterisk-dev] Asterisk CALLINGTON for SS7

On Thu, May 15, 2014 at 9:45 AM, Alberto Rinaudo wrote: > Good morning, > I have a problem with that variable: > I use asterisk 11 + dahdi 2.7.0.1 + libss7 1.0.2 + libpri 1.4 > in another box (with pri cards) I use CALLINGTON. But I recently found out > that this variable is not populated for SS7.

Re: [asterisk-dev] Asterisk CALLINGTON for SS7

On Thu, May 15, 2014 at 11:56 AM, Alberto Rinaudo wrote: > If I'm right CALLINGTON is populated from a variable > "caller.id.number.plan" that is equals to "p->cid_ton" > > I was comparing sig_pri.c and sig_ss7.c using asterisk 11.9 > > > in sig_pri.c >279 caller.id.number.plan = p->

Re: [asterisk-dev] app_confbridge + USER_OPT_TALKER_DETECT

On Wed, May 21, 2014 at 9:19 PM, Jared Mauch wrote: > > Hello, > > I'm trying to finish porting over my app_meetme systems to interact > with app_confbridge in 12.2.0 and I can't see to get the talker detection > data either via CLI or AMI. I've not tried with ARI yet, but don't

[asterisk-dev] Proposed change to how accountcode is propagated to other channels.

The current behavior is to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2

Re: [asterisk-dev] Astobj2 debugging change proposal

On Mon, Jun 9, 2014 at 12:31 PM, George Joseph wrote: > On Mon, Jun 9, 2014 at 11:22 AM, Corey Farrell wrote: > >> One thing I dislike about AST_DEVMODE is that changing it requires >> rerunning ./configure. I feel ./configure should be restricted as much as >> possible to checking dependencies

Re: [asterisk-dev] running pjsip testsuite

Another thing that is important is that the sample configs must be installed. Many tests have some difficulty if this is not the case. For me it was because I had configurations defining the same endpoints with chan_sip and chan_pjsip. The conflicting configs caused crashes in tests that did not u

Re: [asterisk-dev] IAX different reentrant locks depth in the same function

On Tue, Apr 21, 2015 at 2:42 PM, Yousf Ateya wrote: > While testing the patch of > https://issues.asterisk.org/jira/browse/ASTERISK-24983, I found that in > some paths (ex. in call hangup procedure) we acquire the lock multiple > times from the same thread. > > The lock implementation supports re

Re: [asterisk-dev] Contacts, Contact Status and Sorcery

On Thu, Apr 30, 2015 at 2:10 PM, George Joseph wrote: > > Change https://gerrit.asterisk.org/261 sparked some discussion about > contacts and contact status so I'd like to continue that here. > > Today, dynamic contacts (incoming registrations) are created on the fly as > full sorcery objects and

Re: [asterisk-dev] (unreported) uninitialized: struct ast_sockaddr

On Wed, May 13, 2015 at 3:59 AM, Alexander Traud wrote: > > What you're proposing is a change to the semantics of ast_sockaddr. > > Not sure what you mean by semantics. Please, let us ignore ast_sockaddr for > a second and see : > Semantics

Re: [asterisk-dev] pjsip vs cel

On Wed, Jun 10, 2015 at 6:22 PM, James Cloos 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. > > T

Re: [asterisk-dev] Proposing change to Queue and missed calls behavior

On Wed, Dec 9, 2015 at 2:06 AM, Stian Hvatum wrote: > On 12/07/2015 05:02 PM, Olle E. Johansson wrote: > >> On 07 Dec 2015, at 16:53, Mark Michelson wrote: >>> >>> On 12/04/2015 01:00 PM, Stian Hvatum wrote: >>> Hi, I have a problem with an accompanying solution that I wish to share, >

Re: [asterisk-dev] Bridges, T.38, and other good times

On Sun, Dec 6, 2015 at 7:57 PM, Matthew Jordan wrote: > Hello all - > > One of the efforts that a number of developers in the community here at > Digium have been at work at are cleaning up test failures exposed by > Jenkins [1]. One of these, in particular, has been rather difficult to > resolve

Re: [asterisk-dev] How to use a DAHDI kernel driver in linux using The Bridging Framework Tecnology in the Asterisk 13

On Wed, Jan 6, 2016 at 6:27 AM, Diógenes Vila Nova Pereira < d...@cesar.org.br> wrote: > Hi Folks, > > I'm newbie in Asterisk developement tecnology. I had read and seen > documentation that the Asterisk supports new bridging framework tecnology > that has a pluggable interface, allowing a native

