[asterisk-dev] Code Coverage?

When going for a new version of the compiler (GCC and Clang) or Doxygen, those tools, again and again, improve their syntax checker and create warnings for common issues. Avoiding such warnings is quite handy because they can reveal actual bugs. In Asterisk, some parts of the code are not

Re: [asterisk-dev] ASTERISK_VERSION_NUM

Well, the source code which did it in Asterisk 11, is still in the commit history. A roll-back of that should be the easiest solution. The question was more about double-checking whether there is a replacement for ASTERISK_VERSION_NUM, which I missed. If, after three Asterisk LTS releases, I

[asterisk-dev] ASTERISK_VERSION_NUM

Not at runtime but at compile-time, how does the compiler of a module detect which version is used? Before Asterisk 11, I included and the preprocessor macro ASTERISK_VERSION_NUM was available, see [1]. How do I solve that today? My workaround, because I build in-tree anyway, I patched its

Re: [asterisk-dev] Asterisk 19: res_adsi built although deprecated?

> Please create an issue for this on the tracker. There, we go: The current maintainer of that module contacted me off-list. Because res_adsi has a maintainer, it is not going to be removed and is not "deprecated" but "extended"

[asterisk-dev] Current max verbose (10) and max debug level (10): Intended?

While grepping through the source code, I noticed that there are 15 places which emit debug messages at level 10. One place is debug level 9 (apps/app_chanspy). Just six places are debug level 6 (main/dnsmgr and res/res_statsd). There are no level 7 and 8. When it comes to verbose, one place

[asterisk-dev] How to debug config-file parsing? Or XML documentation?

Currently, I try to debug the above issue, just to understand Asterisk a bit better. However, if you spot the cause, do not hesitate to patch or comment at that issue. If I have to continue with that issue, I have to understand more: Is

[asterisk-dev] Asterisk 19: res_adsi built although deprecated?

While creating a minimal installation of the upcoming Asterisk 19 with ./configure make full sudo make install samples config many notices, warnings and even errors pop up on start. Still too many and they get more. For example, the module res_adsi is built on default although it is deprecated.

[asterisk-dev] PJSIP: Is SDP answer? Is SDP offer?

In the function res_pjsip_sdp_rtp:setup_sdes_srtp(.), an upcoming patch of mine needs to determine whether that code processes an SDP answer or an SDP offer. Is there (somewhere) a flag or condition which I can check? In chan_sip, I can use AST_LIST_EMPTY(sip_pvt->offered_media). Does a similar

[asterisk-dev] Security Patches for third-party/patches: How to force a fresh build?

Some folks might not download the whole Asterisk, apply their patches, and build Asterisk but download just the last diff/patch, apply that, and re-build Asterisk. Those diffs/patches are available via which is terrible handy when you

Re: [asterisk-dev] Webpage: HTTPs Mixed-Content: Where to report?

Still a problem. And a new problem: As "Test Release" a former version of Asterisk 17 is shown now again. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-dev] Scope Trace: Often no CR on CLI?

Thanks for clarifying that this should work. Let me start GDB ... the problem is ast_debug(.) with SCOPE_EXIT_RTN_VALUE(0) in channels/chan_pjsip.c:2594, for example. Then, ast_debug(.) is called just with a space which is not stripped. Consequently, the cause is

[asterisk-dev] Scope Trace: Often no CR on CLI?

Currently, I debug something with the new Scope Tracing [1][2] on the Command-Line Interface (CLI) within GDB. Six of the printed messages did not wrap to a new line. In all cases, SCOPE_ did not include a message, and therefore no trailing "\n". With a Regular Expression, I found 32 calls that

Re: [asterisk-dev] =?UTF-8?B?44CQcmVzX3Bqc2lwX3JlZ2lzdHJhcg==?=.c: Error】

> pj_strcspn2 The fix for raised the required version of the PJ Project to 2.6, see [1]: A) go for an external PJ Project, like 2.10 B) go for ./configure --with-pjproject-bundled, see [2] C) reverse the change for ASTERISK-28641, see [3]

Re: [asterisk-dev] Webpage: HTTPs Mixed-Content: Where to report?

