Re: [asterisk-dev] Mailing List Future

Hi, Just looking into this in more detail: Return-path: So yea, that's VERP based. From: "Joshua Colp via groups.io" And that's a very basic form of SRS ... And these headers are present too: List-Subscribe:

Re: [asterisk-dev] Mailing List Future

Hi Joshua, On 2023/12/13 14:30, Joshua C. Colp wrote: On Wed, Dec 13, 2023 at 8:12 AM Floimair Florian wrote: I agree! To me the mailing list is the best source of gathering information, especially in terms of announcements of new Release versions. While there might be

Re: [asterisk-dev] Mailing List Future

Hi, My 5c.  Killing the dev list is a bad idea. Most developers could not care about having to poll forums.  It also means that stuff that would previously get an audience will now get none. github discussions are better than forums at least. May I inquire as to the problem you're having

Re: [asterisk-dev] libjwt is being added to the 'third-party' packages (Attention Asterisk Package Maintainers!)

Hi, Thanks.  Automagic dependency warning then to packagers, explicitly select/unselect res_stir_shaken :). Kind regards, Jaco On 2023/09/26 17:16, George Joseph wrote: On Tue, Sep 26, 2023 at 8:56 AM Jaco Kroon wrote: Hi George, Is the default to build STIR/SHAKEN if libjwt

Re: [asterisk-dev] libjwt is being added to the 'third-party' packages (Attention Asterisk Package Maintainers!)

Hi George, Is the default to build STIR/SHAKEN if libjwt is found, or will it fail by default? In other words, on systems where libjwt is not available, is special action required to order to build?  Does this vary based on whether libjwt can be found or not? Kind regards, Jaco On

Re: [asterisk-dev] Deprecating users.conf

Hi, On 2023/06/30 14:19, Sean Bright wrote: Hi, On 6/30/2023 7:45 AM, aster...@phreaknet.org wrote: I've put up a PR to deprecate users.conf[1], following a discussion earlier this year about this, but I think that was on IRC so wanted to discuss here as well. Apologies - I realized after

Re: [asterisk-dev] Removing configure from tree

Hi Joshua, My 5c below. On 2023/05/04 11:59, Joshua C. Colp wrote: On Thu, May 4, 2023 at 6:45 AM Dennis Buteyn wrote: On 5/3/23 22:48, Joshua C. Colp wrote: Greetings all, Sean Bright brought up on IRC the removal of configure from git. This may be impactful since some

Re: [asterisk-dev] stupid error in configuration - found what causes this

https://gerrit.asterisk.org/c/asterisk/+/19712 https://gerrit.asterisk.org/c/asterisk/+/18689/ Either one of those PRs will sort you out for now. Kind Regards, Jaco Kroon On 2023/01/17 18:43, Wojciech Puchar wrote: if i disable jitterbuffer in iax.conf ringback (bo bo) works

Re: [asterisk-dev] Logging and multi-tenancy

Status Page here <https://status.primevox.net/> ---- *From:* Jaco Kroon *Sent:* Wednesday, January 18, 2023 1:31 AM *To:* Asterisk Developers Mailing List ; Luke Escudé *Subject:* Re: [asterisk-dev] Logging and mult

Re: [asterisk-dev] Logging and multi-tenancy

Hi Luke, On 2022/12/23 18:48, Luke Escudé wrote: So, it's possible to achieve multi-tenancy in Asterisk with well-designed dial plan. By multi-tenancy, in terms of Asterisk, I mean each "customer" has their own separate dial plan, and each customer cannot "see" another. As long as each

Re: [asterisk-dev] Methodologies for validating dialplan

Hi, I find this discussion very interesting.  I do have one question, and this was implied by another poster: How do you intend to deal with dynamic stuff, for example: exten => ??,n,Goto(foo-${bar}) In this case, what's the possible values for ${bar}? Would it make sense to enhance the

Re: [asterisk-dev] Manual endpoint registration

Hi Luke, If you're in a position to migrate astdb along with the instance.  So what we have is astdb in ramdisk, upon *clean* asterisk shutdown we copy that to a clustered filesystem, and prior to startup clone that back onto ramdisk. This works for chan_sip at least. You can presumably make