Re: [asterisk-dev] Asterisk 13.7.0 app PAGE plays participant count

On Wed, Jan 27, 2016 at 6:35 PM, Ross Beer wrote: > > Hi, > > I'm having an issue with the PAGE feature in Asterisk 13.7.0, when > connecting a PAGE it says 'there is only one other participant in the > conference'. > > This shouldn't be happening, I know that PAGE used ConfBridge however it > s

[asterisk-dev] q931.c: Fix DISCONNECT Progress Indicator ie handling. (libpri[1.4])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2426 Change subject: q931.c: Fix DISCONNECT Progress Indicator ie handling. .. q931.c: Fix DISCONNECT Progress Indicator ie handling. There

[asterisk-dev] Add .gitignore (libss7[1.0])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2422 Change subject: Add .gitignore .. Add .gitignore Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6 --- A .gitignore 1 file changed, 11

[asterisk-dev] Add .gitignore (libss7[2.0])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2420 Change subject: Add .gitignore .. Add .gitignore Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6 --- A .gitignore 1 file changed, 11

[asterisk-dev] Add .gitignore (libpri[master])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2424 Change subject: Add .gitignore .. Add .gitignore Change-Id: I11ac3b47a9d5d0a0c1ea4559280b75ef5d866d62 --- A .gitignore 1 file changed, 12

[asterisk-dev] q931.c: Substitute PROGRESS for DISCONNECT with progress ind... (libpri[1.4])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2425 Change subject: q931.c: Substitute PROGRESS for DISCONNECT with progress indicator #8 .. q931.c: Substitute PROGRESS for DISCONNECT with

[asterisk-dev] Add .gitignore (libpri[1.4])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2423 Change subject: Add .gitignore .. Add .gitignore Change-Id: I11ac3b47a9d5d0a0c1ea4559280b75ef5d866d62 --- A .gitignore 1 file changed, 12

[asterisk-dev] Add .gitignore (libss7[master])

Richard Mudgett has uploaded a new change for review. https://gerrit.asterisk.org/2421 Change subject: Add .gitignore .. Add .gitignore Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6 --- A .gitignore 1 file changed, 11

Re: [asterisk-dev] Config reading and scanf with large numbers

On Wed, Jun 1, 2016 at 5:25 AM, snuffy wrote: > Hello All, > > I noticed a bug report ASTERISK-25972, > The referenced issue has nothing to do with what you are talking about. > > Looking through the code we do the following: > > sscanf(string,"%30d",&my_int); > > Now issue is an integer can't

Re: [asterisk-dev] Asterisk Load Performance

On Fri, Jun 17, 2016 at 12:36 PM, Michael Petruzzello < michael.petruzze...@civi.com> wrote: > Hello, > > I am currently working on determining bottlenecks in Asterisk and a Stasis > App. I'm currently trying to handle 83.3 calls/second. For the most part, > Asterisk and the Stasis APP handle that

Re: [asterisk-dev] Asterisk Load Performance

On Tue, Jun 21, 2016 at 11:12 AM, Michael Petruzzello < michael.petruzze...@civi.com> wrote: > >On Fri, Jun 17, 2016 at 1:37 PM, Richard Mudgett > wrote: > >> > >> > >> On Fri, Jun 17, 2016 at 12:36 PM, Michael Petruzzello > >> wrote: >

Re: [asterisk-dev] ASTERISK-26145 - Task Process Issues possibly caused by HEP

On Tue, Jun 28, 2016 at 10:20 AM, Ross Beer wrote: > Hi, > > > I agree that the conversation about HEP default settings doesn't warrant a > lengthy discussion, however, the fact that the 'task processor' causes > asterisk to stop processing packets is a serious issue. This is happening > multiple

Re: [asterisk-dev] Asterisk Load Performance

On Wed, Jun 29, 2016 at 9:55 AM, Michael Petruzzello < michael.petruzze...@civi.com> wrote: > It is very interesting how threading issues on both a stasis application > and Asterisk escalate each other. Using 15 websockets in one stasis > application and removing all thread locking from the applic

Re: [asterisk-dev] Asterisk Load Performance

On Tue, Jul 5, 2016 at 4:03 PM, Jonathan Rose < jonathan.r...@motorolasolutions.com> wrote: > > On Tue, Jul 5, 2016 at 3:43 PM, Michael Petruzzello < > michael.petruzze...@civi.com> wrote: > >> On Wed, Jun 29 at 11:14:04 AM, Richard Mudgett> <https://url

Re: [asterisk-dev] Development of asterisk 1.4.23 Can we please get some development?

On Fri, Jul 15, 2016 at 5:50 PM, Loren Tedford wrote: > Some how you all wound up in my spam folder gotta love it i have been very > busy had a big mistake on my end never chown -Rf apache:apache /var/ lol.. > oops i guess we all make a mistake every now and then.. > > Anyway back to topic I have

Re: [asterisk-dev] Channel Issue with 13.10 and app_queue

On Thu, Aug 4, 2016 at 11:12 AM, Ryan Rittgarn wrote: > Hi Everyone, long time reader, first time poster in this particular > thread. I'm one of the holdouts still using chan_sip for day to day uses as > I have too many applications and implementations that rely on it. That > being said I think I

Re: [asterisk-dev] BRIDGEPEER on multi-party conferences: Thoughts?

On Tue, Aug 9, 2016 at 6:01 PM, Mark Michelson wrote: > Hi folks, > > I've been looking into a Digium customer issue where ConfBridge audio has > been dropping out. The main issue there had to do with DNS, and there is > currently a review [1] up to fix that. > > A secondary issue, though, is tha

Re: [asterisk-dev] Questions about Message/ast_msg_queue

On Mon, Sep 19, 2016 at 4:40 AM, Jean Aunis wrote: > Hello, > > I am currently using a lot of SIP MESSAGEs, which rely on the > Message/ast_msg_queue channel in Asterisk. Unless I misunderstood > something, this channel is acting as a kind of "singleton" : there is only > one instance of this cha

Re: [asterisk-dev] app_queue.c PAUSALL/UNPAUSEALL reason

On Fri, Oct 21, 2016 at 5:46 PM, Troy Bowman wrote: > > I always need to apply this patch before putting Asterisk into Production. > > The pauses for specific queues get the reason logged properly, but the > pausealls don't. I rely on pausalls and their reasons for reporting, > though I'm trying

Re: [asterisk-dev] Queue Gate Concept

On Fri, Oct 21, 2016 at 6:09 PM, Troy Bowman wrote: > Where I work, we're running a call center on Asterisk, using app_queue. > > For a long while, we were using multiple queues for each speciality (or > "skill"), placing agents in several queues, and using the "announce" > queue.conf configurati

Re: [asterisk-dev] AGI:async

On Fri, Nov 25, 2016 at 10:14 AM, Salvatore Franco wrote: > Hi, > It means that the call still waiting something but never enter in Queue > (as if async is not really async). > So I think the dialplan never calls the next step. > You seem to misunderstand what AGI actually is. AGI _is_ dialplan

Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project

On Wed, Nov 30, 2016 at 8:04 AM, Joshua Colp wrote: > On Fri, Nov 25, 2016, at 10:20 AM, Dennis Guse wrote: > > Hey guys, > > > > we continued working on our largest changeset: https://gerrit. > > asterisk.org/#/c/3524/ > > Besides the copyright of HRTFs (still under investigation from our side),

Re: [asterisk-dev] Cannot set CallerId on outgoing call

On Fri, Dec 9, 2016 at 8:54 AM, wrote: > Hi All, > > I have a scenario where an incoming external call comes into Asterisk and > into my Stasis application, I there check my database to find the > destination so I can route the call there. I follow the recommended > procedure: > 1. Create bridge

Re: [asterisk-dev] Subscription behavior when an incoming registration goes away?

On Thu, Dec 22, 2016 at 10:14 AM, Matthew Jordan wrote: > > > On Thu, Dec 22, 2016 at 9:32 AM, George Joseph wrote: > >> When an incoming registration goes away, either because it expires or it >> was explicitly expired, what should we do with subscriptions the contact >> may have? Right now we

Re: [asterisk-dev] Local channel variable propagation: expired hack or a bug?

On Tue, Jan 3, 2017 at 9:38 PM, Kirill Katsnelson wrote: > With Asterisk 1.8 we were relying on the behavior of Originate with Local > channels as mentioned in https://issues.asterisk.org/ji > ra/browse/ASTERISK-17239. This no longer works in Asterisk 13. > > Specifically, when a call is originat

Re: [asterisk-dev] Local channel variable propagation: expired hack or a bug?

On Wed, Jan 4, 2017 at 1:45 PM, Kirill Katsnelson wrote: > On 170103 2114, Richard Mudgett wrote: > >> On Tue, Jan 3, 2017 at 9:38 PM, Kirill Katsnelson >> wrote: >> >> With Asterisk 1.8 we were relying on the behavior of Originate with Local >

Re: [asterisk-dev] Regression: dash a not a character any more?

On Tue, Jan 17, 2017 at 2:03 PM, Kirill Katsnelson wrote: > A change from 1.8 to 13 broke us in some other aspect, and I am trying to > figure out whether this is a bug or things work as expected. In short, the > '-' character is converted to nothing in extension "numbers" > > Here's a snippet th

Re: [asterisk-dev] Regression: dash a not a character any more?

On Tue, Jan 17, 2017 at 3:40 PM, Kirill Katsnelson wrote: > On 170117 1232, Richard Mudgett wrote: > >> On Tue, Jan 17, 2017 at 2:03 PM, Kirill Katsnelson >> wrote: >> >> A change from 1.8 to 13 broke us in some other aspect, and I am trying to >>> fig

Re: [asterisk-dev] Line length restrictions in code changes

On Thu, Mar 16, 2017 at 2:54 PM, Matthew Jordan wrote: > Warning: bike shedding. > > On several code reviews, comments have been left along with a -1 on > the review - indicating that it is not acceptable for merging - due to > line length restrictions. While this is technically correct per our >

Re: [asterisk-dev] AST_MODULE_LOAD_FAILURE vs AST_MODULE_LOAD_DECLINE

On Wed, Apr 5, 2017 at 3:43 PM, George Joseph wrote: > Over the years I've been frustrated at times where Asterisk fails to start > for absolutely no (apparent) reason. No error message, no trap, nothing. > It just ends. Every case of this I've tracked down has been the result of > a module ret

Re: [asterisk-dev] astDB "restart"

On Mon, Apr 10, 2017 at 9:18 AM, Gabriel Ortiz Lour wrote: > Hi, > We are experiencing problems with leaks on our system and we think is > astDB related. > I'm still using 1.8, migrating to 13, but for now i'd like to do > something like an astDB "restart". Close the database and open again..

Re: [asterisk-dev] Asterisk Memory Debugger (MALLOC_DEBUG) and DONT_OPTIMIZE

On Fri, May 5, 2017 at 11:55 AM, bala murugan wrote: > Hi , > > Can someone please help me understand what to look for in the > /var/log/asterisk/mmlog to check where the leak is , since on Exit it > throws me millions of line under Exiting with the following memory not > freed. > > need some kno

Re: [asterisk-dev] Asterisk Memory Debugger (MALLOC_DEBUG) and DONT_OPTIMIZE

On Mon, May 8, 2017 at 11:11 AM, bala murugan wrote: > Thanks a lot Richard for the response and explanation > > with allocations i was able to determine where the leak would be by going > through the code(atleast 5 files) . > > But i was looking for an easier way to see which piece of the code i

Re: [asterisk-dev] pjsip: 180 Ringing contains wrong info in Remote-Party-ID

On Mon, May 15, 2017 at 10:45 AM, Steve Murphy wrote: > Hello-- > > I've got complaints that the phones are presenting the wrong info when > making an outgoing call... instead of displaying the called party info, > it's displaying the caller's info, which is highly uninteresting. I've been > look

Re: [asterisk-dev] pjsip: 180 Ringing contains wrong info in Remote-Party-ID

On Mon, May 15, 2017 at 4:57 PM, Steve Murphy wrote: > Hmmm, according to your refs, none really apply to this situation. > > > the callerid of the target phone is set in the pjsip channel driver > config, not in my dialplan (the same as chan_sip): > > And, my dialplan doesn't care about the

Re: [asterisk-dev] New feature for chan_pjsip : res_pjsip_session : Using DNID variable to change to uri number part

On Thu, May 18, 2017 at 12:39 AM, Yasin CANER wrote: > Hello, > After chan_pjsip is added in asterisk channels and asterisk > improvement goes to chan_sip to chan_pjsip , i tried to move my network to > chan_pjsip. one feature has chan_sip but not chan_pjsip that i use , > exclamation

Re: [asterisk-dev] Updating call limits

On Mon, Jun 12, 2017 at 8:10 AM, Kaloyan Kovachev wrote: > Hi, > I need to update the call limit of an active call, but the old method of > updating "struct ast_bridge_config" is not working anymore. > The fields 'timelimit', 'play_warning' and 'warning_freq' are used just > to populate the new

Re: [asterisk-dev] Updating call limits

ook's timeout isn't enough. You will have to remove it and re-add it to the heap. Otherwise you are likely to corrupt the heap. Richard > > > > На 2017-06-14 01:01, Richard Mudgett написа: > > On Mon, Jun 12, 2017 at 8:10 AM, Kaloyan Kovachev >> wrote: >> >&g

Re: [asterisk-dev] Feature discussion: extract REFER headers into dialplan.

On Thu, Jul 27, 2017 at 10:31 PM, Kirill Katsnelson wrote: > There were users asking for this feature for as long as I have been using > Asterisk, albeit the requests are infrequent. Basically, when the > transferrer initiates transfer with a SIP REFER message, we want access to > some interestin

Re: [asterisk-dev] ASTERISK-27147 - prune_on_boot - ps_contacts - Sanity Check

On Tue, Aug 15, 2017 at 4:50 AM, Ross Beer wrote: > Hi Richard, > > > Can I run something past you regarding the 'prue_on_boot' logic, please? > > > If multiple asterisk servers are using the ps_contacts database table, > does the removal logic check the 'reg_server' before removing the contact >

Re: [asterisk-dev] Where is pjsip_evsub_set_uas_timeout?

On Mon, Nov 6, 2017 at 6:00 AM, Alexander Traud wrote: > > An earlier approach was to add support for setting pjproject's timer > > (via a pjproject patch) and while that patch is still included here, > > we don't use that call at the moment." > > OK, and what do we do with this code now? > I am

Re: [asterisk-dev] bridge_builtin_features.c playback to both channels in bridge

On Mon, Nov 20, 2017 at 7:03 AM, Steve Davies wrote: > Hi, > > Perhaps the answer to this will be pointing me at some documentation - > That is fine, but I've failed to find it so far, so forgive me if the > following is a dumb question. > > With Asterisk-11 I added a built-in feature which allow

Re: [asterisk-dev] Asterisk 13.17.1 Crash on ConfBridge - NetGen ATA

On Tue, Dec 19, 2017 at 1:45 PM, Bryant Zimmerman wrote: > We are having an issue with asterisk 13.17.1 dumping when a call from a > NetGen Smart ATA drops into a confbridge > The call props up and then things just go wrong. I have talked with the > support guys at NetGen and they have requested

[asterisk-dev] Problems with the ASTERISK-27206 patch.

The patch for https://issues.asterisk.org/jira/browse/ASTERISK-27206 which is committed in 7385d1e017e562afe64431606e857e704f86a16d causes a configuration regression by requiring the endpoint and identifier method to agree to match the endpoint. Doing so is redundant for methods that explicitly sp

Re: [asterisk-dev] Problems with the ASTERISK-27206 patch.

On Tue, Jan 2, 2018 at 5:41 PM, Joshua Colp wrote: > On Tue, Jan 2, 2018, at 7:14 PM, Richard Mudgett wrote: > > The patch for https://issues.asterisk.org/jira/browse/ASTERISK-27206 > which > > is committed in 7385d1e017e562afe64431606e857e704f86a16d causes a > > con

Re: [asterisk-dev] Problems with the ASTERISK-27206 patch.

On Wed, Jan 3, 2018 at 5:08 AM, Joshua Colp wrote: > On Wed, Jan 3, 2018, at 12:03 AM, Richard Mudgett wrote: > > On Tue, Jan 2, 2018 at 5:41 PM, Joshua Colp wrote: > > > > > On Tue, Jan 2, 2018, at 7:14 PM, Richard Mudgett wrote: > > > > The patch for htt

Re: [asterisk-dev] timeval.tv_sec is time_t. How to print/scan?

On Sat, Feb 17, 2018 at 5:58 AM, Alexander Traud wrote: > While compiling Asterisk on OpenBSD, I get a lot of warnings about the > usage of data-type time_t, when used in combination with printf/scanf, > for example in the file res_http_media_cache.c: > > struct timeval actual_expires = ast_tvnow

Re: [asterisk-dev] ast_monitor_start has no effect.... sometimes

On Fri, Mar 23, 2018 at 3:00 PM, Steve Murphy wrote: > Hello-- > > I guess I have a weird situation, in that we use a separate process to > turn on call recording for a channel. It gets bridge join events via AMI > and send a StartMonitor action via AMI back to asterisk. > > The trouble is, that

Re: [asterisk-dev] The "Busy" App.... isn't.