> I have forwarded it on. Great. They fixed their feedback page. Now, right now, the DNS resolution of the whole asterisk.org universe does not work for me, even manually set to Cloudflare 1.1.1.1 or Google Public DNS. I had to use your authoritive nameserver "nsx1.digium.com" directly.

Re: [asterisk-dev] Asterisk tries to transcode vp8 <> g729

> codecs [have to be] in "[allow=]audio,video[,text]" order Did you file an issue report for this? If yes, please, mark my just created report as duplicate: In any case, good catch! Yes, Digium Asterisk still assumes that the *first*

Re: [asterisk-dev] Webpage: HTTPs Mixed-Content: Where to report?

> Did not knew that. Great! But where is that linked? Now that I know, I found it via a Community message of yours from September only. By the way, while searching, I found the Sangoma Webpage and

[asterisk-dev] Webpage: HTTPs Mixed-Content: Where to report?

Shows no padlock-icon in Apple Safari and a mixed-content warning in Mozilla Firefox. That background images with the foggy woods at the header and that white snow at the footer are loaded not via HTTPs but HTTP. I am a bit

[asterisk-dev] What is test-law?

In 2009, Asterisk 1.8 introduced a media format called "G.711 test-law" [1]. Furthermore, in chan_sip, default_format_capabilities mentioned testlaw. Five years later, Asterisk 13 removed testlaw from there [2]. However, it remained in general. For me, it looks like not just dead code but a

Re: [asterisk-dev] EVS codec for Asterisk 16

> /usr/lib64/asterisk/modules/codec_evs.so Please, 1) do an "ldd" on that. Does it find all dependencies? 2) Which version/variant of Asterisk do you use? -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-dev] Support for amr codec?

Software bugs love to be talked about. That helps them to hide. Bugs hate to be tracked down. That might kill them. > comfort noise, which is not supported by Asterisk In case of AMR(-WB), Comfort Noise (CN) is part of the decoder itself already. Consequently, no additional support in Asterisk

Re: [asterisk-dev] Webpage Downloads

a) Certified Asterisk 16 is listed twice: 16.8 and 16.3. b) The download for 16.3 is broken. The button "Download FreePBX Now" does nothing for me.

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

A bit late but this might be interesting, too: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or

[asterisk-dev] ./bootstrap: Why not done by core members?

When a contributor changes the file configure.ac, or menuselect/configure.ac, or third-party/pjproject/configure.m4, or third-party/jansson/configure.m4, the Asterisk Team forced the contributor to execute "./bootstrap.sh" via code review. This created hotspots in the file

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

Cannot contribute much because I do not use host!=dynamic. Just two comments on that: >> > seems to be part of a potentially larger patch set No, it was just that isolated change. > the old behaviour was correct in that any TCP based protocol can't control

[asterisk-dev] NET::ERR_CERT_SYMANTEC_LEGACY: Re-issue your RapidSSL certificate!

All asterisk.org (sub-) domains are secured by a SSL/TLS certificate from RapidSSL which chains up to the trust anchor "GeoTrust Global CA". That trust anchor belonged to Symantec. Since Chrome 70, Google removes all trust in former Symantec trust anchors. When you re-issue your certificate,

Re: [asterisk-dev] Testsuite python3 compatibility

> I'm working off a laptop which gets hotter than I'd like I guess, it is time for a new computer. As noted in the Gerrit review

Re: [asterisk-dev] Testsuite python3 compatibility

I love to help. However, before I apply your patch, I need a running Test Suite first. In Ubuntu 18.04 LTS with branch Master, 11 test cases fail for me, for example ASTERISK-27897. In that report you see my concrete steps. Perhaps you spot the mistake. In Ubuntu 16.04 LTS with branch 15.4, 30

Re: [asterisk-dev] Fwd: PJSIP and Preconditioning (UPDATE prior to 200 OK on INVITE)

> is there a proper way to report bugs in PJSIP? You fix the bug and send the (unified) patch to their E-mail address. Then, when accepted, you add that patch to Asterisk, for example: . At least that is the only way I know for external contributors (to Asterisk

[asterisk-dev] Headers: Incomplete and Cyclic Referenced

The Google tool include-what-you-use (iwyu) works in Asterisk (for more details see ASTERISK-25591). You can pick one single file, it goes through the code and checks that all #include are there. I love iwyu, because it helps me to complete the inclusion headers, especially when I submit a new

[asterisk-dev] ./configure --with-pjproject-bundled: GitHub changed/s to TLS 1.2-only!

