Re: [asterisk-dev] [External] Re: Mailing List Future

I definitely would Von: asterisk-dev im Auftrag von Joshua C. Colp Datum: Mittwoch, 13. Dezember 2023 um 14:42 An: aster...@phreaknet.org Cc: Asterisk Developers Mailing List Betreff: [External] Re: [asterisk-dev] Mailing List Future CAUTION: This email originated from outside of the

Re: [asterisk-dev] Mailing List Future

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 more info in the github releases the trigger is always the mailing list. I also agree with some of the others that the mailing list is

Re: [asterisk-dev] [External] Re: Final Preview: docs.asterisk.org

Looks a lot better already. Thanks George! FLORIAN FLOIMAIR Development Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria Phone: +43 662 85 62 25 Mail: f.floim...@commend.com [signature_1051841226] commend.com LG Salzburg /

Re: [asterisk-dev] [External] Re: Final Preview: docs.asterisk.org

d know the content is safe. On Wed, Aug 9, 2023 at 11:12 AM Floimair Florian mailto:f.floim...@commend.com>> wrote: Just to be precise: By “template” I meant the “Material theme” that is used. I honestly don’t like it for exactly that reason. To me it seems this is optimized for vertical displays

Re: [asterisk-dev] [External] Re: Final Preview: docs.asterisk.org

Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria Phone: +43 662 85 62 25 Mail: f.floim...@commend.com<mailto:f.floim...@commend.com> [signature_646497160] commend.com LG Salzburg / FN 178618z Von: asterisk-dev im Auftrag von Floimair Florian Datum: Mittwoch, 9. August 2

Re: [asterisk-dev] Final Preview: docs.asterisk.org

Hi George! Something I noticed (e.g. on this page: https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Module_Configuration/res_pjsip/) The available space in the browser is used badly with the current layout template. There is a lot of blank space to the left and right,

Re: [asterisk-dev] [External] Re: PLEASE CHECK THE RC RELEASE ARTIFACTS!!

Same here! Also our package builds worked right away perfectly fine. No adaptations were needed. From my perspective this is a GO! Thanks guys! FLORIAN FLOIMAIR Development Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria Phone: +43 662 85 62 25 Mail:

Re: [asterisk-dev] [External] Re: Question regarding http_media_cache and astcachedir setting

sk-dev] Question regarding http_media_cache and astcachedir setting CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. On Wed, Feb 8, 2023 at 2:08 PM Floimair Florian mailto:f.floi

[asterisk-dev] Question regarding http_media_cache and astcachedir setting

Hey there! I have a question regarding res_http_media_cache module and the astcachedir setting in asterisk.conf: We see files are cached by this module in /tmp (/tmp/bucket-.wav16) regardless of what we set for astcachedir in asterisk.conf (tried /tmp/asterisk). Is this a bug? FLORIAN

Re: [asterisk-dev] [External] Re: How to debug reinvites not getting forwarded to other call leg (pjsip)

unless you recognize the sender and know the content is safe. On Wed, Jan 12, 2022 at 10:57 AM Floimair Florian mailto:f.floim...@commend.com>> wrote: Hi Joshua! No it does not concern audio in this case, it is a change in media for video. Initially the call is established with a=recvonly (or

Re: [asterisk-dev] [External] Re: How to debug reinvites not getting forwarded to other call leg (pjsip)

terisk-dev] [External] Re: How to debug reinvites not getting forwarded to other call leg (pjsip) CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. On Wed, Jan 12, 2022 at 10:57 A

Re: [asterisk-dev] [External] Re: How to debug reinvites not getting forwarded to other call leg (pjsip)

om outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. On Wed, Jan 12, 2022 at 10:17 AM Floimair Florian mailto:f.floim...@commend.com>> wrote: Hi everybody! I am currently facing an issue with SIP reINVITEs (with changed media

[asterisk-dev] How to debug reinvites not getting forwarded to other call leg (pjsip)

Hi everybody! I am currently facing an issue with SIP reINVITEs (with changed media direction) being acknowledged by Asterisk but not forwarded to the second call leg. My setup is as follows: Device A -> Kamailio -> Asterisk (18.9.0 chan_pjsip) -> Kamailio -> Device B Device A sends a

Re: [asterisk-dev] [External] Re: Codec order changing after hold and unhold

Thanks Joshua! I will look into it. >>There is an open issue[1] for this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit:

[asterisk-dev] Codec order changing after hold and unhold

Hi there! I was wondering if anyone has any idea what the cause of the following behavior could be: 1. We place a call from A to B with codecs (in this order on both sides) opus,g722,g711a,g711u 2. B accepts the call 3. B puts A on hold 4. B puts A off hold 5. Asterisk now

Re: [asterisk-dev] Advanced Codec Negotiation: Need info and uses cases

> Transcoding only if there is an allowed and supported codec on both sides > which are not common. If there is one common codec on both sides - use > this codec and do not transcode. If Alice (OpenStage20, Local Net) offers G.722, alaw and ulaw and Bob (Remote Localtion)

Re: [asterisk-dev] Questions on feature/patch process

Yes, you need to cherry-pick your patch to branch 16. You can do this directly in gerrit though. Be prepared however, that there might be modifications necessary. I therefore personally prefer to do such things locally first in separate branches and then submit the changes on each branch

Re: [asterisk-dev] Question regarding RTCP - NACK

com/> Security and Communication by Commend FN 178618z | LG Salzburg Am 11.02.19, 11:44 schrieb "asterisk-dev im Auftrag von Joshua C. Colp" : On Mon, Feb 11, 2019, at 5:16 AM, Floimair Florian wrote: > > We recently ran into something regarding RTCP –

[asterisk-dev] Question regarding RTCP - NACK

We recently ran into something regarding RTCP – Generic NACK Feedback, which was introduced with https://issues.asterisk.org/jira/browse/ASTERISK-27810 Me and my colleague read through RFC 4585 Section 6.2.1 “Generic NACK” but we didn’t really find an answer to our question, which is the

Re: [asterisk-dev] FW: Question about permit/deny in pjsip endpoints

No one with any idea about this? - UPDATE: Additionally, I see this message in the logs: ["2018-10-25 07:52:40.6837"] WARNING[123176]: named_acl.c:333 ast_named_acl_find: ACL 'deny/permit' does not exist. The ACL will be marked as undefined and will automatically fail if applied.