On Wed, Apr 4, 2018 at 6:04 PM, Steve Murphy wrote: > > > On Wed, Apr 4, 2018 at 3:20 PM, Matt Fredrickson > wrote: > >> On Mon, Apr 2, 2018 at 9:40 PM, Steve Murphy wrote: >> > Someone complained about errant behavior of the Busy and Hangup apps... >> > and I see some strangenesses that might

Re: [asterisk-dev] Unused channel tech ast_kill_tech ?

On Tue, Jun 19, 2018 at 8:42 AM, Jean Aunis wrote: > Hello, > > Digging through Asterisk's code, I stumbled upon a strange channel > technology called "ast_kill_tech" (take a look at channel.c / channel.h). > There is a note stating it is used by Zombie channels, but actually it does > not seem t

Re: [asterisk-dev] Regarding Unique Id In Logs

You are trying to reimplement callid[1] which has been in Asterisk since v11. Callid is accessible from dialplan using CHANNEL(callid)[2]. Accessibility from the dialplan has been in Asterisk 13 since 13.15.0 and in Asterisk 15 since it was first released. The callid is created when the incoming

Re: [asterisk-dev] Regarding Unique Id In Logs

e kindly can you re-verify our approach to this. > > On Tue, Jul 3, 2018 at 2:32 AM Richard Mudgett > wrote: > >> You are trying to reimplement callid[1] which has been in Asterisk since >> v11. Callid >> is accessible from dialplan using CHANNEL(callid)[2]. Acces

Re: [asterisk-dev] Deadlock between transaction and transport

On Thu, Jul 12, 2018 at 5:01 PM, Salahuddin Ahmed wrote: > Hello, > > Recently I have experienced a deadlock situation in Asterisk Version > 13.20.0. According to our gdb analysis there have 3 suspicious thread(3, 11 > and 44). Thread 44 push a task to taskpool, thread 11 execute that task and >

Re: [asterisk-dev] Codec negotiation when incoming re-INVITE has no SDP (ASTERISK-28036)

On Sun, Sep 9, 2018 at 8:23 PM, Daniel Harper wrote: > It has been recommended that I bring this up in order to get some > feedback on ways to move forward regarding this feature "When > recieving an re-Invite without SDP asterisk can re-offer all available > codecs supported" > > See https://iss

Re: [asterisk-dev] Controlling the flow of connected line info after attended transfer of stasis-controlled channels

On Tue, Oct 30, 2018 at 2:02 PM Stephen Davies wrote: > Hi, > > I'm looking to see if I've missed something - if not I think there is an > ARI limitation worth thinking about. > > app_dial offers a flag "I" which is supposed to block propagation of > connected line updates through the dial bridge

Re: [asterisk-dev] Controlling the flow of connected line info after attended transfer of stasis-controlled channels

On Tue, Oct 30, 2018 at 3:26 PM Stephen Davies wrote: > > > On Tue, 30 Oct 2018 at 22:20, Stephen Davies > wrote: > > >> Could you comment on why the Dial with ,,I doesn't block the connected >> line update propagating from the called side of the dial back to the >> calling side. That is what i

Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload

On Thu, Oct 3, 2019 at 3:58 PM Michael Maier wrote: > On 03.10.19 at 15:52 Michael Maier wrote: > > On 02.10.19 at 22:45 Sean Bright wrote: > >> On 10/2/2019 4:02 PM, Michael Maier wrote: > >>> I found one more problem regarding the configuration options, provided > by FreePBX, which should be su

Re: [asterisk-dev] AMD

On Thu, Jan 30, 2020 at 5:19 PM Kevin Harwell wrote: > On Thu, Jan 30, 2020 at 2:58 PM i...@magnussolution.com < > i...@magnussolution.com> wrote: > >> in my country when I dial to a mobile and I not answer the call, the call >> going to the voicemail, but the audio is executed before the 200OK,

Re: [asterisk-dev] Add SIP Header with PJSIP in C module

You add headers in a similar way as before. It is just a matter of adding them to the right channel. You must add them to the outgoing channel for PJSIP. This can be accomplished by using pre-dial handlers [1][2]. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers [2] http

Re: [asterisk-dev] Add SIP Header with PJSIP in C module

; "valuetest"); > pbx_exec(chan, "Dial(PJSIP/101@trunk-test,10)"); > > I didn't have any errors but my header is not added. > > Thanks > > > Le mer. 21 oct. 2020 à 12:23, Richard Mudgett a > écrit : > >> You add headers in a similar way as befo