Currently, Asterisk downloads the tarball of the PJProject not from the original source at Teluu but GitHub. GitHub forces the user to use HTTPs instead of HTTP. This is already an issue as described in ASTERISK-27665 because many tools authenticate the given certificate. Since February, GitHub

[asterisk-dev] Dead code around HAVE_VIDEO_CONSOLE

In the script configure, while going through all AST_EXT_LIB_SETUP, I find more and more code which is dead. That means, external projects are not used anymore too. This time, this is HAVE_FFMPEG, HAVE_SDL, HAVE_SDL_IMAGE, HAVE_VIDEODEV_H, and HAVE_X11. That code is not maintained anymore

Re: [asterisk-dev] JIRA: 15.4 and 13.21 still tagged as unreleased.

> Updated. Mhm. On Jira, both versions still appear as unreleased, when I create a new bug report and select them with "Affects Version". -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-dev] Testsuite python3 compatibility

> who has available hardware to run the complete testsuite > against Asterisk master Would love to help. What is required exactly? I never got the Test Suite running because of its amount of dependencies. Is there an up-to-date guide anywhere? --

[asterisk-dev] JIRA: 15.4 and 13.21 still tagged as unreleased.

Not sure where to report this elsewhere. By the way, when I know when a bug was introduced, should I * selected the latest release version = tested, or * select the version(s) when the bug was introduced? -- _ -- Bandwidth

[asterisk-dev] Web Downloads

> has a button to download the "latest version". That button states you download Asterisk 13. However, the download link points to Asterisk 15. On 26th January, Matthew Fredrickson wrote: >> >> because Asterisk

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

> As that link shows ... The link explains, that one should not assume any datatype for time_t. Therefore, I would use it like an abstract data-type. However, the Asterisk project might have assumptions about the platform. Then, difftime(.) could be used: .

Re: [asterisk-dev] Authenticated downloads of external stuff?

> The external modules might be problematic since their versions are > only tied to major Asterisk releases. Upps. Did not know that. However, that part does not work in FreeBSD at all. And I do not use it in Ubuntu either. Consequently, it does nobody prevent to secure those other parts. As

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

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(); // gettimeofday(, NULL); snprintf(time_buf, sizeof(time_buf),

Re: [asterisk-dev] Authenticated downloads of external stuff?

> downloads.asterisk.org is an https site, so certificate auth and all > that should be verifiable. Currently, Asterisk retrieves its external stuff not via HTTPs but HTTP. One approach would be to change all links to HTTPs within the Asterisk source. However, that is problematic for example in

[asterisk-dev] Authenticated downloads of external stuff?

Asterisk downloads a lot of external stuff while configuring and installing - via HTTP - for example sound files, Digium modules, and the PJProject. These downloads are guarded by checksum/hashes which are - not stored within the Asterisk tarball but - retrieved from the same source as the

[asterisk-dev] Jira: Create issue: Required field: Component

In Jira, when I create a new issue, the fields Summary and Description are marked with a red asterisk. I guess, that indicates a required field. However, for an external contributor like me, the fields Component, Affects Version, Severity, and Issue Guidelines are required as well. This is

Re: [asterisk-dev] Which enviroments are supported, really?

> Testing FreeBSD poses other problems however. None of us really work > with BSD based distributions so it would take more time that we have > available to do any serious testing there. Can you give an example of those anticipated problems? Greater differences allow deeper learning. Especially,

Re: [asterisk-dev] Idle Timers and Keep-Alives

> You'd be OK with keeping an idle timeout it as long as you can turn > it off at runtime. Yes, I do not need it. I do not use keep-alive for SIP-over-TCP (or TLS), neither - within the TCP layer (via its own mechanism), nor - on the TCP layer (via CRLN). Even short SIP-Register must be used

Re: [asterisk-dev] Which enviroments are supported, really?

I am not about the bugs themselves. I am going to fix and/or report them. I fixed already several of them. And more are coming. However, I am interested which environments are supported. After a mayor release like Asterisk 15, I would have expected that someone does a cross-check before the

Re: [asterisk-dev] Webpage Downloads

> 2) Test Release: Is it possible to remove the part about test releases > when the final release is newer? Or link to that final release instead? > Some users might scroll all the way down and install the "latest" test > release

Re: [asterisk-dev] Report a regression?

> FreeBSD is not a platform that we really support. That means, this issue is not a regression, right? That leads me to another question: . >> How does an external contributor report a regression in general: >> A) do I

[asterisk-dev] Which enviroments are supported, really?

Recently, I was hit by a missing dependency of an external library (ASTERISK-27475). Because I was not able to resolve the issue otherwise, I re-visited the first-time experience of Asterisk, thinking that should solve my issue for sure. wget

Re: [asterisk-dev] Help understand TCP/TLS server error

In Wireshark, do you see who is closing the connection? Which connection is closed first: SSL or TCP? In Asterisk CLI, do you see a SIP message at that time when "sip set debug on" was enabled? Is it a SIP-REGISTER? -- _ --

[asterisk-dev] Report a regression?

Because of commit c2850bf (ASTERISK-26563, Gerrit-7673), I am not able to compile Asterisk on the platform FreeBSD anymore. For example in FreeBSD 11.1, the compiler clang fails because of freeswap = (totalswap - usedswap). How does an external contributor like me report a regression in

Re: [asterisk-dev] Help understand TCP/TLS server error

> It doesn’t appear to have a negative effect on the ongoing call; The SIP channel closed. RTP continues to flow. Therefore, the call itself stays up but you cannot send any SIP messages anymore. In Asterisk, the channel driver chan_sip and its SIP-over-TLS is used by many folks out there,

Re: [asterisk-dev] Idle Timers and Keep-Alives

> Do we even WANT an idle timer? I posted my concerns already in : I have a device which crashes when it receives such a keepalive. I could live with a timer when Asterisk is not the registrar but registered somewhere else. But I do not _need_ that either.

Re: [asterisk-dev] Where is pjsip_evsub_set_uas_timeout?

> Asterisk never had a commit that actually used the function so > it does not matter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To

Re: [asterisk-dev] Where is pjsip_evsub_set_uas_timeout?

> 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 not used to leverage a version-controlled source code as pastebin

Re: [asterisk-dev] Webpage Downloads

> 2) Test Release: Is it possible to remove the part about test releases > when the final release is newer? Or link to that final release instead? > Some users might scroll all the way down and install the "latest" test > release

[asterisk-dev] Where is pjsip_evsub_set_uas_timeout?

Because I use not the bundled PJProject, I have to examine whether Configure finds all symbols with my external PJProject 2.7. For "pjsip_evsub_set_uas_timeout" it is not able to do so. Before I investigate why, I would like to know where this is used in Asterisk at all. I found neither that

[asterisk-dev] Gerrit: Draft » Cherry-Pick » Publish = no check

Issue 1: When I 1) git review --draft ... on the command line of my computer, 2) cherry pick that change to other branches via the Web interface, and 3) publish those changes via the Web interface, I have to reply with "recheck" on the initial review, the one I submitted as draft via the command

Re: [asterisk-dev] Adding a new module to Asterisk

> The new [module] relies on a library that I need to introduce to the > linker, however, I've tried to figure out how the autotools work in > there Took me a while to understand as well. Have a look at audio-codec modules like the one for Codec 2: . The

[asterisk-dev] Webpage Downloads

1) The button around "Download Latest - 15.0.0-rc1" is missing: Its A tag misses a class attribute with the value "btn btn-primary". 2) Test Release: Is it possible to remove the part about test releases when the final release

Re: [asterisk-dev] OPUS horrible quality with packet loss

> maybe version is too high [for that Native-PLC patch] You can answer that question yourself by adding debug output into codecs/codec_opus_open_source.c. For example, put one ast_log(…) for each "/* Case x". Case 5 and 6 are the perfect case without loss and therefore without activating

Re: [asterisk-dev] OPUS horrible quality with packet loss

With Opus Codec, speech frames interdepend. When one RTP packet is lost, several frames are corrupted. Furthermore, when the RTP packet is not lost but just late, Asterisk simply forwards it down to the transcoding modules. Consequently, a late packet corrupts even more frames. The Opus Codec

Re: [asterisk-dev] Adding an audio format in Asterisk

> adding a new line "CODEC_REGISTER_AND_CACHE" would magically do > everything required, but now I see there is some more work to do At you find several examples. If you do not need transcoding and just pass-through, the patch file for

Re: [asterisk-dev] Dynamic Payloads

> Is there any advantage to falling back to [the currently used] numbers as > defaults for a given format? If the remote party cares about a dynamic RTP payload type number but ignores its "=rtpmap:", this is a software bug for sure [1][2][3]. Consequently, you are asking whether SIP/SDP

[asterisk-dev] Adding Audio/Video/Text Format to IAX2: Formal Procedure?

Thanks to a user of one of my audio transcoding modules, I noticed that my modules work in chan_sip and res_pjsip but not in the channel driver for IAX2 of Asterisk 13. He changed: - include/asterisk/format_compatibility.h - channels/iax2/format_compatibility.c - main/format_compatibility.c

[asterisk-dev] chan_sip: auto_force_rport, auto_comedia: IPv6 and Firewalls?

In Asterisk 11, two auto_* values were added to the configuration parameter "nat". Those are about IPv4. Recently, I added IPv6 to my Asterisk. Because IPv6 does not have any NAT (normally), those auto_* disable both "force_rport" and "comedia". Therefore for IPv6 traffic,

[asterisk-dev] codecs.conf code for Opus and SILK?

Since Asterisk 13.12, Digium offers those audio codecs thanks to commercial but free transcoding modules. Those are closed source. Is it possible to get (just) the code related to loading the configuration file codecs.conf? You know, I have open-source modules of those audio-codecs available on

Re: [asterisk-dev] Asterisk 14.0.1 Now Available

>Has anybody updated the version of the [Opus transcoding] patch for 14 and/or >master? The story continues there. I updated the code and I try my best to maintain that fork (because I use it myself). Currently, it should be compatible from 13.7 up to

Re: [asterisk-dev] Asterisk 14.0.1 Now Available

>What features [of Opus-Codec] are supported? >* Forward Error Correction (FEC) >* Packet Loss Concealment (PLC) As of today, Asterisk itself does not allow *Native* PLC, see . No *Dynamic* FEC either, because a codec module has no

Re: [asterisk-dev] [BOUNTY] offered : Allow 256 bit SRTP cipher suites

>libSRTP supports the stronger cipher suites already, so I *believe* only >Asterisk source code needs minor changes. Desired cipher suites: > AES_CM_256_HMAC_SHA1_32 > AES_CM_256_HMAC_SHA1_80 It got a bit more complex because the code in Asterisk expected a fixed key length of 30 bytes.

[asterisk-dev] BuildSystem: make --jobs without amount of thread hangs

On Ubuntu 16.04 LTS, when I build Asterisk 13.9.1 via wget downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz tar zxf asterisk-1*.tar.gz sudo apt install libssl-dev libncurses-dev libnewt-dev libxml2-dev libsqlite3-dev uuid-dev libjansson-dev libedit-dev

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

> If I get a "codec2" stream, which rate (and/or other parameters) are used? Faced the same question, when I started with Codec 2. I am glad, somebody is interested. I am going to add this to the Read Me in my GitHub repository: Currently, FreeSWITCH and CSipSimple support only the first Codec 2

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

> If you need testing, I can support you. Thanks for the offer! Just give them a try and report all issues via their GitHub repository or privately, as you like. By the way just to avoid a misunderstanding: All five modules are finished and passed several tests of my own. They all support

[asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

Thanks to the codec/format changes which were introduced with Asterisk 13, adding new trancoding modules is possible within one working day. Thanks to format-attribute modules, the debugging of the SDP/fmtp-negotiation resides in one source file. Therefore, I was able to port five formats to

Re: [asterisk-dev] Purpose of Codecs with Multiple Sample Rates?

> I'm not sure if I'm willing to make such a substantial change with > changing codec name [for Speex]. That is OK. If you are interested, I extend the change for ASTERISK-25535 to cover slin and speex in ast_format_cap_append_by_type(.) as well. I do not need such an additional change for AMR

[asterisk-dev] Purpose of Codecs with Multiple Sample Rates?

I asked this question in : Since Asterisk 13, Signed-Linear and Speex share the same codec name for all sample rates, see main/codec_builtin.c:CODEC_REGISTER_AND_CACHE_NAMED(.). I added SILK and AMR(-WB) as formats to my Asterisk 13. They

Re: [asterisk-dev] Purpose of Codecs with Multiple Sample Rates?

>> ast_format_cache_get(.) does not work for Speex. ASTERISK-25535 shows this, for example with chan_sip: ast_format_cap_append_by_type(sip_tech.capabilities, AST_MEDIA_TYPE_AUDIO); ast_format_cap_append_by_type ast_format_cache_get(ast_codec->name) Here, the source code deals with

[asterisk-dev] Interactive Debugging of chan_pjsip with gdb?

Really like the trick to have an interactive session to debug my Asterisk installation: . Thanks for that Wiki entry! Saved me a lot of time already. Today, I face an issue in chan_pjsip

Re: [asterisk-dev] SIP/SDP: ptime in translation module?

> 2) ... codec modules are fed [with] frames already ... Uh. That is a bit too abstract, yet: When do I tag the frames exactly? ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance)) provides the expected packetization value in res/res_rtp_asterisk.c, yes. However, the transcoding

[asterisk-dev] SIP/SDP: ptime in translation module?

I am creating a translation module for AMR-WB. In one scenario on the SIP/SDP layer, a higher ptime was negotiated than the default one. For example, 60ms were negotiated instead of the AMR default 20ms. Now, Asterisk should send three frames per RTP packet. I try to play one of the recorded

Re: [asterisk-dev] Zuul, Bamboo, and Co. Failed?

I noticed Zuul gave me a fail on verification on Gerrit. However this time, I do not understand the above console output. Asterisk Team, please, look into job 635 and tell me what I did wrong, and where I find a human-readable message next time. Please, disregard that question. Fixed. :)

[asterisk-dev] Zuul, Bamboo, and Co. Failed?

https://jenkins.asterisk.org/jenkins/job/check-asterisk/635/ git remote set-url origin https://gerrit.asterisk.org/asterisk fatal: Not a git repository (or any of the parent directories): .git Build step 'Execute shell' marked build as failure Recording test results Publisher

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

Here, the Opus Format Attribute does not proxy fmtp-parameters either. For https://issues.asterisk.org/jira/browse/ASTERISK-25160, I added an additional feature to proxy (at least) maxplaybackrate (patch A). I have not gone through review process with that patch, because I still (need a) a more

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

If I receive an INVITE with fmtp from a peer, it won't be used to build the INVITE to the egress right? With Asterisk 13/chan_sip, it is possible to copy over the fmtp - even 1:1 - I do this here with AMR-WB. I created a res/res_format_attr_ and adopted

[asterisk-dev] 13: RTP pass-through creates no-media situations

No, I do not want to start another thread about codec-negotiation in chan_sip. However, while playing with the audio-codec Opus in Asterisk 12 and Asterisk 13, I was able to establish a call without audio: sip.conf directmedia=no disallow=all allow=opus,ulaw leg 1: VoIP/SIP client is calling

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

B) Change my patch not to use a char* but char[128]. Your easiest option with less chance of regression elsewhere would be this. Yes. Anyway: Is the Asterisk team interested in a patch at least for the 5 affected files? These are 54 changes. I am not sure if the path via an issue (Jira) and a

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

In a patch of mine (DANE for Asterisk 13/chan_sip; available on request), a char* was added in the struct ast_sockaddr to store the DNSSEC failure reason (why_bogus). Not to create any memory leaks, this pointer has to be freed. For this, the pointer must be initialized to NULL, for example via

Re: [asterisk-dev] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

, no if/def/version is introduced in this patch. Thanks, Alexander Traud -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-dev] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

://reviewboard.asterisk.org/r/4441/#review15180 --- On April 13, 2015, 1:28 p.m., Alexander Traud wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

in this patch. Thanks, Alexander Traud -- _ -- 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] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4441/#review14529 --- On March 30, 2015, 8:34 a.m., Alexander Traud wrote

Re: [asterisk-dev] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

in this patch. Thanks, Alexander Traud -- _ -- 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

[asterisk-dev] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

is sufficient. Because no new symbol of OpenSSL was used, I do not see a reason why this patch should not be compatible with older OpenSSL releases. Therefore, no if/def/version is introduced in this patch. Thanks, Alexander Traud

Re: [asterisk-dev] [Code Review] 3653: chan_sip: (Optionally) poll even on first part of TLS message

committed to tcptls.c just recently (revision 415907). Anyway, let us fix this bug as well. Diffs - trunk/channels/chan_sip.c 416319 Diff: https://reviewboard.asterisk.org/r/3653/diff/ Testing --- Asterisk 12.3 Thanks, Alexander Traud

Re: [asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

if for the EINTR case. However, I am not sure if the coding-style guides have a rule for this. @Richard I am just curious after reading the code: When is EINTR possible? Or is that just a coding convention? - Alexander Traud On Aug. 6, 2014, 2:21 a.m., ebroad wrote

Re: [asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

On Aug. 6, 2014, 6:58 a.m., Alexander Traud wrote: trunk/channels/chan_sip.c, line 3042 https://reviewboard.asterisk.org/r/3882/diff/2/?file=66262#file66262line3042 Because the rest of the Asterisk code does it this way: instead of bitwiseOr | please, a logicalOr

Re: [asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3882/#review13035 --- Ship it! Ship It! - Alexander Traud On Aug. 6, 2014, 6:04

Re: [asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

On Aug. 6, 2014, 6:58 a.m., Alexander Traud wrote: trunk/channels/chan_sip.c, line 3042 https://reviewboard.asterisk.org/r/3882/diff/2/?file=66262#file66262line3042 Because the rest of the Asterisk code does it this way: instead of bitwiseOr | please, a logicalOr

Re: [asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

On Aug. 6, 2014, 6:56 p.m., Alexander Traud wrote: Ship It! Not sure, if I am allowed to set that flag. Anyway, because I was invited, I think I have to. Everything fine from my side. (One part of the) community says thanks you for your contribution! - Alexander

Re: [asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

over UMTS (SIP INVITE = two TCP messages, 1.3 seconds latency between those messages); an issue introduced by the new variable exclusive_input (revision 416071). - Alexander Traud On July 31, 2014, 6:14 p.m., ebroad wrote

Re: [asterisk-dev] [Code Review] 3653: chan_sip: (Optionally) poll even on first part of TLS message

On June 27, 2014, 2:42 p.m., Matt Jordan wrote: Alexander Traud wrote: hence a 'retry once' poll may not be sufficient regardless to read all of the data from the socket. I am not sure, I understand you guys. Just to clarify my intentions: The proposed patch

Re: [asterisk-dev] [Code Review] 3709: configure.ac: Check OpenSSL for support of Elliptic Curve cryptography

On July 4, 2014, 7:48 a.m., Alexander Traud wrote: As I am the contributor of the original patch: Argg, I thought I double-checked all symbols. Anyway, mistakes happen. I am not sure why I wasn’t invited as reviewer for this one here. EC_KEY_new_by_curve_name() is the only culprit

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