Re: [asterisk-dev] Clear a secret from sip.conf and phone sill can dial

Hi, Without secret= the phone is never challenged for authentication and the initial invite is accepted on that basis. You can use tcpdump or (my preference for these kind of things) sngrep to verify. Kind Regards, Jaco n 2021/07/05 13:49, SAMPro wrote: > I have set an extension in sip.conf

Re: [asterisk-dev] Asterisk console hangs up

Hi Rajesh, On 2021/04/27 05:46, Rajesh wrote: > We are using Asterisk 13.18.2. Asterisk CLI hangs up intermittently. -- Do you have any additional information?  Last messages?  Anything really. I've seen that it "hangs" during shutdown, but haven't analysed the stack traces at that point yet. 

Re: [asterisk-dev] func_odbc auto commit at answer time

Hi Dennis, Thank you.  Becoming more and more nasty by the moment ... On 2021/02/24 10:07, Dennis Buteyn wrote: > On 2/24/21 1:01 AM, Jaco Kroon wrote: >> >> Hi, >> >> I still think the changes for forcecommit should be done for >> func_odbc, however, after

Re: [asterisk-dev] func_odbc auto commit at answer time

nd Regards, Jaco On 2021/02/22 05:09, Jaco Kroon wrote: > > Hi, > > On 2021/02/21 13:46, Dennis Buteyn wrote: >> On 2/19/21 1:03 PM, Jaco Kroon wrote: >> >> >> >>> The specific ODBC connection here is merely used for inserting "traces" >

Re: [asterisk-dev] func_odbc auto commit at answer time

Hi, On 2021/02/21 13:46, Dennis Buteyn wrote: > On 2/19/21 1:03 PM, Jaco Kroon wrote: > > > >> The specific ODBC connection here is merely used for inserting "traces" >> into the database, so exactly this: >> >> [trace] >> dsn=ulsdb >

Re: [asterisk-dev] func_odbc auto commit at answer time

Hi Dennis, Thank you for taking the time. On 2021/02/18 16:48, Dennis Buteyn wrote: > On 2/18/21 12:53 PM, Jaco Kroon wrote: >> Hi All, >> >> So I've been updating some code to use func_odbc vs previous solutions >> and have bumped into a few concerns (1 I'd appreciat

[asterisk-dev] func_odbc auto commit at answer time

Hi All, So I've been updating some code to use func_odbc vs previous solutions and have bumped into a few concerns (1 I'd appreciate feedback on the review, 2 is prelude to 3, which is the main issue currently): 1.  ARGn from Gosub() getting leaked into func_odbc if adequate ARG parameters are

Re: [asterisk-dev] func_cdr performance issue

Hi, On 2021/01/11 13:47, Joshua C. Colp wrote: > On Mon, Jan 11, 2021 at 7:38 AM Jaco Kroon <mailto:j...@uls.co.za>> wrote: > > Hi All, > > I am very much new to the way in which statis functions.  Is there > documentation somewhere around the design of thi

[asterisk-dev] func_cdr performance issue

Hi All, I am very much new to the way in which statis functions.  Is there documentation somewhere around the design of this I should be looking at? Our primary issue is that both cdr_read and cdr_write in func/func_cdr.c gets very slow (up to four seconds, sporadically).  We're using cdr_odbc

[asterisk-dev] elimination of -fnested-functions on gcc + -fblocks on clang

Hi All, Long story short:  I'd like to potentially eliminate the use of those options.  They're used for RAII_VAR (and only RAII_VAR assuming I'm not missing something obvious). Long story: They're use is in conjuction with __attribute__((cleanup(...))) (which is supported by both gcc and

Re: [asterisk-dev] Pain Points For Large Scale Instance Provisioning

ould be great.  This is actually something I should be able to attend to. Kind Regards, Jaco On 2020/10/21 10:35, Jaco Kroon wrote: > Hi, > > On 2020/10/20 23:32, Michael Cargile wrote: > >> Towards the end of DevCon, Matt asked if there were any pain points >> for provisioni

Re: [asterisk-dev] Pain Points For Large Scale Instance Provisioning

Hi, On 2020/10/20 23:32, Michael Cargile wrote: > Towards the end of DevCon, Matt asked if there were any pain points > for provisioning large numbers of Asterisk instances and I mentioned I > would talk to my colleague who handles such things. He provided this > list: > > * Sanity checks within

Re: [asterisk-dev] ICE and STUN address calculations

Hi Alexander, That is interesting, but I don't see exactly how that related to my original problem statement? I can see the problem you have there, but I'm not sure what you're proposing? BTW: regarding the below:  https://gerrit.asterisk.org/c/asterisk/+/13975 Which doesn't change anything

[asterisk-dev] ICE and STUN address calculations

Hi All, Following up on some work I've been busy with (https://gerrit.asterisk.org/c/asterisk/+/13362) a few potential issues came to the fore on which I need some input on what the "correct" behaviour should be. Note that none of this affect me directly (currently). For the purposes of this

Re: [asterisk-dev] [BOUNTY] Confbridge feature

Hi, Please see below. Kind Regards, Jaco Kroon C.E.O. *T:* +27 (0)12 021 | *F:* +27 86 648 8561 | *E:* j...@iewc.co.za *W:* iewc.co.za <https://www.iewc.co.za/> | *A:* Unit 201, Building 2B, Sunwood Park, Queen's Crescent Lynnwood, Pretoria   Facebook <https://www.fac

Re: [asterisk-dev] [BOUNTY] Confbridge feature

ils. Then there you simply do: [ringers] exten => s,1,Dial(PJSIP/101/102) Kind Regards, Jaco Kroon C.E.O. *T:* +27 (0)12 021 | *F:* +27 86 648 8561 | *E:* j...@iewc.co.za *W:* iewc.co.za <https://www.iewc.co.za/> | *A:* Unit 201, Building 2B, Sunwood Park, Queen's Crescent Lynnwo

Re: [asterisk-dev] Asterisk 16.9.0-rc1 Now Available - unusable with Deutsche Telekom

Hi, On 2020/03/09 20:04, George Joseph wrote: <--snip--> > > BTW: > While you're at it: it would be a great oportunity to get it > sorted out > completely by globaly adding session stability (one trunk always uses > same IP destination for all actions. If the destionation

[asterisk-dev] hashtab memory corruption and resulting segfaults due to lack of thread safety

Hi All, So I've received a few backtraces from a customer (Still using ast 11 ... we're in progress of getting them migrated).  They all follow this pattern (full stack below): #0  ast_hashtab_lookup_internal (tab=0x7f41bc277920, tab=0x7f41bc277920, h=, obj=0x7f4203176a50) at hashtab.c:602  

Re: [asterisk-dev] acl.c - option to turn off logging

. There are then two wrapper functions, one with identical interface to existing, which if purpose is given as NULL it'll pass "" to _internal (thus enabling logging as per current implementation), and _nolog variant which will always pass log_prefix = NULL. Kind Regards, Jaco Kroon On 2019/1

[asterisk-dev] acl.c - option to turn off logging

ing to embed the ACL into rtp.conf. Kind Regards, Jaco Kroon C.E.O. *T:* +27 (0)12 021 | *F:* +27 86 648 8561 | *E:* j...@iewc.co.za *W:* iewc.co.za <https://www.iewc.co.za/> | *A:* Unit 201, Building 2B, Sunwood Park, Queen's Crescent Lynnwood, Pretoria   Facebook <https://www.

Re: [asterisk-dev] pjsip and RTP

Hi, On 2019/11/27 13:16, Jaco Kroon wrote: > > Hi Joshua, > > On 2019/11/26 17:19, Joshua C. Colp wrote: <--snip--> Regarding ice_blacklist: >> It's strictly a blacklist currently, not a deny/allow list. > > I'd like to make an attempt at this.  I'm thinking make

Re: [asterisk-dev] *** SPAM *** Re: pjsip and RTP

Hi Joshua, On 2019/11/26 17:19, Joshua C. Colp wrote: > On Tue, Nov 26, 2019 at 11:14 AM Jaco Kroon <mailto:j...@uls.co.za>> wrote: > > Hi Joshua, > > On 2019/11/25 16:59, Joshua C. Colp wrote: >> On Mon, Nov 25, 2019 at 10:44 AM Jaco Kroon >

Re: [asterisk-dev] pjsip and RTP

Hi Joshua, On 2019/11/25 16:59, Joshua C. Colp wrote: > On Mon, Nov 25, 2019 at 10:44 AM Jaco Kroon <mailto:j...@uls.co.za>> wrote: > > Hi, > > chan_pjsip has some issues w.r.t. RTP that's causing problems for > me.  My setup is rather unique, so

[asterisk-dev] pjsip and RTP

Hi, chan_pjsip has some issues w.r.t. RTP that's causing problems for me.  My setup is rather unique, so I also get why few (if any) others have bumped into this before. I'm trying to wrap my head around PJSIP and ICE. I've got hosts with 200+ IP addresses.  I need a way to restrict the

Re: [asterisk-dev] REFER too soon?

Hi Dennis, I've your going to issue a REFER directly after 200 OK, why not simply 302 redirect instead? Kind Regards, Jaco On 2019/08/29 13:18, Joshua C. Colp wrote: > On Thu, Aug 29, 2019, at 7:45 AM, Dennis Buteyn wrote: >> Been trying to perform a blind transfer for hours with a SIP client

Re: [asterisk-dev] PJSIP and RTP address selection

Hi Joshua, Thanks for giving this some thought, I really do appreciate. I'll aim towards implementing both the first options, users can then pick for themselves.  I think both the options I've proposed could work, but one may have benefits over the other that I can't envison currently. I did

[asterisk-dev] PJSIP and RTP address selection

Hi, I've got a scenario where (when using PJSIP, using chan_sip does what I expect) PJSIP will advertise one address in the SDP during a conversation but then start transmitting from another.  In my case PJSIP is advertising 197.96.209.1 in the SDP, but 197.96.209.251 is being used to send. I

[asterisk-dev] chap_sip guest over tcp + insecure=port

Hi All, The following bugs refers: ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP commit b2c4e8660a9c89d07041271371151779b7ec75f6 ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentification error change set

Re: [asterisk-dev] ast_safe_system() from within chan_sip.c

Hi, And please keep in mind that SIP calls (in asterisk 11 at least) are very much serialized and you're going to affect the response times to SIP frames by introducing extra work there - really do look at the ASF. Frankly, if you're after what I think you're after, I've been parsing the logs

Re: [asterisk-dev] Lockups in Asterisk 11

main/io.c:292 ast_io_wait() (0x8132d50+175) > channels/chan_sip.c:29484 do_monitor() > main/utils.c:1223 dummy_start() > :0 start_thread() > libc.so.6 clone() (0xb77270d0+5E) > === --- > === >

Re: [asterisk-dev] Lockups in Asterisk 11

Hi Mark, I suspect the following relates: http://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests That should explain the underlying problem for you, and potentially provide you with a fix depending on your situation. We've been using the proposed

Re: [asterisk-dev] [Code Review] 4599: chan_iax2 - unprotected access of iaxs[peer-callno] potentially results in segfault

years) in multiple production environments. Many fewer segfaults. Thanks, Jaco Kroon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-dev] [Code Review] 4599: chan_iax2 - unprotected access of iaxs[peer-callno] potentially results in segfault

://reviewboard.asterisk.org/r/4599/diff/ Testing --- Been running since 11.2.1 with this patch, specifically 6/3/2013 (more than two years) in multiple production environments. Many fewer segfaults. Thanks, Jaco Kroon

Re: [asterisk-dev] [Code Review] 4599: chan_iax2 - unprotected access of iaxs[peer-callno] potentially results in segfault

. Thanks, Jaco Kroon -- _ -- 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] res_fax_spandsp segfaults during fax detection - FIXED?

lly, just double checked, ulaw/alaw is (was now) allowed, so someone is possibly trying to run in bypass mode, resulting in the t38 gateway instead of t38 pass through. I downgraded to 11.6.0 and hadn't had a crash since but I opted to dis

Re: [asterisk-dev] [HELP]: Auto-answering calls placed from call files

Pass it via a Local/ channel that does the setup. Think this really belongs on the users list though, not the dev list. Kind Regards, Jaco Kroon On 28/01/2014

Re: [asterisk-dev] scalability issue with realtime lastms update on qualify state change

Hi, On 16/11/2013 10:02, Olle E. Johansson wrote: 16 nov 2013 kl. 00:57 skrev Damon Estep da...@soho-systems.com mailto:da...@soho-systems.com: I typically have 250 concurrent channels active per server. RTP is not handled in the sip channel, but qualify and the realtime database