[asterisk-dev] FW: Question about permit/deny in pjsip endpoints

UPDATE: Additionally, I see this message in the logs: ["2018-10-25 07:52:40.6837"] WARNING[123176]: named_acl.c:333 ast_named_acl_find: ACL 'deny/permit' does not exist. The ACL will be marked as undefined and will automatically fail if applied. Hello all! I wanted to try limiting incoming

[asterisk-dev] Trigger chan_pjsip NOTIFY via ARI

I did a quick search, but so far haven't found anything aside from triggering notify via Asterisk CLI. So my question is: Is there any way to trigger a NOTIFY via ARI? Any help much appreciated With best regards Florian Floimair Innovation - Software-Development COMMEND

Re: [asterisk-dev] Question regarding SIP MESSAGE log verbosity in chan_pjsip

<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg Am 05.07.18, 21:10 schrieb "asterisk-dev im Auftrag von Matt Fredrickson" : On Tue, Jul 3, 2018 at 7:18 AM, Floimair Florian wrote: > I’m not exactly sure if the cur

[asterisk-dev] Question regarding SIP MESSAGE log verbosity in chan_pjsip

I’m not exactly sure if the current implementation (tested with 15.4.1) of SIP MESSAGE in chan_pjsip is logging with the correct loglevel. E.g.: If a SIP MESSAGE is sent to an endpoint (in my case via ARI) where there is currently no registered contact (the phone is offline), Asterisk throws an

Re: [asterisk-dev] Dual contact entries in "pjsip show contacts"

"pjsip show contacts" On Thu, Mar 22, 2018, at 7:22 AM, Floimair Florian wrote: > Hi! > > I am currently using an Asterisk setup with a realtime-db as backend. > We create endpoints/aors/auths using ARI Push configuration and when > that happens we see 2 contact ent

[asterisk-dev] Dual contact entries in "pjsip show contacts"

Hi! I am currently using an Asterisk setup with a realtime-db as backend. We create endpoints/aors/auths using ARI Push configuration and when that happens we see 2 contact entries when executing "pjsip show contacts" in the CLI. This also sporadically leads to calls being refused by Asterisk

Re: [asterisk-dev] Asterisk 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3 Now Available (Security)

Pjproject was rather quick this time around. Your patches 0070 & 0071 (in third-party/pjproject/patches) have already been incorporated and released in 2.7.2. So I guess you can update the reference to 2.7.2 and remove the patches in the branches. With best regards Florian Floimair

Re: [asterisk-dev] PJSIP endpoints created via ARI do not transfer some values into Realtime Database

into Realtime Database On Thu, Nov 9, 2017, at 10:53 AM, Floimair Florian wrote: > Joshua, > > I'm pretty sure I have found the problem, and actually I'm not the > first to encounter it. > The solution is to use OPT_YESNO_T in pjsip_configuration.c instead of > OPT_BOOL_T for the

Re: [asterisk-dev] PJSIP endpoints created via ARI do not transfer some values into Realtime Database

: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] Im Auftrag von Floimair Florian Gesendet: Mittwoch, 8. November 2017 13:47 An: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Betreff: Re: [asterisk-dev] PJSIP endpoints created via

Re: [asterisk-dev] PJSIP endpoints created via ARI do not transfer some values into Realtime Database

via ARI do not transfer some values into Realtime Database On Wed, Nov 8, 2017, at 08:13 AM, Floimair Florian wrote: > Hi guys! > > In my lab environment I ran into an issue with PJSIP endpoints created > via ARI with a realtime database as backend. > I'm using a MySQL database

[asterisk-dev] PJSIP endpoints created via ARI do not transfer some values into Realtime Database

Hi guys! In my lab environment I ran into an issue with PJSIP endpoints created via ARI with a realtime database as backend. I'm using a MySQL database in my lab setup. I noticed that some of the values of the endpoint object do not find their way into the database. When creating the endpoint

Re: [asterisk-dev] Epcot trip before DevCon

Nice! I was planning on doing the trip to NASA on Friday anyway so I’d be willing to join. I just have to check the planned time as my flight back to Europe is leaving at 8pm on Friday. With best regards Florian Floimair COMMEND INTERNATIONAL GMBH Security and Communication by Commend FN

Re: [asterisk-dev] Problems authenticating to gerrit.asterisk.org via ssh

Security and Communication by Commend FN 178618z | LG Salzburg Von: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] Im Auftrag von Floimair Florian Gesendet: Donnerstag, 24. August 2017 16:30 An: Asterisk Developers Mailing List <asterisk-dev@lists.digium.

[asterisk-dev] Problems authenticating to gerrit.asterisk.org via ssh

Hey guys! Since I already have a signed contributor license agreement, I wanted to submit a patch for ASTERISK-27168 to gerrit. I proceeded as described in https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage including signing in and adding my public key for SSH. However I am getting a

Re: [asterisk-dev] Asterisk 15 Beta Released

I agree! I would very much welcome releasing 14.6 as an LTS release as well. With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg Von: