Re: [asterisk-users] Patch to remove numbers from the logs
If you need something quick you could create a batch script with sed or awk
to remove the log lines you want and attach it to the prerotate script of
logrotate (in case you use any of these in your env). Certainly this is not
a final solution but it is already something that doesn't depend on an
asterisk patch.
On Sat, 10 Jul 2021 at 01:13, Dovid Bender wrote:
>
>
> On Fri, Jul 9, 2021 at 10:49 AM George Joseph wrote:
>
>>
>>
>> On Thu, Jul 8, 2021 at 3:58 PM Dovid Bender wrote:
>>
>>> Hi,
>>>
>>> We have a project where people will be making payments over the phone. I
>>> would like block Asterisk from logging any time the system is processing a
>>> card. So be it SayDigits(123456789), when the user enters DTMF or when I
>>> pass a card number as a variable to an AGI etc. I assume this affects
>>> others and I would like to have the patch created in a way that a. will be
>>> accepted by Sangoma and b. will work for anyone else that has this issue.
>>>
>>
>> Are you talking strictly about normal messages generated by the dialplan
>> or all messages, even warnings, errors, etc generated internally?
>>
>> Mainly verbose messages (so no warnings, errors etc.)
>
>>
>>
>>> My idea was to have a channel variable
>>> for exampleSet(CHANNEL(LOG_DIGITS)=OFF) and then have ast_logger check to
>>> see if the variable is set.
>>>
>>
>> Would you need to do this on a channel-by-channel basis or could you set
>> a global variable?
>>
> Because we would want to enable and disable it. Meaning we want all the
> logging on in general however at certain times where there is sensitive
> information we would not want it logged
>
>>
>>
>>> The problem I faced that wherever the logger is called a string is
>>> passed. So any digits (e.g. channel ID, thread ID etc.) would have the
>>> digits removed which I assume would hurt people. My solution was to have a
>>> configuration file where you would put in regex strings that we would
>>> replace. For instance if I set LOG_DIGITS=OFF and in the Dialplan I had
>>> CARD=4111
>>> EXP_MOTH=12
>>> EXP_YEAR=2025
>>>
>>
>> Are those variables set on the channel?
>>
> Yes they are. For instance Exten => _XX,1,Set(Month=${EXTEN})
>
>>
>>
>>>
>>> In the configuration file I would have
>>> CARD=([0-9]{15,16})
>>> EXP_MONTH=([1-2]?[0-9])
>>> EXP_YEAR=(202[5-9])
>>>
>>
>> I'd skip the config file and make teh regexes global dialplan variables.
>>
> But then how would you have multiple? Have the strings comma delimited or
> one large string (e.g.
> CARD=([0-9]{15,16})|EXP_MONTH=([1-2]?[0-9])|EXP_YEAR=(202[5-9])). One
> possible is the limit of a line in Asterisk if I recall correctly, was 256
> characters.
>
>>
>>
>>> The system would then look for any of the above expressions and then
>>> replace the numbers with an X. Does that seem like a patch that would get
>>> accepted? is that completely in left field? Any thoughts on a better way of
>>> doing it? I know I can change the verbosity to 0 but then I would get
>>> nothing at all in the logs.
>>>
>>> TIA.
>>>
>>> Dovid
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Asterisk and CentOS 8
In case anyone out there is working with CentOS, you might reconsider that decision: https://blog.centos.org/2020/12/future-is-centos-stream/ On Mon, 11 May 2020 at 23:53, George Joseph wrote: > > > On Sun, May 3, 2020 at 6:07 PM Patrick Wakano wrote: > >> Hello George, >> Hope this finds you well! >> I wonder if there has been any progress on this matter? >> > > Sorry, I missed this last week. We have an internal issue open to > revisit CentOS 8 and it's at the top of our backlog right now so we should > have something in the next few weeks. > > Thanks for sharing your experience! > > >> >> I've been working to have Asterisk running on CentOS 8 and our jump from >> CentOS 6 to 8 doesn't look too bad The missing packages found are: >> gmime-devel, iksemel-devel, corosynclib-devel, libresample-devel, hoard and >> python-devel. Python-devel could be replaced by python2-devel or >> python3-devel (python36-devel), but I am not sure if there is any python >> incompatibility... gmime-devel, iksemel-devel and hoard seems of no use >> (and they were already missing for CentOS 6 too), so only corosynclib (if >> you are using it) and libresample may be a problem Does anyone knows >> what is the use of libresample-devel package? >> >> Just to share the experience so far, to build Asterisk it is needed to >> install the epel-release and enable PowerTools. By ignoring the missing >> packages, the compilation works and we can start Asterisk. My only concern >> was about the libresample-devel because in the past I think its absence >> caused issues when loading/starting pjsip modules, but it didn't happen >> this time (but I am not using pjsip, I just loaded it with the sample >> config files) >> Following on and making Asterisk a systemd service instead of using >> init.d, I first checked the service file under contrib/systemd but it seems >> to serve a different purpose, so it could not be used. Instead a very >> simple one does the job. Just make sure your exec cmd does not have the >> '-c' option (used in safe_asterisk), because this was causing me a 100% CPU >> usage (related to this: >> https://unix.stackexchange.com/questions/191621/systemd-service-using-100-of-my-cpu-when-it-doesnt-if-i-start-it-without-syste/368037#368037 >> ) >> Be aware that SELinux may cause permissions and access problems when >> running as systemd service It took me a whole day to figure out that >> under our configuration (that have included files outside the /etc/asterisk >> folder) the "problem" was SELinux. >> Finally, configure firewalld to allow the SIP and RTP ports and you >> should be ready to go. >> Also, we are using rpmbuild to create an Asterisk package. rpmbuild (or >> new red hat policies) is more strict and now complains about the python >> shebangs that do not have a version (files contrib/script/ref*.py -> what >> is this used for??). When using the DONT_OPTMIZE flag the compiler warns >> for every single file saying, so it is quite annoying but does not to cause >> issues: >> /usr/include/features.h:381:4: warning: #warning _FORTIFY_SOURCE >> requires compiling with optimization (-O) [-Wcpp]. >> Anyway, these problems do not happen if you manually build with the >> simple configure and make commands. >> >> Cheers, >> Patrick Wakano >> >> On Fri, 18 Oct 2019 at 11:54, Carlos Chavez wrote: >> >>> They only problem I have found so far is while trying to install >>> Alembic for SQLAlchemy (for realtime configs). Those are the only packages >>> that I cannot get working properly. Vanilla Asterisk works fine with the >>> only extra package needed being libedit-devel that is not included in any >>> "official" repo. You need to download the Fedora Core 29 packages to in >>> order to successfully compile Asterisk. That being said, I would not >>> recommend trying to put this in production any time soon. >>> On 10/17/2019 11:19 AM, George Joseph wrote: >>> >>> At the current time, we do not recommend attempting to build Asterisk on >>> CentOS 8. Many packages Asterisk uses are not yet available and would >>> require building from their sources. The Asterisk packages are also not >>> available in the EPEL 8 or CentOS 8 repositories yet for the same reason. >>> >>> We'll update you when we think it's safe. >>> >>> >>> -- >>> *George Joseph* >>> Digium - A Sangoma Company | Software Developer | Software Engineer
Re: [asterisk-users] Exceptionally long queue length queuing
Hello list, Hope you are all doing well! Sorry for the long email but I tried to explain all I've seen regarding this issue I am going to open a ticket for this issue but I found it useful to already explain it here. I've also recently faced this "Exceptionally long queue length queuing" error in some servers running 16.8.0 and after A LOT of investigation, I've discovered what is causing it in my case (not sure if is the same case as initially reported by Dovid) So, what I discovered is that in case the diaplan has a Wait() and during this wait period many "deferrable frames" are received, the final piece of the Wait function (ast_safe_sleep_conditional) can throw the "Exceptionally long queue length queuing" error message. It is not really common to get a lot of "deferrable frames" (see ast_is_deferrable_frame) but we can have that by simply putting on hold and off hold while the channel is on "wait". My production case is a little more complex and happens while on Wait and with the AST_CONTROL_SRCCHANGE event. This happens when an old Asterisk 13.13.0 receives a hold REINVITE and starts a new RTP stream (with different SSRC) towards some other victim Asterisk with the Music on Hold while still sending the original caller RTP (so at this point 2 RTP streams with different SSRCs are sent to the destination that happens to be a channel currently on Wait). I could not find any ticket for such bug, but version 16.8.0 does not have this problem and instead seems to inject the Music on Hold in the existing RTP stream so there aren't 2 streams at the same time to the same destination. The problem with 2 streams is that each RTP package of the new stream generates a source change frame (AST_CONTROL_SRCCHANGE), and at 50pps this builds up a lot of events fast and so if the Wait is high the issue pops up (I agree having a channel on Wait for too long is not a good practice but noone is perfect...). Also if multiple channels are facing this situation then the problem just escalates very bad. When the "Exceptionally long queue length queuing" is happening all AMI commands seem to fail, BYE messages seem to be missed/delayed, internal timeouts expire and all sort of weird things happen to all channels in the server, apparently the whole Asterisk process gets locked in this loop affecting everything else... and this is what makes this issue really bad because Asterisk can become completely unresponsive while the error message is happening... if it only could throw the error and keep working fine I guess it would be reasonable leaving it up to the dialplan logic to avoid a long Wait() Anyway, to replicate the issue is enough a simple dialplan like this: exten => ,1,Answer() same => n,Wait(30) same => n,NoOp(After wait) same => n,Playback(goodbye) same => n,Hangup() After calling in and while in the Wait(30), repeatedly press Hold/Unhold in the telephone (with Linphone just press the hold once and then press and hold the spacebar which will repeatedly do the hold and unhold several times). After Wait is finished the "Exceptionally long queue length queuing" will show up but gets resolved very fast just because I think there weren't enough frames queued to really freeze Asterisk for too long. Thank you, Kind regards, Patrick Wakano On Fri, 3 Jul 2020 at 19:29, Joshua C. Colp wrote: > On Fri, Jul 3, 2020 at 3:32 AM Dovid Bender wrote: > >> >> >> On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp wrote: >> >>> On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender >>> wrote: >>> >>>> Hi, >>>> >>>> We have a box up and we are starting to see a lot of "Exceptionally >>>> long queue length queuing" in the logs. From all the research so far it >>>> seems like this leads to their systems crashing and being unreachable. In >>>> our case the box remains up and takes calls. We are running Asterisk >>>> 16.6.1. We are using MusicOnHold to play online music streams via ffmpeg. >>>> Any idea on how to troubleshoot this further to see why this is happening? >>>> >>> >>> The message occurs when either a channel is deadlocked or the processing >>> time to handle what a channel is doing is substantial, and things build up. >>> Getting a backtrace[1] at the time it occurs will show what the system is >>> actually doing and can provide insight into what is causing it. >>> >>> [1] >>> https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-Runningast_coredumperfordeadlocks,taskprocessorbackups,etc >>> . >>> >>> -- >>> >> I ended up re-writing my code to use less Local channels and most of the >> errors went awa
Re: [asterisk-users] Asterisk and CentOS 8
Hello George, Hope this finds you well! I wonder if there has been any progress on this matter? I've been working to have Asterisk running on CentOS 8 and our jump from CentOS 6 to 8 doesn't look too bad The missing packages found are: gmime-devel, iksemel-devel, corosynclib-devel, libresample-devel, hoard and python-devel. Python-devel could be replaced by python2-devel or python3-devel (python36-devel), but I am not sure if there is any python incompatibility... gmime-devel, iksemel-devel and hoard seems of no use (and they were already missing for CentOS 6 too), so only corosynclib (if you are using it) and libresample may be a problem Does anyone knows what is the use of libresample-devel package? Just to share the experience so far, to build Asterisk it is needed to install the epel-release and enable PowerTools. By ignoring the missing packages, the compilation works and we can start Asterisk. My only concern was about the libresample-devel because in the past I think its absence caused issues when loading/starting pjsip modules, but it didn't happen this time (but I am not using pjsip, I just loaded it with the sample config files) Following on and making Asterisk a systemd service instead of using init.d, I first checked the service file under contrib/systemd but it seems to serve a different purpose, so it could not be used. Instead a very simple one does the job. Just make sure your exec cmd does not have the '-c' option (used in safe_asterisk), because this was causing me a 100% CPU usage (related to this: https://unix.stackexchange.com/questions/191621/systemd-service-using-100-of-my-cpu-when-it-doesnt-if-i-start-it-without-syste/368037#368037 ) Be aware that SELinux may cause permissions and access problems when running as systemd service It took me a whole day to figure out that under our configuration (that have included files outside the /etc/asterisk folder) the "problem" was SELinux. Finally, configure firewalld to allow the SIP and RTP ports and you should be ready to go. Also, we are using rpmbuild to create an Asterisk package. rpmbuild (or new red hat policies) is more strict and now complains about the python shebangs that do not have a version (files contrib/script/ref*.py -> what is this used for??). When using the DONT_OPTMIZE flag the compiler warns for every single file saying, so it is quite annoying but does not to cause issues: /usr/include/features.h:381:4: warning: #warning _FORTIFY_SOURCE requires compiling with optimization (-O) [-Wcpp]. Anyway, these problems do not happen if you manually build with the simple configure and make commands. Cheers, Patrick Wakano On Fri, 18 Oct 2019 at 11:54, Carlos Chavez wrote: > They only problem I have found so far is while trying to install > Alembic for SQLAlchemy (for realtime configs). Those are the only packages > that I cannot get working properly. Vanilla Asterisk works fine with the > only extra package needed being libedit-devel that is not included in any > "official" repo. You need to download the Fedora Core 29 packages to in > order to successfully compile Asterisk. That being said, I would not > recommend trying to put this in production any time soon. > On 10/17/2019 11:19 AM, George Joseph wrote: > > At the current time, we do not recommend attempting to build Asterisk on > CentOS 8. Many packages Asterisk uses are not yet available and would > require building from their sources. The Asterisk packages are also not > available in the EPEL 8 or CentOS 8 repositories yet for the same reason. > > We'll update you when we think it's safe. > > > -- > *George Joseph* > Digium - A Sangoma Company | Software Developer | Software Engineering > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct/fax: +1 256 428 6012 > Check us out at: https://digium.com · https://sangoma.com > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option
That makes sense Kevin! Thanks for the explanation, I will create a ticket for this then! Kinds regards, Patrick Wakano On Wed, 26 Feb 2020 at 09:33, Kevin Harwell wrote: > On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote: > >> Hi Kevin! >> Thanks very much for your reply! Much appreciated! >> > > You're welcome! > > >> So I just have a remaining question from this, if the with-ssl is not >> mandatory to have the encryption support, what is it actually used for? >> > > In Asterisk is allows you to set a path to the openssl files. Typically > used if your install of openssl is in a non-default alternative location. > For example: > > --with-ssl=/path/to/ssl_files > > I'm not really sure how Asterisk accepts the parameter without a file path > specified. My guess is it either uses the default path, or potentially sets > some flag meaning ssl is required. > > >> Maybe it is some old flag which is not needed anymore and so can be >> ignored for now and possibly removed from the configure/makefile stuff for >> future releases? >> >> > See above. The flag is still needed in Asterisk. However, the setting > appears to propagate in some way into the bundled pjproject configuration. > This is in some way affecting the build of pjproject, then subsequently > causing res_pjsip in Asterisk to not load at runtime. Further investigation > is required as to why though. > > -- > Kevin Harwell > Software Developer > Sangoma Technologies > Check us out at: https://sangoma.com & https://asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option
Hi Kevin! Thanks very much for your reply! Much appreciated! So I just have a remaining question from this, if the with-ssl is not mandatory to have the encryption support, what is it actually used for? Maybe it is some old flag which is not needed anymore and so can be ignored for now and possibly removed from the configure/makefile stuff for future releases? Kind regards, Patrick Wakano On Wed, 26 Feb 2020 at 06:33, Kevin Harwell wrote: > On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano wrote: > >> Hello list, >> Hope you are all doing well! >> >> I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box >> and I wonder if someone can put some light on it. >> Log history short, install_prereq fails to install the packages (not sure >> how important they actually are): speexdsp-devel, gmime-devel, >> uriparser-devel, iksemel-devel, uw-imap-devel, hoard >> Then, I am running the following commands to build Asterisk: >> ./configure --with-crypto --with-srtp --with-ssl >> make menuselect.makeopts >> menuselect/menuselect --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES >> --enable MALLOC_DEBUG --disable BUILD_NATIVE --enable app_macro >> menuselect.makeopt >> make OPT=-fPIC >> make install >> make samples >> >> After this, when I start Asterisk, I get the following error with pjsip >> modules: >> ERROR[6253]: loader.c:2396 load_modules: Error loading module >> 'chan_pjsip.so': /usr/lib/asterisk/modules/chan_pjsip.so: undefined symbol: >> ast_sip_cli_traverse_objects >> ERROR[6253]: loader.c:2396 load_modules: Error loading module >> 'res_pjsip.so': /usr/lib/asterisk/modules/res_pjsip.so: undefined symbol: >> pjsip_tls_transport_start2 >> ERROR[6253]: loader.c:2396 load_modules: Error loading module >> 'res_pjsip_config_wizard.so': >> /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol: >> ast_sip_get_sorcery >> >> After a lot of investigation I found this post ( >> https://asteriskfaqs.org/2018/09/25/asterisk-users/asterisk-1561-symbol-pjsip_tls_transport_start2-not-found.html) >> which pointed me to the with-ssl parameter. >> So if I run all previous commands, but remove the "--with-ssl" options >> from the configure, then I don't have the pjsip errors when I start >> Asterisk. >> >> My first question is, what will be the impact of removing the --with-ssl >> option? Will TLS and WSS still be possible? >> > > If Asterisk is able to autodetect, and find the appropriate encryption > libraries then yes TLS and WSS should be available. I do not use those > options when building and am able to still make secure calls. > > >> I didn't have time to test these things, however I did compared the >> configure and make outputs and the only difference are below, so looks like >> nothing extra gets compiled when the with-ssl is used... >> With --with-ssl: >> [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject >> --disable-speex-codec --disable-speex-aec --disable-bcg729 >> --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec >> --disable-g722-codec --disable-g7221-codec --disable-opencore-amr >> --disable-silk --disable-opus --disable-video --disable-v4l2 >> --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv >> --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc >> --without-external-pa --without-external-srtp --disable-resample >> --disable-g711-codec --enable-epoll >> checking for mandatory modules: PJPROJECT CRYPTO SRTP *OPENSSL*... ok >> Without --with-ssl: >> [pjproject] Configuring with --prefix=/opt/pjproject >> --disable-speex-codec --disable-speex-aec --disable-bcg729 >> --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec >> --disable-g722-codec --disable-g7221-codec --disable-opencore-amr >> --disable-silk --disable-opus --disable-video --disable-v4l2 >> --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv >> --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc >> --without-external-pa --without-external-srtp --disable-resample >> --disable-g711-codec --enable-epoll >> checking for mandatory modules: PJPROJECT CRYPTO SRTP... ok >> >> Also, why am I having the pjsip startup errors when the --with-ssl is >> used? I could not find a clear explanation for this problem and how to fix >> it >> > > There appears to be a bug here. I configured, built, and ran with the same > options mentioned (--with-ssl, etc...) and received similar pjsip module > load errors. Please
[asterisk-users] pjsip startup errors when using "with-ssl" configure option
Hello list, Hope you are all doing well! I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and I wonder if someone can put some light on it. Log history short, install_prereq fails to install the packages (not sure how important they actually are): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands to build Asterisk: ./configure --with-crypto --with-srtp --with-ssl make menuselect.makeopts menuselect/menuselect --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable MALLOC_DEBUG --disable BUILD_NATIVE --enable app_macro menuselect.makeopt make OPT=-fPIC make install make samples After this, when I start Asterisk, I get the following error with pjsip modules: ERROR[6253]: loader.c:2396 load_modules: Error loading module 'chan_pjsip.so': /usr/lib/asterisk/modules/chan_pjsip.so: undefined symbol: ast_sip_cli_traverse_objects ERROR[6253]: loader.c:2396 load_modules: Error loading module 'res_pjsip.so': /usr/lib/asterisk/modules/res_pjsip.so: undefined symbol: pjsip_tls_transport_start2 ERROR[6253]: loader.c:2396 load_modules: Error loading module 'res_pjsip_config_wizard.so': /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol: ast_sip_get_sorcery After a lot of investigation I found this post ( https://asteriskfaqs.org/2018/09/25/asterisk-users/asterisk-1561-symbol-pjsip_tls_transport_start2-not-found.html) which pointed me to the with-ssl parameter. So if I run all previous commands, but remove the "--with-ssl" options from the configure, then I don't have the pjsip errors when I start Asterisk. My first question is, what will be the impact of removing the --with-ssl option? Will TLS and WSS still be possible? I didn't have time to test these things, however I did compared the configure and make outputs and the only difference are below, so looks like nothing extra gets compiled when the with-ssl is used... With --with-ssl: [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject --disable-speex-codec --disable-speex-aec --disable-bcg729 --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec --disable-g722-codec --disable-g7221-codec --disable-opencore-amr --disable-silk --disable-opus --disable-video --disable-v4l2 --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc --without-external-pa --without-external-srtp --disable-resample --disable-g711-codec --enable-epoll checking for mandatory modules: PJPROJECT CRYPTO SRTP *OPENSSL*... ok Without --with-ssl: [pjproject] Configuring with --prefix=/opt/pjproject --disable-speex-codec --disable-speex-aec --disable-bcg729 --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec --disable-g722-codec --disable-g7221-codec --disable-opencore-amr --disable-silk --disable-opus --disable-video --disable-v4l2 --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc --without-external-pa --without-external-srtp --disable-resample --disable-g711-codec --enable-epoll checking for mandatory modules: PJPROJECT CRYPTO SRTP... ok Also, why am I having the pjsip startup errors when the --with-ssl is used? I could not find a clear explanation for this problem and how to fix it Any idea is much appreciated! Thank you, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor
Hi Stefan, > Hi all > > I maintain the above - it was set up by an external party with whom relations > have now been severed by my employer. > > Quite early after the deployment it became evident that all .gsm audio files > produced on this virtual instance at Azure via MixMonitor are corrupt. [snip] Is the CentOS 7 installation/image the same across your bare-metal hosts and the one on azure? AFAIK there is still no official CentOS 7 image provided by the CentOS Project on the azure marketplace. Instead it's created by a third party [1]. So there may be differences that could cause issues. On your azure host, check the repo files in /etc/yum.repos.d/. If the mirrorlist/basurl points to openlogic or roguewave than it's a third-party image. IIRC Amazon and GCP have official CentOS 7 images provided by the CentOS Project. Maybe try one of those to see if the issue persists? Alternatively create your own CentOS 7 VM from the official CentOS 7 repositories using kickstart and try that on azure. Best, Patrick [1] https://azuremarketplace.microsoft.com/en-us/marketplace/apps/RogueWave.CentOS76?tab=Overview -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audit AMI actions commands
Hello list, Hope you all doing well! I've been after a way to log every AMI action received by Asterisk. Does anyone knows if Asterisk itself have this facility? I could not find anything related. Or does someone have any idea on how I could achieve that? My Asterisk has some applications connected via AMI and I want it logging all the actions (and the details of these actions) it was asked to do. Any idea is much appreciated! Thanks, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How best to run a SIPp test on a remote host
I've never worked with fabric nor pysipp, but if you want to run sipp in background so you can log out from your remote server and leave the test running, I suggest you can use screen. It works well for me. Patrick Wakano On Sat, 20 Oct. 2018, 01:55 Olivier, wrote: > Hello, > > I'm curently setting a lab environment for load testing an Asterisk > instance. > > This environment includes: > - a management workstation where I would like to run scripts and store > test reports > - a box hosting SIPp > - the Asterisk box I'm load testing (System Under Test) > - an other Asterisk box that simply anwers and echoes incoming media > > The SIPp commande I'm currently using looks like: > sipp -bg -nostdin -trace_screen -screen_file > /home/foobar/reports.d/campaign_123456789_1_1.txt -sf > /home/foobar/custom_with_10_minutes_audio.xml 3.4.49.3 -i 3.4.58.206 -s > 123456789 -r 1 -m 1 -d 6 > > Explainations: > - Run in background mode (more on that later) > - Last SIPp screen with successful/failed call counters saved in a file > - SIPp reads a 10 minutes audio file and stops reading after 6 ms (see > -d parameter) > - Media comes from SIPs, pass though System Under Test and then comes back > > 1. How can you run such "sipp -bg ..." command from management workstation > on remote SIPp box ? > I tried using Fabric (I'm discovering) but it always fails with a 99 exit > code, as if SIPp/Fabric required a different options set. > > 2. This project pysipp [1] seems interesting. > Has anyone tried it ? > How can you get started ? > > 3. Suggestions ? > > Best regards > > [1] https://github.com/SIPp/pysipp > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI timeout option
Thanks everyone for the answers!
I did explored some options at the PHP level and probably will do something
in this direction, but in fact what I was really looking was something in
the Asterisk side, not in the script side.
Because in my opinion regardless of the language or AGI type, Asterisk
itself should be able to timeout a long running script and return to the
dialplan. However looks like there is nothing of this sort.
Kind regards,
Patrick Wakano
On Sat, 15 Sep 2018 at 03:56, Eric Wieling wrote:
> I don't know AGIspeedy, but I have some PHP scripts where I set a
> connect timeout using streams.
>
> Example using https, but should be easily adaptable to non-s http.:
>
> $pbxsh_bin = @file_get_contents("https://blah.blah.blah";, FALSE,
> @stream_context_create(array('https' => array('timeout' => 5,
> "verify_peer"=>false, "verify_peer_name"=>false;
>
> On 09/14/2018 01:40 PM, Carlos Chavez wrote:
> > On 9/13/2018 8:04 PM, Patrick Wakano wrote:
> >
> >> Hello list,
> >> Hope you all doing well!
> >>
> >> Recently, I had an issue with a FastAGI PHP script, which under some
> >> specific situation would run into an infinity loop, consuming all CPU
> >> resources. This also was preventing Asterisk to terminated the call
> >> properly because it was waiting for the AGI to return... The
> >> application uses AGIspeedy to process the AGI calls, not sure if this
> >> can be affecting this situation somehow
> >> Due to this problem I started looking for some option to timeout the
> >> AGI call and return to the dialplan after XYZ seconds and so this
> >> would protect Asterisk preventing the dialplan to get stuck due to
> >> some external script problem that is actually outside of Asterisk
> >> control. Does Asterisk provide some control of this sort? I searched
> >> around and could not find any.
> >> Any idea is appreciated!
> >>
> >> Kind regards
> >> Patrick Wakano
> >>
> >
> > I think this is what you may be looking for:
> >
> > http://php.net/manual/en/function.set-time-limit.php
> >
>
> --
> http://help.nyigc.net/
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11! Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] AGI timeout option
Hello list, Hope you all doing well! Recently, I had an issue with a FastAGI PHP script, which under some specific situation would run into an infinity loop, consuming all CPU resources. This also was preventing Asterisk to terminated the call properly because it was waiting for the AGI to return... The application uses AGIspeedy to process the AGI calls, not sure if this can be affecting this situation somehow Due to this problem I started looking for some option to timeout the AGI call and return to the dialplan after XYZ seconds and so this would protect Asterisk preventing the dialplan to get stuck due to some external script problem that is actually outside of Asterisk control. Does Asterisk provide some control of this sort? I searched around and could not find any. Any idea is appreciated! Kind regards Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 400 reply to INVITE not properly treated
Hello list, Hope you all doing fine!! I am using chan_sip of Asterisk 13.6.0 and eventually I have a carrier which replies with 400 to some INVITES which happen to timeout I know the SIP reply code is not correct, but anyway I want to understand what is happening, because I think it is a bug In this situation, Asterisk is not failing the Dial, but sends back an ACK. Given the carrier has previously sent a 180, Asterisk just lets the user ringing forever Because the Dial() doesn't fail in this case, the user must hangup to terminate the call So question is, is this a known behaviour? I could not find anything related In my opinion, Asterisk should at fail the Dial and proceed with whatever was configured in the dialplan I tried some other 4XX SIP codes, but the only one I found not behaving properly is the 400 one Thanks, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to steal an answered call?
By the way, bear in mind this is exactly what a blind transfer from B to C would do, but with a lot of more work On Mon, 9 Jul. 2018, 16:36 Patrick Wakano, wrote: > Not sure how elegant this is, but I think you can try to elaborate some > logic that when phone C dials something, it would retrieve you the channel > phone A is connected and use the Bridge application to force the connection > of phone C to phone A. So you need first to save the channels you have > connected in a call, later on read this info based on whatever the phone C > has dialed in and just use this as parameter to the Bridge app. When this > happens channel C gets connected to channel A and channel B should > automatically get disconnected (or maybe continue your dialplan execution). > > On Mon, 9 Jul 2018 at 13:17, David Cunningham > wrote: > >> Hello, >> >> I'm familiar with Pickup/PickupChan for taking a ringing call, but does >> anyone know how a phone can "steal" an already answered call from another >> phone? Our users have decided that call parking is too long-winded and >> don't want to use that. >> >> For example: phone A calls phone B, phone B answers the call, phone C >> dials something to "steal" the call from B, and finally A and C are talking. >> >> Searching on voip-info.org shows a "BristuffSteal" command but it's very >> out of date (Asterisk 1.2). >> >> Thanks in advance for any suggestions. >> >> Kind regards, >> >> -- >> David Cunningham, Voisonics Limited >> http://voisonics.com/ >> USA: +1 213 221 1092 >> New Zealand: +64 (0)28 2558 3782 >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to steal an answered call?
Not sure how elegant this is, but I think you can try to elaborate some logic that when phone C dials something, it would retrieve you the channel phone A is connected and use the Bridge application to force the connection of phone C to phone A. So you need first to save the channels you have connected in a call, later on read this info based on whatever the phone C has dialed in and just use this as parameter to the Bridge app. When this happens channel C gets connected to channel A and channel B should automatically get disconnected (or maybe continue your dialplan execution). On Mon, 9 Jul 2018 at 13:17, David Cunningham wrote: > Hello, > > I'm familiar with Pickup/PickupChan for taking a ringing call, but does > anyone know how a phone can "steal" an already answered call from another > phone? Our users have decided that call parking is too long-winded and > don't want to use that. > > For example: phone A calls phone B, phone B answers the call, phone C > dials something to "steal" the call from B, and finally A and C are talking. > > Searching on voip-info.org shows a "BristuffSteal" command but it's very > out of date (Asterisk 1.2). > > Thanks in advance for any suggestions. > > Kind regards, > > -- > David Cunningham, Voisonics Limited > http://voisonics.com/ > USA: +1 213 221 1092 > New Zealand: +64 (0)28 2558 3782 > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor multiple times to the same file
Hello list, Hope you are all doing fine! While playing around with the MixMonitor, I've found out that it is possible to start the MixMonitor multiple times to the same output file. It is very easy to reproduce, via the CLI or via the dialplan. If the MixMonitor is called twice from the same channel, to the same output file and with option append. After that, with the "mixmonitor list" command we can see two running instances, and also from Linux open files "lsof" we can see two Asterisk process with the same file opened. Nevertheless, in this case the file does not gets corrupted and the recording is completely fine. However after observing such behavior I decided to call MixMonitor twice to the same file but from different channels. The observed behavior is that the output file has the contents of the first call until the second MixMonitor command is called for the second call. After that only the second call is recorded. If this is stopped, the first call is recorded again. I know we can consider an application/dialplan error to call MixMonitor twice for the same output file like this, but I am just curious to know if this situation is actually being handled by the Asterisk/MixMonitor code or if it is completely undefined behavior, since the behavior I described is actually consistent. Anyway, to avoid confusion and weird behavior, I think it would be probably better if Asterisk just not allow creating a MixMonitor thread to a file that is already opened by some other MixMonitor thread. Or is there any reason/situation in which this is not desired? Kind regards, Cheers, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy whisper
Thanks for the reply Joshua! I did look the code, but it's too complicated for my old C knowledge :( So I guess I am left with the Monitor app Also a ConfBridge would also work instead of ChanSpy with whisper Cheers, Patrick Wakano On 6 July 2018 at 08:51, Joshua Colp wrote: > On Thu, Jul 5, 2018, at 7:37 PM, Patrick Wakano wrote: > > Hello Asterisk list, > > Hope you are all doing well! > > > > We are using the MixMonitor application to record the calls and under > some > > situations the call can be spied using ChanSpy with whisper enabled. > > Sometimes the spying channel is a person who can interact in the call, > and > > some other times it is a sound file playing a message. The problem is > that > > for some reason the MixMonitor does not record whatever is injected via > > whisper in the call. It records only the call itself... I've done some > > research and apparently the Monitor application would be able to record > > everything (but I didn't verify it myself) Anyway why MixMonitor > can't? > > Also does anyone have an idea on how to record everything in the same > file? > > The implementation was never written to allow this and no feature added to > control it. Someone would need to go into the code, define what needs to > happen and how it can be controlled, and implement it (in regards to > MixMonitor and ChanSpy). > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor and ChanSpy whisper
Hello Asterisk list, Hope you are all doing well! We are using the MixMonitor application to record the calls and under some situations the call can be spied using ChanSpy with whisper enabled. Sometimes the spying channel is a person who can interact in the call, and some other times it is a sound file playing a message. The problem is that for some reason the MixMonitor does not record whatever is injected via whisper in the call. It records only the call itself... I've done some research and apparently the Monitor application would be able to record everything (but I didn't verify it myself) Anyway why MixMonitor can't? Also does anyone have an idea on how to record everything in the same file? Thanks, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor recording when in the holding bridge
Hello list, Hope you all doing fine! We are using the MixMonitor to record the call with the option 'b' so the recording is done only when this call is bridged. Under some situations the call is put on a wait state and the recording should not happen anymore until the channel is bridged back. This used to be done with a loop using the Wait application. Recently it was changed to use the BridgeWait, which I believe is much more suitable for this purpose. However by using the BridgeWait, the MixMonitor keeps recording the music on hold played to the channel during the entire wait time... with the Wait application this doesn't happen. I think this happens actually because the channel is bridged to the holding bridge, so the checks for "is bridged" passes. However, I think under a holding bridge the recording should be avoided too. Is this something that could be done? If not automatically maybe with some new flag Thank you, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tones in MixMonitor recording
I agree! I have my SBC and asterisk servers all configured with rfc2833, so it should be ok! No need for auto mode! Thanks again! Cheers Patrick On Tue, 1 May 2018, 20:07 Joshua Colp, wrote: > On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote: > > Thanks very much for the reply Joshua! > > So I guess that setting dtmfmode=auto would be the safest choice in order > > to strip out the DTMFs from the recording, right? > > Cheers! > > It should work. Personally I prefer explicit configuring instead of having > things just try to figure out what is in use. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tones in MixMonitor recording
Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > > > I have stumbled over some piece of dialplan code in which apparently they > > were trying to avoid recording the DTMF tones in the wav file. It is > really > > messy and I am not sure if this really works. So after a bit of research > I > > found this comment ( > > https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which > it is > > said: > > > > *"Asterisk strips the DTMF from the audio stream when configured for > > inband, so internal stuff can react to the DTMF and so the other side > does > > not hear the tone unless they are using inband (in which case it is > > regenerated)"* > > So my questions are, what are the cases in which Asterisk regenerates the > > DTMFs? Does it cause the recording to have the tone as well, or is it > only > > transmitted to the other leg without being generated to the recording > file? > > Also, what if one or both legs are RFC2833? From my tests the RFC2833 > > events never show up in the recording, but I just want to confirm that > this > > is always true. > > If properly configured then Asterisk will always strip and regenerate the > DTMF tone. You have to purposely misconfigure things to cause it to not get > stripped. IE: DTMF is actually inband but you configure it for RFC2833. > Since Asterisk wouldn't be listening to the audio stream, it would go right > through and get recorded. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF tones in MixMonitor recording
Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the DTMF from the audio stream when configured for inband, so internal stuff can react to the DTMF and so the other side does not hear the tone unless they are using inband (in which case it is regenerated)"* So my questions are, what are the cases in which Asterisk regenerates the DTMFs? Does it cause the recording to have the tone as well, or is it only transmitted to the other leg without being generated to the recording file? Also, what if one or both legs are RFC2833? From my tests the RFC2833 events never show up in the recording, but I just want to confirm that this is always true. Thanks, Kind regards, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alias for country in indications.conf
Just did Tzafrir suggestion and it worked like a charm! Thanks very much! Cheers, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> On 23 April 2018 at 23:45, Patrick Wakano wrote: > That's quite interesting Tzafrir! > I will give it a try! > Thank you very much for the idea! > Cheers, > Patrick Wakano > > > On 23 April 2018 at 23:42, Tzafrir Cohen wrote: > >> Also, >> >> On Mon, Apr 23, 2018 at 04:08:58PM +1000, Patrick Wakano wrote: >> > Hello list, >> > Hope you all doing fine! >> > I've tried to use the 'alias' directive in the indications.conf file but >> > apparently it doesn't work >> > It looks like maybe this feature was removed, because old sample for the >> > indications.conf file have example using the alias parameter, but newer >> > samples don't have it anymore also I couldn't find any ticket saying >> > this parameter was deprecated Anyway when trying to use it, it >> doesn't >> > work. Anyone aware of some change related to this? >> > I am using Asterisk 13.6.0 and have this in indications.conf: >> > [uk] >> > *alias = gb* >> >> Given that aliases don't work, you can alternatively use: >> >> ;;; >> [uk] >> description= >> ringcadence= >> ... >> >> [gb](uk) >> ;;; >> >> -- >>Tzafrir Cohen >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >> http://www.xorcom.com >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alias for country in indications.conf
That's quite interesting Tzafrir! I will give it a try! Thank you very much for the idea! Cheers, Patrick Wakano On 23 April 2018 at 23:42, Tzafrir Cohen wrote: > Also, > > On Mon, Apr 23, 2018 at 04:08:58PM +1000, Patrick Wakano wrote: > > Hello list, > > Hope you all doing fine! > > I've tried to use the 'alias' directive in the indications.conf file but > > apparently it doesn't work > > It looks like maybe this feature was removed, because old sample for the > > indications.conf file have example using the alias parameter, but newer > > samples don't have it anymore also I couldn't find any ticket saying > > this parameter was deprecated Anyway when trying to use it, it > doesn't > > work. Anyone aware of some change related to this? > > I am using Asterisk 13.6.0 and have this in indications.conf: > > [uk] > > *alias = gb* > > Given that aliases don't work, you can alternatively use: > > ;;; > [uk] > description= > ringcadence= > ... > > [gb](uk) > ;;; > > -- >Tzafrir Cohen > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alias for country in indications.conf
Hello Richard! Thanks very much for your answer! It all makes sense. I will just duplicated the tones config then! Thanks again! Cheers! Patrick Wakano On 23 April 2018 at 23:17, Richard Mudgett wrote: > It looks like any support for "alias" as a tone zone alias feature was > removed > back in 2009 (in git commit 4ec301360cdd84be911b06cd0adda2459d66bc6e) as > part of a code cleanup likely because alias was poorly conceived and didn't > work. What you have done by defining "alias" now is to define a tone event > called alias with a tone cadence of "gb". > > Richard > > On Mon, Apr 23, 2018 at 1:08 AM, Patrick Wakano wrote: > >> Hello list, >> Hope you all doing fine! >> I've tried to use the 'alias' directive in the indications.conf file but >> apparently it doesn't work >> It looks like maybe this feature was removed, because old sample for the >> indications.conf file have example using the alias parameter, but newer >> samples don't have it anymore also I couldn't find any ticket saying >> this parameter was deprecated Anyway when trying to use it, it doesn't >> work. Anyone aware of some change related to this? >> I am using Asterisk 13.6.0 and have this in indications.conf: >> [uk] >> *alias = gb* >> description = United Kingdom >> ringcadence = 400,200,400,2000 >> ; These are the official tones taken from BT SIN350. The actual tones >> ; used by BT include some volume differences so sound slightly different >> ; from Asterisk-generated ones. >> dial = 350+440 >> ; Special dial is the intermittent dial tone heard when, for example, >> ; you have a divert active on the line >> specialdial = 350+440/750,440/750 >> ; Busy is also called "Engaged" >> busy = 400/375,0/375 >> ; "Congestion" is the Beep-bip engaged tone >> congestion = 400/400,0/350,400/225,0/525 >> ; "Special Congestion" is not used by BT very often if at all >> specialcongestion = 400/200,1004/300 >> unobtainable = 400 >> ring = 400+450/400,0/200,400+450/400,0/2000 >> callwaiting = 400/100,0/4000 >> ; BT seem to use "Special Call Waiting" rather than just "Call Waiting" >> tones >> specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000 >> ; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT >> claim it >> ; is and I've not used a payphone for years) >> creditexpired = 400/125,0/125 >> ; These two are used to confirm/reject service requests on exchanges that >> ; don't do voice announcements. >> confirm = 1400 >> switching = 400/200,0/400,400/2000,0/400 >> ; This is the three rising tones Doo-dah-dee "Special Information Tone", >> ; usually followed by the BT woman saying an appropriate message. >> info = 950/330,0/15,1400/330,0/15,1800/330,0/1000 >> ; Not listed in SIN350 >> record = 1400/500,0/6 >> stutter = 350+440/750,440/750 >> >> The CLI shows Asterisk definitely understood the 'alias' parameter: >> CLI> indication show uk >> Country Indication PlayList >> = >> uk 400,200,400,2000 >> *uk alias gb* >> uk dial350+440 >> uk specialdial 350+440/750,440/750 >> uk busy400/375,0/375 >> uk congestion 400/400,0/350,400/225,0/525 >> uk specialcongesti 400/200,1004/300 >> uk unobtainable400 >> uk ring400+450/400,0/200,400+450/400,0/2000 >> uk callwaiting 400/100,0/4000 >> uk specialcallwait 400/250,0/250,400/250,0/250,400/250,0/5000 >> uk creditexpired 400/125,0/125 >> uk confirm 1400 >> uk switching 400/200,0/400,400/2000,0/400 >> uk info950/330,0/15,1400/330,0/15,1800/330,0/1000 >> uk record 1400/500,0/6 >> uk stutter 350+440/750,440/750 >> >> But the dialplan execution causes this: >> ERROR[20778][C-0006]: func_channel.c:661 func_channel_write_real: >> Unknown country code '*gb*' for tonezone. Check indications.conf for >> available country codes. >> >> Any info is much appreciated! >> Cheers, >> Patrick Wakano >> >> >> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> >> Virus-free. >> www.avg.com >> <http://www.avg.com/email-signature?utm_medium=email&utm_source=
[asterisk-users] Alias for country in indications.conf
Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore also I couldn't find any ticket saying this parameter was deprecated Anyway when trying to use it, it doesn't work. Anyone aware of some change related to this? I am using Asterisk 13.6.0 and have this in indications.conf: [uk] *alias = gb* description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 ; Special dial is the intermittent dial tone heard when, for example, ; you have a divert active on the line specialdial = 350+440/750,440/750 ; Busy is also called "Engaged" busy = 400/375,0/375 ; "Congestion" is the Beep-bip engaged tone congestion = 400/400,0/350,400/225,0/525 ; "Special Congestion" is not used by BT very often if at all specialcongestion = 400/200,1004/300 unobtainable = 400 ring = 400+450/400,0/200,400+450/400,0/2000 callwaiting = 400/100,0/4000 ; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000 ; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it ; is and I've not used a payphone for years) creditexpired = 400/125,0/125 ; These two are used to confirm/reject service requests on exchanges that ; don't do voice announcements. confirm = 1400 switching = 400/200,0/400,400/2000,0/400 ; This is the three rising tones Doo-dah-dee "Special Information Tone", ; usually followed by the BT woman saying an appropriate message. info = 950/330,0/15,1400/330,0/15,1800/330,0/1000 ; Not listed in SIN350 record = 1400/500,0/6 stutter = 350+440/750,440/750 The CLI shows Asterisk definitely understood the 'alias' parameter: CLI> indication show uk Country Indication PlayList = uk 400,200,400,2000 *uk alias gb* uk dial350+440 uk specialdial 350+440/750,440/750 uk busy400/375,0/375 uk congestion 400/400,0/350,400/225,0/525 uk specialcongesti 400/200,1004/300 uk unobtainable400 uk ring400+450/400,0/200,400+450/400,0/2000 uk callwaiting 400/100,0/4000 uk specialcallwait 400/250,0/250,400/250,0/250,400/250,0/5000 uk creditexpired 400/125,0/125 uk confirm 1400 uk switching 400/200,0/400,400/2000,0/400 uk info950/330,0/15,1400/330,0/15,1800/330,0/1000 uk record 1400/500,0/6 uk stutter 350+440/750,440/750 But the dialplan execution causes this: ERROR[20778][C-0006]: func_channel.c:661 func_channel_write_real: Unknown country code '*gb*' for tonezone. Check indications.conf for available country codes. Any info is much appreciated! Cheers, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE
That's really good info Tony! Thanks very much for the response! I will consider this to implement a better approach for the failed cases! Cheers, Patrick Wakano On 14 March 2018 at 20:44, Tony Mountifield wrote: > In article t...@mail.gmail.com>, > Patrick Wakano wrote: > > > > Thanks Dovid! > > Indeed looks a bug but regardless of this, this problem made me think > that > > the HANGUPCAUSE could be used for this purpose with benefits. > > I couldn't find an explanation about when DIALSTATUS would actually be > > better. > > The HANGUPCAUSE was reworked in version 11 ( > > https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't > find > > someone actually stating it is a better alternative or replacement to the > > DIALSTATUS or something similar. > > I think you should always check DIALSTATUS, as that will be set regardless > of > the way in which a dial fails. I believe HANGUPCAUSE is set to the Q.931 > code > received from PRI or SIP when a call is rejected or terminated. However, > there > could be other mechanisms for failure (such as failure to create a channel > within Asterisk, or an attempt to send to an unreachable peer), that may > set > DIALSTATUS without setting HANGUPCAUSE. > > So HANGUPCAUSE should be considered as extra detail, rather than a > replacement > or alternative to DIALSTATUS. > > Cheers > Tony > -- > Tony Mountifield > Work: t...@softins.co.uk - http://www.softins.co.uk > Play: t...@mountifield.org - http://tony.mountifield.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE
Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better alternative or replacement to the DIALSTATUS or something similar. Cheers, Patrick Wakano On 14 March 2018 at 13:30, Dovid Bender wrote: > I would think that is a bug since the only time DIALSTATUS = BUSY is where > you got a 486 or 600 (as per https://wiki.asterisk.org/ > wiki/display/AST/Hangup+Cause+Mappings). > > On Tue, Mar 13, 2018 at 10:11 PM, Patrick Wakano > wrote: > >> Hello list, >> Hope all doing well! >> >> I've been checking some cases when a Dial fails and dialplan execution >> continues to handle this. I am finding it a little confusing how we should >> handle the DIALSTATUS and the HANGUPCAUSE in this situation >> More specifically, I am facing a case in version 13.6.0 where I am >> getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP >> error. Seems wrong to me, since 480 should be converted to HANGUPCAUSE=19 >> and DIALSTATUS = NOANSWER (https://wiki.asterisk.org/wik >> i/display/AST/Hangup+Cause+Mappings). Anyway I am thinking about >> actually not checking the DIALSTATUS anymore and just rely on the >> HANGUPCAUSE, which seems more powerful. >> Looks like for a pure SIP environment the HANGUPCAUSE would have a more >> accurate information about the error. So question is can I always use this >> info and completely ignore what the DIALSTATUS is? >> Or does someone knows exactly where is more suitable to use one over the >> other? >> >> Thanks, >> Kind regards, >> Patrick Wakano >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIALSTATUS vs HANGUPCAUSE
Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems wrong to me, since 480 should be converted to HANGUPCAUSE=19 and DIALSTATUS = NOANSWER (https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings). Anyway I am thinking about actually not checking the DIALSTATUS anymore and just rely on the HANGUPCAUSE, which seems more powerful. Looks like for a pure SIP environment the HANGUPCAUSE would have a more accurate information about the error. So question is can I always use this info and completely ignore what the DIALSTATUS is? Or does someone knows exactly where is more suitable to use one over the other? Thanks, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crash on core show channel
Thanks for you answer Marcus, So maybe this means some bug was fixed? Anyone aware of something related? >From the release notes, I couldn't find any direct change that could fix this Thanks, Kind regards, Patrick Wakano On 21 February 2018 at 20:29, Marcus Kvarsell wrote: > Hello, i found upgrading to asterisk 15 helped. > > > > *Från:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *För *Patrick Wakano > *Skickat:* den 21 februari 2018 04:29 > *Till:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Ämne:* [asterisk-users] Asterisk crash on core show channel > > > > Hello Asterisk list, > > I am facing some Asterisk crashes which are consistently pointing to the > same backtrace, which is the following (using DONT_OPTIMIZE, > BETTER_BACKTRACES and MALLOC_DEBUG): > Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)): > #0 0x7f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6 > #1 0x004a91ca in cdr_object_get_by_name_cb () > #2 0x00463c60 in internal_ao2_traverse () > #3 0x00463fc3 in __ao2_callback () > #4 0x004a9b7d in cdr_object_get_by_name () > #5 0x004a9d80 in ast_cdr_serialize_variables () > #6 0x004de411 in handle_showchan () > #7 0x004e15a5 in ast_cli_command_full () > #8 0x004e175b in ast_cli_command_multiple_full () > #9 0x00455246 in netconsole () > #10 0x0060f192 in dummy_start () > #11 0x7f1f9cd3b9d1 in start_thread () from /lib64/libpthread.so.0 > #12 0x7f1f9be908fd in clone () from /lib64/libc.so.6 > > So this happens after issuing a "core show channel ". This > command is being executed by one script that periodically checks the > channels and some of its variables. Thing is that sometimes the channel has > something which causes this crash (happens around 2 times per month). Does > anyone have any idea of what may be happening/wrong? > I am running Asterisk version 13.6.0 on CentOS 6.6 kernel > 2.6.32-504.el6.x86_64 and on CentOS 6.9 kernel 2.6.32-696.el6.x86_64. > > Any idea is very appreciated! > > Best regards, > > Patrick Wakano > > > > > > > <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> > > Virus-free. www.avg.com > <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crash on core show channel
Hello Asterisk list, I am facing some Asterisk crashes which are consistently pointing to the same backtrace, which is the following (using DONT_OPTIMIZE, BETTER_BACKTRACES and MALLOC_DEBUG): Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)): #0 0x7f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6 #1 0x004a91ca in cdr_object_get_by_name_cb () #2 0x00463c60 in internal_ao2_traverse () #3 0x00463fc3 in __ao2_callback () #4 0x004a9b7d in cdr_object_get_by_name () #5 0x004a9d80 in ast_cdr_serialize_variables () #6 0x004de411 in handle_showchan () #7 0x004e15a5 in ast_cli_command_full () #8 0x004e175b in ast_cli_command_multiple_full () #9 0x00455246 in netconsole () #10 0x0060f192 in dummy_start () #11 0x7f1f9cd3b9d1 in start_thread () from /lib64/libpthread.so.0 #12 0x7f1f9be908fd in clone () from /lib64/libc.so.6 So this happens after issuing a "core show channel ". This command is being executed by one script that periodically checks the channels and some of its variables. Thing is that sometimes the channel has something which causes this crash (happens around 2 times per month). Does anyone have any idea of what may be happening/wrong? I am running Asterisk version 13.6.0 on CentOS 6.6 kernel 2.6.32-504.el6.x86_64 and on CentOS 6.9 kernel 2.6.32-696.el6.x86_64. Any idea is very appreciated! Best regards, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP + Stasis causing high memory consumption
Thank you very much for the response Matthew! I'll try the stasis configuration change first. I can't say if a malformed RTCP packet triggered the issue, but I did actually checked a tcpdump while problem was happening and the amount of RTCP packets and AMI events were reasonably normal Apparently it is an internal only problem, not actually outputting abnormal traffic in the network Best regards! Patrick Wakano On 16 November 2017 at 02:34, Matthew Jordan wrote: > > > On Mon, Nov 13, 2017 at 11:42 PM, Patrick Wakano > wrote: > >> Hello Asterisk list, >> >> I've facing a memory allocation issue that happens occasionally but on a >> consistent basis. >> The problem happens as follow, suddenly Asterisk starts consuming a lot >> of memory, in a rate of more than 1GB per hour. Kernel will eventually kill >> it via the OOM killer when memory is really exausted... This situation does >> not generate backtrace because Asterisk is responsive during the whole time >> and can be stopped/started. >> After compiling Asterisk with the MALLOC_DEBUG flag, when the problem >> happened we could see this output with the most consuming files, using >> "memory show summary" command: >> # cat memory_show_summary_201711101144.log | sort -nr | head >> 3060904798 bytes allocated (431496 in caches) in 16185877 allocations >> 1467180225 bytes in1647238 allocations in file stasis_channels.c >> 1217035109 bytes in 10229320 allocations in file json.c >> 240064732 bytes in1765287 allocations in file stasis_message.c >> 56402000 bytes in1762250 allocations in file taskprocessor.c >> 26125344 bytes in 203171 allocations in file rtp_engine.c >> 17308827 bytes in 307848 allocations in file stasis_cache.c >>9548128 bytes in 35482 allocations in file stasis_bridges.c >>3923172 bytes in 92169 allocations in file res_rtp_asterisk.c >>3099771 bytes in 29185 allocations in file strings.c >> >> Next minute it was already like this: >> # cat memory_show_summary_201711101145.log | sort -nr | head >> 3091009032 bytes allocated (375561 in caches) in 16345842 allocations >> 1482476997 bytes in1663899 allocations in file stasis_channels.c >> 1228978956 bytes in 10329516 allocations in file json.c >> 242483220 bytes in1783070 allocations in file stasis_message.c >> 57063120 bytes in1782910 allocations in file taskprocessor.c >> 26354360 bytes in 205090 allocations in file rtp_engine.c >> 17481272 bytes in 310935 allocations in file stasis_cache.c >>9628576 bytes in 35782 allocations in file stasis_bridges.c >>3828232 bytes in 93040 allocations in file res_rtp_asterisk.c >>3125118 bytes in 29427 allocations in file strings.c >> >> So as we can see, total allocations increased 159965 or 30MB in one >> minute.. >> A regular "memory show summary" during the day would be like this: >> # cat memory_show_summary_201711140400.log | sort -nr | head >> 30595537 bytes allocated (406915 in caches) in 77906 allocations >> 11021007 bytes in 5081 allocations in file stasis_channels.c >> 3825007 bytes in 12643 allocations in file cdr.c >> 1855344 bytes in 15837 allocations in file pbx.c >> 1766744 bytes in183 allocations in file translate.c >> 1467681 bytes in 1451 allocations in file chan_sip.c >> 140 bytes in280 allocations in file res_rtp_asterisk.c >> 980474 bytes in 4740 allocations in file xmldoc.c >> 895369 bytes in167 allocations in file file.c >> 800900 bytes in 2397 allocations in file channel.c >> 280728 bytes in613 allocations in file rtp_engine.c >> 243147 bytes in 2556 allocations in file stasis.c >> 130066 bytes in 1082 allocations in file stasis_cache.c >>24928 bytes in 82 allocations in file stasis_bridges.c >>51796 bytes in486 allocations in file stasis_message.c >> >> We also tried to get a "memory show allocations" when the problem was >> happening. Unsurprisingly it halted Asterisk and after we cancelled it, it >> had generated a 1GB file. Both memory logs hinted us to some problem in the >> RTP module, specifically in the RTCP handling. After investigating the >> code, looks like for some reason, the res_rtp_asterisk.c (line 4116) when >> doing the ast_rtcp_read() ended up entering in a loop, causing lots of >> allocations in the rtp_engine.c (line 2047 - ast_rtp_publish_rtcp_message). >> Also apparently, another loop (or maybe the same, not sure...) happene
[asterisk-users] RTCP + Stasis causing high memory consumption
Hello Asterisk list, I've facing a memory allocation issue that happens occasionally but on a consistent basis. The problem happens as follow, suddenly Asterisk starts consuming a lot of memory, in a rate of more than 1GB per hour. Kernel will eventually kill it via the OOM killer when memory is really exausted... This situation does not generate backtrace because Asterisk is responsive during the whole time and can be stopped/started. After compiling Asterisk with the MALLOC_DEBUG flag, when the problem happened we could see this output with the most consuming files, using "memory show summary" command: # cat memory_show_summary_201711101144.log | sort -nr | head 3060904798 bytes allocated (431496 in caches) in 16185877 allocations 1467180225 bytes in1647238 allocations in file stasis_channels.c 1217035109 bytes in 10229320 allocations in file json.c 240064732 bytes in1765287 allocations in file stasis_message.c 56402000 bytes in1762250 allocations in file taskprocessor.c 26125344 bytes in 203171 allocations in file rtp_engine.c 17308827 bytes in 307848 allocations in file stasis_cache.c 9548128 bytes in 35482 allocations in file stasis_bridges.c 3923172 bytes in 92169 allocations in file res_rtp_asterisk.c 3099771 bytes in 29185 allocations in file strings.c Next minute it was already like this: # cat memory_show_summary_201711101145.log | sort -nr | head 3091009032 bytes allocated (375561 in caches) in 16345842 allocations 1482476997 bytes in1663899 allocations in file stasis_channels.c 1228978956 bytes in 10329516 allocations in file json.c 242483220 bytes in1783070 allocations in file stasis_message.c 57063120 bytes in1782910 allocations in file taskprocessor.c 26354360 bytes in 205090 allocations in file rtp_engine.c 17481272 bytes in 310935 allocations in file stasis_cache.c 9628576 bytes in 35782 allocations in file stasis_bridges.c 3828232 bytes in 93040 allocations in file res_rtp_asterisk.c 3125118 bytes in 29427 allocations in file strings.c So as we can see, total allocations increased 159965 or 30MB in one minute.. A regular "memory show summary" during the day would be like this: # cat memory_show_summary_201711140400.log | sort -nr | head 30595537 bytes allocated (406915 in caches) in 77906 allocations 11021007 bytes in 5081 allocations in file stasis_channels.c 3825007 bytes in 12643 allocations in file cdr.c 1855344 bytes in 15837 allocations in file pbx.c 1766744 bytes in183 allocations in file translate.c 1467681 bytes in 1451 allocations in file chan_sip.c 140 bytes in280 allocations in file res_rtp_asterisk.c 980474 bytes in 4740 allocations in file xmldoc.c 895369 bytes in167 allocations in file file.c 800900 bytes in 2397 allocations in file channel.c 280728 bytes in613 allocations in file rtp_engine.c 243147 bytes in 2556 allocations in file stasis.c 130066 bytes in 1082 allocations in file stasis_cache.c 24928 bytes in 82 allocations in file stasis_bridges.c 51796 bytes in486 allocations in file stasis_message.c We also tried to get a "memory show allocations" when the problem was happening. Unsurprisingly it halted Asterisk and after we cancelled it, it had generated a 1GB file. Both memory logs hinted us to some problem in the RTP module, specifically in the RTCP handling. After investigating the code, looks like for some reason, the res_rtp_asterisk.c (line 4116) when doing the ast_rtcp_read() ended up entering in a loop, causing lots of allocations in the rtp_engine.c (line 2047 - ast_rtp_publish_rtcp_message). Also apparently, another loop (or maybe the same, not sure...) happened calling ast_channel_publish_blob() because we got lots of create_channel_blob_message and ast_channel_snapshot_create allocations too. Json, taskprocessor, stasis_message and stasis_cache allocations seems to be side effect of so many stasis messages being created and published. We could check memory logs 3 times for this problem and they are similar so it seems that a bug exists somewhere leading to this problem. So anyone has any idea of what could be happening or if it may be related to some known bug? We are running Asterisk 13.6.0 in CentOS 6.6. Thanks very much! Best regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting DoS attacks via SIP
Hi Mike, On 15-08-17 21:37, mdiehl wrote: Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [snip] I have to turn on sip debugging to find out who's hitting me. However, I can't just leave it on because it would kill my logging system. So, how are other people handling this? Is there an AMI event I want watch for? I watch for PeerStatus, but since there's no actual peer in the attack, I don't seem to get an event from AMI. Any ideas? You can block sipvicious/friendly scanner in iptables with something like: -A INPUT -p udp --dport 5060 -m string --string "friendly-scanner" --algo bm -j DROP You can also look at xtables with geoip to drop countries (per destination port) that should not connect to your Asterisk box. It's a big hammer but it works really well. Or put a proxy like Kamailio or OpenSIPS in front of the Asterisk box. That's what the telco's/service providers do. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Surrogate channels
Thanks very much for the explanation Richard!! Best Regards! Patrick <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Surrogate channels
Hello Asterisk list! I've been facing some scenarios in my dialplan where I see the "h" extension being executed for Surrogate channels. For me, it is kind of a mystery what these Surrogate channels are... I couldn't find good information about them... the source code is where I could find the most detailed info: /*! \brief Channel technology used to extract a channel from a running application. The * channel created with this technology will be immediately hung up - most external * applications won't ever want to see this. */ It mentions "*won't ever want to see this*", so I am unsure if they are only internal to Asterisk core and I should not be seeing these channels executing my dialplan, or if indeed I will see them in my dialplan, and then I have to handle them somehow (even if I don't want them) Anyway, does anyone know a good explanation about these surrogate channels, and what is expected of them when it comes to dialplan execution? Cheers! Patrick <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crash in ast_find_ourip
Hello list,
We've got an Asterisk crash in one of our servers and the core dump showed
following call tree.
Is this anyhow helpful to someone? Seems like a regular RTP / RTCP handling
that lead to a malloc crash
Grateful for any help!
Cheers,
Patrick
Thread 1 (Thread 0x7f8d6b023700 (LWP 14915)):
#0 0x7f8ead2d9252 in _int_malloc () from /lib64/libc.so.6
#1 0x7f8ead2da6b1 in malloc () from /lib64/libc.so.6
#2 0x7f8ead36f902 in make_request () from /lib64/libc.so.6
#3 0x7f8ead36fa5a in __check_pf () from /lib64/libc.so.6
#4 0x7f8ead332d17 in getaddrinfo () from /lib64/libc.so.6
#5 0x005435f5 in ast_sockaddr_resolve (addrs=0x7f8d6b01fee8,
str=0x7f8d6b01ffb0 "**", flags=768, family=0) at
netsock2.c:304
#6 0x0043484d in resolve_first (addr=0x7f8d6b022250,
name=0x7f8d6b01ffb0 "**", family=,
flag=768) at acl.c:792
#7 0x00434ce6 in ast_find_ourip (ourip=0x7f8d6b022250,
bindaddr=, family=0) at acl.c:970
#8 0x7f8e504f4d37 in ast_rtcp_read (instance=0x7f8e293df0f8) at
res_rtp_asterisk.c:4077
#9 0x7f8e504f5a45 in ast_rtp_read (instance=0x7f8e293df0f8,
rtcp=) at res_rtp_asterisk.c:4233
#10 0x7f8dfecdc7e1 in sip_rtp_read (ast=0x7f8e290f57c8) at
chan_sip.c:8298
#11 sip_read (ast=0x7f8e290f57c8) at chan_sip.c:8401
#12 0x004b44c5 in __ast_read (chan=0x7f8e290f57c8, dropaudio=0) at
channel.c:3874
#13 0x00476655 in bridge_handle_trip
(bridge_channel=0x7f8e28d669f8) at bridge_channel.c:2272
#14 bridge_channel_wait (bridge_channel=0x7f8e28d669f8) at
bridge_channel.c:2442
#15 0x00477658 in bridge_channel_internal_join
(bridge_channel=0x7f8e28d669f8) at bridge_channel.c:2587
#16 0x00468610 in bridge_channel_ind_thread (data=0x7f8e28d669f8)
at bridge.c:1690
#17 0x005bdbbb in dummy_start (data=) at
utils.c:1232
#18 0x7f8eae18d9d1 in start_thread () from /lib64/libpthread.so.0
#19 0x7f8ead3488fd in clone () from /lib64/libc.so.6
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0 0x7f8ead2d9252 in _int_malloc () from /lib64/libc.so.6
#0 0x7f8ead2d9252 in _int_malloc () from /lib64/libc.so.6
No symbol table info available.
#1 0x7f8ead2da6b1 in malloc () from /lib64/libc.so.6
No symbol table info available.
#2 0x7f8ead36f902 in make_request () from /lib64/libc.so.6
No symbol table info available.
#3 0x7f8ead36fa5a in __check_pf () from /lib64/libc.so.6
No symbol table info available.
#4 0x7f8ead332d17 in getaddrinfo () from /lib64/libc.so.6
No symbol table info available.
#5 0x005435f5 in ast_sockaddr_resolve (addrs=0x7f8d6b01fee8,
str=0x7f8d6b01ffb0 "**", flags=768, family=0) at
netsock2.c:304
hints = {ai_flags = 0, ai_family = 0, ai_socktype = 2, ai_protocol
= 0, ai_addrlen = 0, ai_addr = 0x0, ai_canonname = 0x0, ai_next = 0x0}
res =
ai =
s = 0x7f8d6b01fe00 "**"
host = 0x7f8d6b01fe00 "**"
port = 0x0
e =
i =
res_cnt =
__PRETTY_FUNCTION__ = "ast_sockaddr_resolve"
#6 0x0043484d in resolve_first (addr=0x7f8d6b022250,
name=0x7f8d6b01ffb0 "**", family=,
flag=768) at acl.c:792
addrs =
addrs_cnt =
#7 0x00434ce6 in ast_find_ourip (ourip=0x7f8d6b022250,
bindaddr=, family=0) at acl.c:970
ourhost = "**", '\000'
root = {ss = {ss_family = 1384, __ss_align = 56, __ss_padding =
"\320\302\000\000\000\000\000\000\060M\177)\000\000\000\000\n\000\000\000\000\000\000\000\002",
'\000' "\300, ", '\000' ,
"\n\000\000\000\062\000\000\000[\000\000\00
0|\000\000\000w\000\000\000n", '\000' ,
"8\000\000\000\000\000\000"}, len = 49872}
res = 1795293104
port = 0
__PRETTY_FUNCTION__ = "ast_find_ourip"
#8 0x7f8e504f4d37 in ast_rtcp_read (instance=0x7f8e293df0f8) at
res_rtp_asterisk.c:4077
i =
pt =
length =
rc =
message_blob =
rtcp_report = 0x7f8e2812c198
rtp = 0x7f8e285da140
addr = {ss = {ss_family = 2, __ss_align = 0, __ss_padding =
"0\000\000\000\060\000\000\000\220)\002k\215\177\000\000\320(\002k\215\177\000\000\340\323[\000\000\000\000\000H\360p(\216\177\000\000\021\260Q\000\000\000\000\000\245\006\000\000\216\177\000\000\356O0\255\216\177\000\000\020]\022)\216\177\000\000\000\000\000\000\000\000\000\000\245$\002k\215\177\000\000\000\000\000\000\216\177\000\000H\360p(\216\177\000\000\001\000\000\000\000\000\000"},
len = 16}
rtcpdata = '\000' , "\001", '\000' "\323,
[\000\000\000\000\000\201\310\000\fMm\216\347\334lf\256͑g\205Ms_\205\000\000\003\253\000\002J\340b{\370\363\000\000\000\000\000\000\225\310\000\000\000\017\000\000\000\000\000\000\000\000\201\312\000\aMm\216\347\001
Re: [asterisk-users] fail2ban Asterisk 13.13.1
On 02-03-17 13:52, Bryant Zimmerman wrote: John V Are you using pjsip? We are have several test servers and I just checked my /etc/fail2ban/filter.d/asterisk.conf and it is not updated for pjsip implementations. Looking at the security log files and the regex I noticed that some items are being banned but others are not due to changes in the messages for pjsip. Anyone got an updated asterisk.conf for fail2ban. The latest upstream version of asterisk.conf can be found here: https://github.com/fail2ban/fail2ban/blob/0.10/config/filter.d/asterisk.conf This commit mentions improved pjsip support: https://github.com/fail2ban/fail2ban/commit/f85fb45b29768f687546ba25f805977cf00b6e43 HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Execution of pre-bridge handlers
What an excellent response Richard!!! Thank you very much for that!! Best regards! Patrick On Wed, Feb 15, 2017 at 5:23 AM, Richard Mudgett wrote: > > > On Tue, Feb 14, 2017 at 6:24 AM, Patrick Wakano wrote: > >> Hello Asterisk Users, >> >> Hope you all doing fine! >> I am working with a quite complex dialplan, and I've come to some >> situations where it makes some nasty use of pre-bridge handlers. >> The pre-bridge handlers wiki (https://wiki.asterisk.org/wik >> i/display/AST/Pre-Bridge+Handlers) doesn't have the big warning the >> pre-dial one has indicating it must return and must not put the >> caller/callee in other applications (https://wiki.asterisk.org/wik >> i/display/AST/Pre-Dial+Handlers). So apparently, looks like they >> wouldn't have this restriction... However I had the feeling this was not >> true, so after some research I found this issue >> https://issues.asterisk.org/jira/browse/ASTERISK-25690, that says "*Connected >> line subroutines are meant** to be fast and as a result there is an >> expectation that applications invoked will not consume frames*". I am >> assuming that connected lines subroutines are just different words for >> pre-bridge handlers, right? >> Anyway my question is, what happens if I do not return straight away from >> the pre-bridge handler? Or even worst, if I execute a Dial application for >> example? Will I fall in some "undefined behaviour"? >> >> Anyone has experienced something like this? >> > > There are several handler routines available and each handles situations > for the > different states of a call. It makes no sense to use the Hangup() > application in > any of them and you must return from all of them. Most of the handlers > operate > from within the Dial application. > > Pre-dial handlers > The purpose of these routines is to setup a channel to place a call. > The pre-dial > routines can be run on the calling and called channels. See the Dial > application > documentation. > > For the calling channel, you can do most anything to the calling channel > except > hangup because you are still within the Dial application's control. The > reason > for the ability to execute a pre-dial routine on the calling channel > instead of doing > all the setup before executing Dial is to eliminate a window of > opportunity when using > the Lock/Unlock applications with Dial. > > For the called channel, you can only setup the channel. At this point, > the channel > exists but is not connected to anything nor has the call been placed. > Do your > channel setup and return. > > Redirecting interception handlers > This routine normally executes on the calling channel because the called > channel > has indicated that the call is being diverted/forwarded/redirected to > somewhere > else. The purpose of this routine is to get the REDIRECTING party > information > setup as you want and then return. You do not have control of the media > nor should > you hangup. You also should be quick about it. > > Pre-bridge handlers > At this point the called channel has answered and all other called > channels that were > dialed have been hung up. The called channel is about to be bridged > with the calling > channel. > > The purpose of this routine is to give the called person an opportunity > to decide if > he even wants to talk to the caller. You have control of the media > stream to the called > party. You cannot hangup the channel in the routine because you must > return. If you > want to abort bridging the call with the channel you must set a return > value as > documented by the Dial application. You need to remember that the > caller is > waiting to be connected the entire time you are in this routine. > > Connected-line interception handlers > At this point the channels are bridged together and may have been > talking for awhile. > > The purpose of this routine is to get the CONNECTEDLINE party > information setup > as you want and then return. The bridged peer has changed identity > likely because > of a transfer. You do not have control of the media nor should you > hangup. You also > need be quick about it or you risk causing a noticeable interruption to > the media. > > Hangup handlers > At this point the channel is hungup and you should be gathering > information about > the call for further processing later. You should not be doing > extensive post call > analysis at this time because you are delaying the channel technology > hangup > sequence. You hav
[asterisk-users] Execution of pre-bridge handlers
Hello Asterisk Users, Hope you all doing fine! I am working with a quite complex dialplan, and I've come to some situations where it makes some nasty use of pre-bridge handlers. The pre-bridge handlers wiki (https://wiki.asterisk.org/ wiki/display/AST/Pre-Bridge+Handlers) doesn't have the big warning the pre-dial one has indicating it must return and must not put the caller/callee in other applications (https://wiki.asterisk.org/ wiki/display/AST/Pre-Dial+Handlers). So apparently, looks like they wouldn't have this restriction... However I had the feeling this was not true, so after some research I found this issue https://issues.asterisk.org/ jira/browse/ASTERISK-25690, that says "*Connected line subroutines are meant** to be fast and as a result there is an expectation that applications invoked will not consume frames*". I am assuming that connected lines subroutines are just different words for pre-bridge handlers, right? Anyway my question is, what happens if I do not return straight away from the pre-bridge handler? Or even worst, if I execute a Dial application for example? Will I fall in some "undefined behaviour"? Anyone has experienced something like this? Many thanks, Cheers, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing PBX during a call in progress
Keepalived + heartbeatd allows you to maintain a a floating IP between two machines. If those two machines had configs, internal state synced, and the IP is configured to float automatically between the two based on which is actively up, would it be possible to not drop a call should the active host go down. On Thu, Jan 12, 2017 at 12:21 PM A J Stiles wrote: On Thursday 12 Jan 2017, Telium Technical Support wrote: > This was asked many years ago but I thought I would check to see if things > have changed. Is it possible to take over a call in progress - using a > replacement Asterisk server? > > In other words, if 2 user agents are connected through an Asterisk PBX, and > I tracked the call ID, IP of each UA (and anything else needed), could I > remove the PBX and put a new one in its place (at the same IP address) and > resume the call? Somehow keeping the call up on the UA's and telling > Asterisk to just resume a call given specified parameters (so the UA's > wouldn't notice the change)? I doubt there is any chance whatsoever of that working! For a start, you can't have two machines on the same subnet with the same IP address. It just does not work. And there is all manner of internal state that would have to be replicated onto the new server. When the clicky-clicky exchange in the village where I grew up was updated to System Y sometime in the early 1990s, all the phones went dead for about 30 minutes and calls in progress were cut off. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?
On 03-01-17 19:06, Joshua Colp wrote: On Fri, Dec 30, 2016, at 05:04 AM, Kevin Long wrote: Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I’m sure I’m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully, and place a SIP call, with no certificate warnings. But shortly after I place that first call and hang up, I receive a certificate name mismatch error in the softphone, the error presenting me with the *IP adddress* of my Asterisk server, not the hostname, and of course the TLS certificates only have the hostname, not the IP, and I have configured the soft phone to use the hostname, not the IP, to connect. I’m guessing there is some currently unset hostname setting within asterisk/pjsip that is defaulting to sending the IP in the sip messages, and then when the soft phone tries to make a new tls sip connection to asterisk, perhaps to signal to asterisk that the call is complete, it then connects to the IP instead of the hostname, and the mismatch occurs ? This might be the Contact header. Right now there is no ability to configure this to a hostname instead of an automatically determined IP address. Thank you for your feedback Joshua. Does "right now" mean that this will be fixed in the (near) future? Should I file a Jira ticket? Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting
Hi Travis, On 04-03-16 15:23, Ryan, Travis wrote: I start asterisk 13.7.2 and it dies before I can rasterisk into it. I’ve tried getting a coredump, but it doesn’t coredump. I know there are a lot of errors in the log below, but most of those just say it’ll not load a module, and no big deal. When launching from commandline (not service script) here is what happens. http://pastebin.com/3GFe6fG9 Two things: [Mar 3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module: Error loading module 'res_monitor.so': /usr/lib/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop [Mar 3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module: Error loading module 'res_ari_events.so': /usr/lib/asterisk/modules/res_ari_events.so: undefined symbol: stasis_app_register_all Undefined symbol errors are not good. Not sure why that's just a WARNING. Maybe something went wrong during the build? The output of the build should show you more information. In the mean time try disabling these two modules just to see if that clears up the problem. [Mar 3 15:19:39] == Parsing '/etc/asterisk/extensions.conf': Found [Mar 3 15:19:39] WARNING[8439]: config.c:2228 config_text_file_load: Unterminated comment detected beginning on line 386 That needs fixing. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr_odbc: Error in ExecDirect: -1
On 01/13/16 14:48, Vitor Mazuco wrote:
Hi everybody!
I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer
I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC
When I make a call the CLI returns for me
See the log:
== Using SIP RTP CoS mark 5
-- Executing [2021@ramais:1] Dial("SIP/2020-",
"SIP/2021,60,tT") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/2021
-- SIP/2021-0001 is ringing
> 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
[Jan 13 11:31:07] NOTICE[2279][C-]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
-- SIP/2021-0001 answered SIP/2020-
> 0x7fd3b4004eb0 -- Probation passed - setting RTP source
address to 192.168.25.100:8000
> 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
> cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] WARNING[2279][C-]: res_odbc.c:604
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
> cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] ERROR[2279][C-]: cdr_odbc.c:177 odbc_log:
CDR direct execute failed
See my res_odbc.conf
[asterisk]
enabled = yes
dsn = asterisk-connector
username = root
password = 100567
pooling = no
limit = 1
pre-connect = yes
What can be happened?
Thank in advanced.
Just a guess but try setting "pooling" to yes and "limit" to a higher value.
Best,
Patrick
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Re: [asterisk-users] Signaling ringing on other extension
On 12/30/15 12:24, Luca Bertoncello wrote: Ishfaq Malik schrieb: Do you have a link to the user guide for your exact phone model? Unfortunately not... I have a Thomson ST2022, but I can just find in Internet manual for the ST2030... The administrator manual can be found at: http://www.manualslib.com/manual/909341/Thomson-St2020.html?page=5 To download click the green Download button at the top. In the right column there is also a link to the User Guide. Cheers, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fritzbox 7490
On 08-06-15 19:00, Christian wrote: Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks! I use a 7360 and it works ok but if the 7490's firmware is anything like the 7360 then be prepared for some fixing before it works with Asterisk & SIP on port 5060: http://blog.laimbock.com/2014/03/27/how-to-make-asterisk-work-behind-fritz-box/ HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP2140
Hi Everyone. I have a customer looking to deploy about 20 Grandstream GXP2140 phones. Normally they would deploy Yealink brand phones but they want a phone with gigabit pass through and the Yealinks with gigabit are too expensive for their budget. Does anyone on the list have experience with the GXP2140? Is it a reliable phone? Does anyone have recommendations for other phones with gigabit pass through? Regards, Patrick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition
I have seen a similar problem occasionally. We will be doing maintenance on a customer's server and they will have one or two "ghost" channels on their machine hundreds of hours old but with no call associated with them. So far we have just been rebooting their server or issuing a hangup command to the channels. From: asterisk-users-boun...@lists.digium.com on behalf of Alex Villacís Lasso Sent: 08 April 2015 00:33 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition El 07/04/15 a las 17:38, Alex Villacís Lasso escribió: > I am trying to collect enough information about an problem a client is having > with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, > and we upgraded to 11.15.0 and then to 11.17.0 with no solution. > > Background: this client is a telemarketing call-center that generates > outgoing calls with close to a hundred agents operating simultaneously in > peak hours. The system uses asterisk with FreePBX 2.8. In order to generate > the calls, I wrote a program that > connects to Asterisk using the AMI protocol. This program expects the SIP > agent extensions to be assigned as members of queues, of which there are > about 20, as shown below: > > 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s > talktime), W:0, C:581, A:260, SL:82.6% within 60s >Members: > SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls > (last was 800 secs ago) > SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls > (last was 708 secs ago) > SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls > (last was 656 secs ago) > SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls > (last was 789 secs ago) >No Callers > > The program runs "queue show" through AMI every few seconds. For each queue > to be used in telemarketing, the program counts the number of members that > are "Not In Use". If at least one is found, it reads that many phone numbers > from the database and uses > the AMI Originate command on each one, as follows: > > Action: Originate > Channel: Local/NN@from-internal > Exten: > Context: from-internal > Priority: 1 > Async: true > ActionID: xxx > > Here, NN is the number read from the database and is the queue > extension in the FreePBX-created context that eventually runs the Queue() > dialplan application for the corresponding queue. This causes the call to be > connected between the > outgoing number and the queue, and is then assigned to a queue member by > Asterisk. The dialplan is configured to route NN through one of a > series of SIP trunks using the outbound routes as configured by FreePBX. > > The issue is that although this strategy works correctly on the user's > machine for a few days, we have been observing that eventually the > application stops placing calls. The agents are all idle (all 90 to 100 of > them), but the "queue show" command shows > them to be "In Use" on all queues. Furthermore, in normal operation, the > "core show channels" command shows at most one channel for each configured > SIP client in the "Up" state, but when calls stop being placed, the same > command reports multiple channels > in the "Up" state, as follows (after sorting): > > Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192 > Local/9759315789@from-internal-a456;1!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing > Line)!9759315789!!!3!500!(None)!1428426084.169326 > Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323 > SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224 > SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing > Line)!110!!!3!590!(None)!1428425994.169124 > SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760 > SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273 > SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384 > SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850 > SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113 > SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436 > SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47
Re: [asterisk-users] Call Quality Measuring
Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor in the near future. So can I assume from the lack of discussion nobody is using the “sip show channelstats” stuff? Regards, Patrick. On 31/03/2015 08:23, "Olivier" wrote: >Some SIP hardphones (Polycom) or softphones (Counterpath) embed a >module that metter MOS. > > >Regards > >2015-03-25 14:21 GMT+01:00 Patrick Beaumont >: >> Hi everyone. >> >> We regularly get customers complaining about call quality issues. Most >>of >> the time it turns out to be their own broadband. Very occasionally >>server >> load. Does anyone have any advice or links to advice on measuring call >> quality? >> >> I’ve been playing around with “sip show channelstats” but can’t other >>than >> measuring the packet loss I don’t really know what I’m supposed to be >> looking for in order to say “ah ha! that’s the problem!”. I also don’t >> know what it’s limits are. Will the stats in “sip show channelstats” >>show >> a customer using a torrent client and saturating their own broadband >> connection? >> >> Regards, >> Patrick. >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Quality Measuring
Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I’ve been playing around with “sip show channelstats” but can’t other than measuring the packet loss I don’t really know what I’m supposed to be looking for in order to say “ah ha! that’s the problem!”. I also don’t know what it’s limits are. Will the stats in “sip show channelstats” show a customer using a torrent client and saturating their own broadband connection? Regards, Patrick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam score:11%]
I encountered a bug in some Polycom models where it would refuse to register to a domain that started with a number (e.g 3something.voip.com). Could that be applicable here? Regards, Patrick. From: Jordan Cook - Gyron Networks mailto:jordan.c...@gyron.net>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Date: Friday, 23 January 2015 16:24 To: "asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>" mailto:asterisk-users@lists.digium.com>> Subject: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam score:11%] Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes – the phone does not re-try to register and if you try to dial out on the phone it says “URI Dialing is Disabled” Has anyone else had this issue? I'm running asterisk 11.7.0. This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam score:10%]
My suspicion would be that the line o=Z 0 0 IN IP4 146.115.163.234 is causing the problem. Your SIP client is reporting it's external IP address for the audio stream rather than it's internal one. I would look at the settings in your sip client to see if it has any automatic NAT stuff (like using a STUN server) and disable it. Regards, Patrick. From: asterisk-users-boun...@lists.digium.com on behalf of Sonny Rajagopalan Sent: 09 January 2015 01:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam score:10%] Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24<http://192.168.1.0/24> network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is appreciated. Does the fact that Asterisk is running on a VirtualBox VM on the same machine as one of the SIP phones matter? I am able to access the ARI REST interface of the Asterisk server quite fine on the host machine. I suspect it has to do with RTP not being set up, but all the codec support is there. Here's a log for the SIP request from 192.168.1.50<http://192.168.1.50/>: <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009<http://192.168.1.50:64009/> ---> INVITE sip:6002@192.168.1.139<mailto:sip%3A6002@192.168.1.139>;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 146.115.163.234:64009;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Max-Forwards: 70 Contact: To: mailto:sip%3A6002@192.168.1.139>;transport=UDP> From: mailto:sip%3Ademo-alice@192.168.1.139>;transport=UDP>;tag=b661670b Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.21933 r21903 Authorization: Digest username="demo-alice",realm="asterisk",nonce="[removed]",uri="sip:6002@192.168.1.139<mailto:sip%3A6002@192.168.1.139>;transport=UDP",response="[removed]",cnonce="[removed]",nc=0001,qop=auth,algorithm=md5,opaque="[removed]" Allow-Events: presence, kpml Content-Length: 245 v=0 o=Z 0 0 IN IP4 146.115.163.234 s=Z c=IN IP4 146.115.163.234 t=0 0 m=audio 8000 RTP/AVP 0 3 110 8 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009<http://192.168.1.50:64009/> ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 146.115.163.234:64009;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. From: mailto:sip%3Ademo-alice@192.168.1.139>>;tag=b661670b To: mailto:sip%3A6002@192.168.1.139>> CSeq: 2 INVITE Content-Length: 0 Any help is appreciated. A topology is shown below in ASCII. < ( Big bad Internet ) > GW/NAPT/Router | -- / \ || Host A Host B - - | Alice | | Bob | | 192.168.1.50 | | 192.168.1.149 | |---| |---| | Asterisk sr | |(VM) | | 192.168.1.239 | |---| On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan mailto:sonny.rajagopa...@gmail.com>> wrote: Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24<http://192.168.1.0/24> ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!)
Re: [asterisk-users] status - Unmonitored, how to change it [Spam score:11%]
I believe the "Unmonitored" status is linked to the "qualify" setting for each user. If they aren't set to "qualify=yes" then it won't check their status. Regards, Patrick. From: asterisk-users-boun...@lists.digium.com on behalf of Joseph Sent: 30 December 2014 18:58 To: asterisk-users@lists.digium.com Subject: [asterisk-users] status - Unmonitored, how to change it [Spam score:11%] How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/UsernameHost Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server(null) (D) 255.255.255.255 0 Unmonitored voip 184.89.249.114 (S) 255.255.255.255 4569 OK (91 ms) -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 11.5.0: blindxfer problems [Spam score:10%]
Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using “sip
info” for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
dialled any digits.
On 20/12/2014 20:52, "sean darcy" wrote:
>On 12/20/2014 03:22 PM, sean darcy wrote:
>> On 12/19/2014 09:42 AM, Rusty Newton wrote:
>>> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy
>>>wrote:
I've got a confbridge set up which works if dialed locally:
-- Executing [266@internal:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [266@internal:2] SendDTMF("DAHDI/1-1", "1") in new
stack
-- Executing [266@internal:3] ConfBridge("DAHDI/1-1", "1") in
new stack
-- Playing 'conf-onlyperson.ulaw' (language 'en')
...
extensions.conf:
[globals]
...
GOTO_ON_BLINDXFR="internal,266,1"
features.conf:
[featuremap]
blindxfer => #1
But:
-- Executing [s@DialOut:14] Dial("DAHDI/1-1",
"motif//+12345678...@voice.google.com,,rTt") in new stack
-- Called motif//+12345678...@voice.google.com
-- Motif/+12345678...@voice.google.com-688c is proceeding
passing it to
DAHDI/1-1
-- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
-- Started music on hold, class 'default', on
Motif/+1234567...@voice.google.com-688c
-- Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
builtin_blindtransfer: No digits dialed.
-- Playing 'pbx-invalid.ulaw' (language 'en')
I'm expecting the blind transfer to GOTO internal,266,1.
If I input 266 at the transfer dial tone, the blind transfer occurs.
Do I have this set up incorrectly?
>>>
>>>
>>>https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Var
>>>iables
>>>
>>>
>>> "${GOTO_ON_BLINDXFR} - Transfer to the specified
>>> context/extension/priority after a blind transfer (use ^ characters in
>>> place of | to separate context/extension/priority when setting this
>>> variable from the dialplan)"
>>>
>>> Try using ^ characters as it mentions there.
>>>
>>
>> Thanks for the response, but no joy:
>>
>>
>> == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1'
>>
>> Playing 'pbx-transfer.ulaw' (language 'en')
>> [Dec 20 15:12:03] WARNING[12336][C-0012]: features.c:2550
>> builtin_blindtransfer: No digits dialed.
>>
>>
>> sean
>>
>>
>
>I also tried setting up a transfer as an applicationmap.
>
>conference => *7,peer/both,ConfBridge,1
>
>Seems to load:
>
>features reload
> == Parsing '/etc/asterisk/features.conf': Found
> == Registered Feature 'conference'
> == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7'
>
>but when the caller dials *7, there's no action, Nothing in the cli. The
>dtmf is just sent to the callee.
>
>Also tried having the callee dial *7, same result.
>
>Any help appreciated.
>
>
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Re: [asterisk-users] Asterisk y Ldap
On 16-12-14 14:00, Dario Estupinan wrote: Como integrar asterisk con Ldap.?? https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver Best, Patrick ps this mailing list uses the English language -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] [Spam score:8%]
Thank you once again Richard. I think that covers all my confusion. Regards, Patrick. From: Richard Mudgett mailto:rmudg...@digium.com>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Date: Tuesday, 9 December 2014 21:48 To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] [Spam score:8%] On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont mailto:p.beaum...@hatsoffsoftware.co.uk>> wrote: Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? That is correct. bridge_softmix is optimized for multi-party conferencing where passing control frames such as hold/unhold to other parties in the bridge is not a good idea. For example, if three parties are in a bridge and if party A pressed its hold button then that should not necessarily prevent parties B and C from talking to each other. Using bridge_softmix for a normal two party call is a last resort. It works reasonably well as a normal two party bridge technology but it is computationally expensive and not intended for that purpose. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%]
Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? From: asterisk-users-boun...@lists.digium.com on behalf of Richard Mudgett Sent: 09 December 2014 20:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont mailto:p.beaum...@hatsoffsoftware.co.uk>> wrote: Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple. What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI about hold music starting or stopping. Unloading bridge_softmix and then loading bridge_simple fixes the issue. Also does anyone have any documentation on what bridges I should be using? I can't seem to find anything in the upgrade documentation that says "MOH will no longer work in softmix, you should use simple". This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly.? The bridging technology bridge_softmix is only used by app_confbridge in Asterisk v11. Nothing else in v11 uses the bridging framework. Unless you were using app_confbridge, you were not using bridge_softmix in v11. The various bridging technology modules in v12 and later are for different scenarios. The bridging framework is smart enough to pick the best bridging technology available for the situation. If the situation changes during a call, the bridging framework can change the bridge technology to support the new situation. * bridge_simple is for normal two party communication. * bridge_native_rtp is a special case of two party bridge were both parties use RTP for media exchange. The native technology allows for direct media. * bridge_softmix is for multi-party bridges where you can have 1 to n users communicating in a conference. As you found out, bridge_softmix can be used as a fallback if bridge_simple is not available because it allows two party communication. * bridge_holding is a parking bridge technology to hold calls for later connection. Parties in a holding bridge cannot communicate with each other. * bridge_builtin_features and bridge_builtin_interval_feature provide functionality used by features.conf. These two modules are actually not bridging technologies but support code for features.conf functionality. You usually need to install all of the bridging technologies. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridge configuration in Asterisk 13
Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple. What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI about hold music starting or stopping. Unloading bridge_softmix and then loading bridge_simple fixes the issue. Also does anyone have any documentation on what bridges I should be using? I can't seem to find anything in the upgrade documentation that says "MOH will no longer work in softmix, you should use simple". This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly.? Regards, Patrick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ICE consuming High CPU
On 05-12-14 08:25, Mayank Kumar Gour wrote: Any help will be appreciated. Help us help you by providing much much much more information then you have right now. http://www.catb.org/esr/faqs/smart-questions.html HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 & LDAP
On 04-12-14 11:23, Phil Daws wrote: Is there still an LDAP driver as do not see it in the CentOS 6 repository ? AFAICT in EPEL only Asterisk 1.8.32.1 has one: http://koji.fedoraproject.org/koji/buildinfo?buildID=594932 If you need a newer version then you'll have to build it yourself or find a repo you trust that carries Asterisk 13 with the LDAP module. The OpenLDAP developers recommend to always use the latest OpenLDAP release and skip the distro shipped one. If you are going to build Asterisk packages yourself then you may want to build them against OpenLDAP 2.4.40 and use that. Or switch to CentOS7 which has an almost current OpenLDAP. Both Symas and the LTB Project have current OpenLDAP RPMs for EL6 (& EL7): https://symas.com/products/symas-openldap-directory/ http://ltb-project.org/wiki/ HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bristuff-0.4.0-RC4-xr7
On 21-10-14 12:36, Ray Image wrote: [snip] Thanks for your response. As far as I can see I have no choice but use Zaptel as that is downloaded as part of the bristuff script. If there is a better way of getting a HFC PCI card to play nicely with Asterisk please let me know :-) The HFC based Asterisk B410P and Hx8 + B400M cards are supported by DAHDI. If you use something with a HFC-S chip then you can use the sources below which add support for HFC-S to DAHDI. Note that they may not use the latest DAHDI version so you may have to figure out a way to get the patches applied to current DAHDI. The one time I tried the dahdi-hfcs stuff it seemed to work fine (very light usage only). http://sourceforge.net/projects/dahdi-hfcs/ http://www.openvox.cn/pub/drivers/dahdi-linux-complete/ HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.9.0 crash and restart
On 21-10-14 08:54, 為近 吉摩（情報システム本部）- Tamechika Yoshikiyo - wrote: Hi, My Asetrisk restarted after to output following warning message. [Oct 16 15:59:58] WARNING[17102][C-8e34]: chan_sip.c:4696 update_provisional_keepalive: Unable to cancel schedule ID 738278. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4696). That looks similar to this bug: https://issues.asterisk.org/jira/browse/ASTERISK-21387 Why not install the latest Asterisk version (11.13.1) so you have all the latest fixes and can see if the bug is still present? Also your Asterisk 11.9.0 version is subject to the POODLE vulnerability for which a fix is available in 11.13.1. http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.13.1.tar.gz HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS
Hi Bryan,
On 10/18/2014 11:47 PM, Bryan Burroughs wrote:
> All,
>
> Has anyone seen this before? This appears to be a Swift or app_swift
> bug. I'm having a difficult time finding any information or support on this.
I haven't used app_Swift with Cepstral but iirc it wasn't deemed very
stable.
> Asterisk version:
> Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux
> on 2014-08-11 13:55:25 UTC
> OS:
> Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST
> 2014 x86_64 x86_64 x86_64 GNU/Linux
If you are not tied to the certified Asterisk version then perhaps try
using the latest Asterisk version (currently 11.13.0).
> When Asterisk crashes, the backtrace always looks something like the
> following:
[snip]
> The "out of bounds" line looks like it may be pointing to the issue.
>
> *argv = {0x0, 0xb9b0 bounds>, 0x0}*
Have you tried contacting the app_swift developer and/or filed a bug at
https://issues.asterisk.org/jira/secure/Dashboard.jspa ?
> Should I look into using another TTS engine?
You could try UniMRCP which sits between Asterisk and Cepstral replacing
app_swift: http://unimrcp.org
HTH,
Patrick
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Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13
On 15-09-14 17:22, Rainer Piper wrote: Hi Patrick, github done ;-) Thanks! what is HTH ??? Hope this/that helps http://www.internetslang.com/ http://www.urbandictionary.com/define.php?term=internet%20slang HTH :) Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13
Hi Rainer,
On 15-09-14 09:07, Rainer Piper wrote:
Hi,
Info !!! not a question !!!
the pjsip logger is different:
[Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request
from '"1001" ' failed for '85.25.197.23:5071'
(callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching endpoint found
and here the RegEx for fail2ban to catch this log:
|NOTICE.* .*: Request from '.*' failed for '(:[0-9]{1,5})?' (.*) -
No matching endpoint found
Thanks for sharing. If you use github it would be nice if you could
submit a pull request so that it becomes part of the Asterisk rules in
the next Fail2ban version (0.9.1).
https://github.com/fail2ban/fail2ban/pulls
HTH,
Patrick
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Re: [asterisk-users] Asterisk secure fine tune - stop attack
On 04-09-14 16:44, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306 <http://213.136.81.166:9306>) to extension '34422' rejected because extension not found in context 'default'. Have a look at Fail2ban: http://www.fail2ban.org/wiki/index.php/Main_Page HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
On 02-09-14 22:52, Eric Wieling wrote: A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs enabled. Thanks Eric. The obvious difference is that your co-worker was using H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present in my sip.conf so it might be the codec. Time for more tinkering. Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
On 02-09-14 21:15, Eric Wieling wrote: As long as you are NOT transcoding video should work in Asterisk. Both apps were configured with identical (codec) settings so I don't see how it would require transcoding. If you did get it to work I would appreciate it if you could tell me which clients you used, the Asterisk version, the OS and the relevant Asterisk config. Thanks, Patrick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setup Own IP PBX Server
On 01-09-14 12:31, Chandran Manikandan wrote: [snip] I have installed Freepbx server and tried to configure sip extension. It's working fine. A better place for FreePBX related questions and to get help is: http://community.freepbx.org/ Or hire their professional FreePBX support: http://www.freepbx.org/support-and-professional-services If you want to learn more about Asterisk in general then a good start is to first read "Asterisk: The Definitive Guide, 4th Edition" and go through the wiki at http://wiki.asterisk.org. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RDNIS with tel: vs. sip: header
On 28-08-14 11:57, Positively Optimistic wrote:
Has anyone had success patching chan_sip.c so that Asterisk will
recognize the tel: header for RDNIS information?
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, "sip:", 4)) {
exten += 4;
} else if (!strncasecmp(exten, "sips:", 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header
(%s)?\n", exten);
return -1;
}
Audiocodes Mediant 2000 devices send this header as a tel:...
*[Aug 28 02:25:42] WARNING[1283][C-1574] chan_sip.c: Huh? Not an
RDNIS SIP header (tel:41068558XX)?*
*
*
*(number obscured for privacy purposes)*
Not a dev but have you tried something like this (hope the formatting
stays sane):
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, "sip:", 4)) {
exten += 4;
} else if (!strncasecmp(exten, "tel:", 4)) {
exten += 4;
} else if (!strncasecmp(exten, "sips:", 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", exten);
return -1;
}
HTH,
Patrick
--
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Re: [asterisk-users] Understanding local channels
On 25-08-14 17:06, Mitch Claborn wrote: Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. How about the info on the Asterisk wiki: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels On the left side there's a menu with examples and modifiers. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi > CAPI migration
full.4:[Aug 7 12:39:58] ERROR[1654] chan_dahdi.c: Unable to open channel 1: No such device or address full.4:[Aug 7 12:39:58] ERROR[1654] chan_dahdi.c: Unable to register channel '1-2' full.4:[Aug 7 12:39:58] VERBOSE[1654] logger.c: -- codec_dahdi: using generic PLC full.4:[Aug 7 12:39:58] ERROR[1654] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory full.4:[Aug 7 12:39:58] VERBOSE[1654] logger.c: codec_dahdi.so => (Generic DAHDI Transcoder Codec Translator) full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Registered application 'DAHDIBarge' full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: app_dahdibarge.so => (Barge in on DAHDI channel application) full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Setting global variable 'TRUNK' to '"DAHDI/G2"' full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Registered application 'DAHDIRAS' full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: app_dahdiras.so => (DAHDI ISDN Remote Access Server) full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Setting global variable 'OUT_1' to 'DAHDI/g0' full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Setting global variable 'OUT_2' to 'DAHDI/g1' full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Setting global variable 'OUT_3' to 'DAHDI/g3' full.4:[Aug 7 12:39:59] VERBOSE[1654] logger.c: == Setting global variable 'ASTCHANDAHDI' to '1' full.4:[Aug 7 18:12:04] VERBOSE[1707] logger.c: == Registered application 'DAHDIScan' full.4:[Aug 7 18:12:04] VERBOSE[1707] logger.c: app_dahdiscan.so => (Scan DAHDI channels application) full.4:[Aug 7 18:12:04] VERBOSE[1707] logger.c: == Registered application 'DAHDISendKeypadFacility' full.4:[Aug 7 18:12:04] VERBOSE[1707] logger.c: == Parsing '/etc/asterisk/chan_dahdi.conf': [Aug 7 18:12:04] VERBOSE[1707] logger.c: == Found full.4:[Aug 7 18:12:04] VERBOSE[1707] logger.c: == Parsing '/etc/asterisk/dahdi-channels.conf': [Aug 7 18:12:04] VERBOSE[1707] logger.c: == Found full.4:[Aug 7 18:12:04] VERBOSE[1707] logger.c: == Parsing '/etc/asterisk/chan_dahdi_additional.conf': [Aug 7 18:12:04] VERBOSE[1707] logger.c: == F ound Maybe the wcb4xx module is interfering with the mISDN modules. Try blacklisting the wcb4xx kernel module and restart DAHDI and mISDN. Or try stopping mISDN, restart DAHDI with the wcb4xx module configured and see if DAHDI is happy. Or something might be wrong with the udev rules. The xen-pciback hide's setup properly on the machines so the xen guests have access to the card. The only difference is between the 2 host machines host kernels: old: Linux 2.6.32-5-xen-686 new: Linux 3.13-0.bpo.1-amd64 With such an exotic setup and ancient versions of something that no longer exists, I think you are SOL. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi > CAPI migration
On 08-08-14 10:09, Toney Mareo wrote: Hello Thank you for your response. I thought it could be easier moving the old card to the new machine and using the DAHDI driver. Unfortunately my first attempt for this failed. The card shows up in the original machine as: dahdi_hardware -v pci::00:00.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card IIRC the wcb4xx module is correct for this card. 00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST] Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- What do you mean with "the OS-es were clones ..."? Did you create an image of the old Trixbox machine and installed that on the new machine? Which versions are you using of: the OS? dahdi-linux? dahdi-tools? libpri? asterisk? Are you using mISDN too? Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) No hardware timing source found in /proc/dahdi, loading dahdi_dummy In general it could be that the card is not recognized or that there's a conflict with another kernel module or your dahdi config is incorrect or the udev rules are missing or wrong, etc. Any ideas how to proceed from this point? Since you have provided not very much information one can only guess. Only by providing as much information as possible can you help us help you. So please answer all the questions, provide the versions as asked, give the output of lsdahdi, check the logfiles (/var/log/messages etc) for dahdi info, check the dahdi config and blacklist files and provide that with anything else you can think of. AFAIK the Junghanns QuadBRI ISDN card should work fine with a recent DAHDI so why do you want to change to an AVM C4 card that requires chan_capi that hasn't seen any recent development and does not support Asterisk 11 or later? Asterisk 1.8 will only be supported for another year and then you are stuck again with an obsolete system that no longer gets any (security!) updates. If you can, why not stay with the Junghanns card and put a fresh CentOS 6.5 + updates on the new machine together with the latest version of dahdi-linux, dahdi-tools, libpri and asterisk. And then copy your old dialplan over to the new machine and make the required changes so it works with the latest Asterisk 11. Or you could install the latest FreePBX iso on the new machine if you prefer a GUI. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi > CAPI migration
Hi Toney, Comments inline. On 07-08-14 12:10, Toney Mareo wrote: Hello Folks, I looking to migrate a pbx from one server to another. The original server has this ISDN card: 00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) The new server: 00:00.0 I2O: Digital Equipment Corporation StrongARM DC21285 (rev 04) << AVM ISDN Controller C4 Now I'm not fully aware of both's cards functions since the manuals are very brief. What I heard is that the later card does not support some NT commands what I might going to need at this migration. I can't really say if one card is "better" than the other one. Unfortunately the two uses completely different drivers. The first card uses dahdi, the second uses capi. I want this migration go as smooth as possible with the least downtime so I looking for some help maybe someone has more experience with these cards. In the past I have used Eicon Diva Server cards with chan_capi for years and they worked great. I I also read something about CAPI uses completely different dial plans. So all the Asterisk configurations are migrated from the old server to the new one. That is correct although the differences/changes you need to make are limited. Adjusting the Dial command will go a long way. By following this guide: http://www.asteriskguru.com/tutorials/avm_c4.html I have the C4 modul loaded. My asterisk box is Asterisk 1.6.0.26-FONCORE-r78 (Trixbox). That Asterisk version is rather old and assuming that's the old Trixbox CE it is without security updates since 2012. I recommend you use a fresh install of something like CentOS 6.5 + all updates and Asterisk 11.11.0 (or later if available) with the latest dahdi-linux, dahdi-tools and libpri releases. Also get the latest chan_capi from here: ftp://ftp.chan-capi.org/chan-capi/ The version with -HEAD in the name has the latest fixes and is the one I always used. How can I see that this C4 card is really working from asterisk? Once you have the AVM C4 kernel module loaded and Asterisk with chan_capi is installed and capi is enabled in the Asterisk config, start Asterisk and then you should have a 'capi' command in the CLI. Executing it should show you info about the status of your ISDN channels. What is the difference between the chan_dahdi.conf and chan_capi.conf? One is for DAHDI supported cards and the other is for CAPI cards like the Eicon Diva Server and the AVM C4. Can't I just tell it somewhere to use the new card and I don't have to touch the existing dialplans etc? Nope. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security Architecture or Security Evaluations Docs?
On 28-07-14 12:28, Jeffrey Walton wrote: [snip] Is there anything that includes the development process? I'm interested in the secure development items and testing. Info about the development of Asterisk can be found here: http://asterisk.org/community/developers https://wiki.asterisk.org/wiki/display/AST/Development Development related questions can best be asked on the asterisk-dev mailing list or on irc.freenode.net in #asterisk-dev. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security Architecture or Security Evaluations Docs?
On 26-07-14 14:23, Jeffrey Walton wrote: Does anyone know of Security Architecture or Security Evaluations documents that I could read? Searching is turning up no hits. For example, http://www.google.com/#q=security+evaluation+site:asterisk.org and http://www.google.com/#q=security+architecture+site:asterisk.org. Assuming "security+evaluation" refers to Common Criteria, I'm not aware of any Common Criteria initiatives in relation to Asterisk (nor FreeSWITCH, OpenSIPS, Kamailio, Yate or any other Open Source VoIP project I'm aware of). Asterisk is a toolbox with many flexible building blocks and not a product like Cisco CallManager with pre-defined features set in stone. As such it doesn't really make sense to get Asterisk certified, if possible at all. It would be like trying to certify C or Python. If EALx certification is your requirement then have a look at the CallManager as iirc it's EAL1 certified. Re "asterisk+architecture", Asterisk Security related best practices are described here: http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over 3G/4G Data
On 18-07-14 17:59, Tech Dude wrote: What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff’s breaking the connection. This will be for fixed wireless. I have read to use G.729 codec, and TCP for signaling to bypass firewalls. Besides that, what other settings are recommended? Changes in MTU? Changes in voice payload ms? Is there a better codec to use? Header compression? Use TLS/SRTP so the carrier can't do packet inspection/snooping and mess with or block your VoIP connections. They might throttle/block port 5060 and 5061 anyway in which case you should use different ports. I use Asterisk 11 with TLS/SRTP, G.722 and Android phones (4G, CSipSimple or Bria), iPhones (4G, Bria) and Polycom phones. G.722 on inter-office calls has been working great so no need for G.729 and Asterisk uses standard alaw/ulaw for regular PSTN calls. I use TCP, have made no MTU changes and use standard 20ms voice payload. Packet loss, latency and jitter are the enemy. Your router might be top notch but if the cell tower is overloaded and experiences too much packet loss, delays of more than 150ms and lots of jitter than you may get crappy sound quality. If possible, get some prepaid 4G sim cards for your router from different carriers and test which carrier consistently provides the best signal, least delay, packet loss and jitter. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and LDAP
On 20-06-14 15:05, Linus Lüssing wrote: [snip] having "[test_phone_120d]" in my sip.conf works fine. Ah wait - do I need to have a user both in LDAP and sip.conf and the only thing LDAP can do for me is the authentication/password checking? As far as I know, yes :) Cheers, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and LDAP
On 18-06-14 23:06, Linus Lüssing wrote: Hi, I'm trying to get Asterisk running with LDAP to be able to authenticate sip user registrations. I'm using Asterisk (1.8.13.1~dfsg1-3+deb7u3) on a Debian server. Unfortunately I wasn't successful so far. My res_ldap.conf looks like this (so pretty minimal): --- [_general] ;url=ldaps://ldap.chaotikum.org url=ldap://ldap.chaotikum.org protocol=3 basedn=dc=chaotikum,dc=org [sip] name = uid IIRC the recommendation in the latest Asterisk book is to use only a-z, numerics (0-9) and underscore. So if you have [t...@chaotikum.org] in sip.conf then that might not work because of the '@'. You can easily test this by adding a peer [test_1234] (so with the recommended syntax) and add it to your LDAP server with a password and then check if it registers. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SSL/TLS weakness impact on Asterisk authentication
On 10-06-14 23:44, Michelle Dupuis wrote: After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Asterisk' SIP authentication uses a digest. See http://tools.ietf.org/html/rfc3261 for more info (20.6 and onwards). That does not mean that the recent OpenSSL issues have no impact on Asterisk. They do if you configure SIP to use TLS transport or enable TLS for other parts (for example AMI). So it's highly recommended to install the updated OpenSSL packages containing the fixes. My Asterisk packages link dynamically against the OpenSSL libraries. Assuming your packages do the same then, once you have updated the OpenSSL packages to the latest ones with the fixes and restart Asterisk, you should be good to go. While the recent OpenSSL issues don't directly expose your account passwords, the Heartbleed bug can expose (parts of) the private key used by TLS. Once the Men in Black have your private key its possible to setup a Man (in Black) in the Middle attack and sniff those passwords. See http://heartbleed.com/ Unless you want to mess around with the Men in Black and leave your system vulnerable to attack, you should install all security updates ASAP and then restart the services that rely upon them. HtH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphone
On 09-06-14 08:52, Giles Coochey wrote: On 08/06/2014 22:01, Mark Robinson wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? Has anyone tried MicroSIP? http://www.microsip.org/ Nope but if it doesn't meet your needs then maybe have a look at Jitsi https://jitsi.org/ or Linphone https://www.linphone.org/ I prefer the client to have at least the following features: Security: - TLS - SRTP - ZRTP Codecs: - G722 - G729 Fight NAT (if IPv6 is not an option): - STUN - TURN - ICE Cheers, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iPhone TLS reg problem: FILE * open failed
On 06/08/14 16:32, Mehroz Ashraf wrote: Random guess: 1. Make sure you have enabled tls/ssl for that particular user/extension 2. Check the iphone date/time, should be within the certificate validation. Thank you for you suggestions. The iPhone has the correct date & time which is also within the date/time range of the client certificate. The extension is TLS enabled. Using the same details on an Android phone work fine. I just tried another iPhone client called "Join" and, in spite of a terrible interface, TLS registration works and TLS/SRTP calls work too. Guess it's a problem in Linphone. Cheers, Patrick On Jun 8, 2014 6:50 PM, "Patrick Laimbock" mailto:patr...@laimbock.com>> wrote: Hi, I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register with TLS to an Asterisk 11.10.0 box. The registration fails and I see this in the Asterisk console: == Problem setting up ssl connection: error:: lib(0):func(0):reason(0) [Jun 8 15:33:39] WARNING[8555]: tcptls.c:274 handle_tcptls_connection: FILE * open failed! Anyone know what that error means? The source code does not tell me much. FWIW the same setup works fine with an Android phone. Thanks! Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/__mailman/listinfo/asterisk-__users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iPhone TLS reg problem: FILE * open failed
Hi, I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register with TLS to an Asterisk 11.10.0 box. The registration fails and I see this in the Asterisk console: == Problem setting up ssl connection: error:: lib(0):func(0):reason(0) [Jun 8 15:33:39] WARNING[8555]: tcptls.c:274 handle_tcptls_connection: FILE * open failed! Anyone know what that error means? The source code does not tell me much. FWIW the same setup works fine with an Android phone. Thanks! Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get last dialed number in a context?
On 03-06-14 11:31, Stefan Gofferje wrote:
Hi,
I would like to implement an auto-redial function in a context. The idea
is about like this:
Dial a number
Hear busy
Hangup
Pick up again
Dial a code like *123
=> jumps into a context which redials until callresult is not busy
Maybe like this:
[autoredial]
exten => s,1,Set(number=${CHANNEL(lastdialed)})
exten => s,2,Dial(SIP/${number}@account,60,g)
exten => s,3,Wait(15)
exten => s,4,GotoIf( [ "${DIALSTATUS}" = "BUSY" ]?2)
exten => s,5,Hangup
For that I'd need to somewhere get the last dialed number from the
channel/line I'm initiating the call from. Is something like this
already implemented?
Have you looked at Call Completion Supplementary Services (CCSS)?
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243096
Cheers,
Patrick
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Re: [asterisk-users] Kernel and DAHDI
On 12-05-14 09:30, Thorsten Göllner wrote: That's correct. When you update the kernel package youhave also to recompile dahdi package. AFAIK that's not true for RHEL6/CentOS6 as the EL6 kernels are ABI compatible so you don't need to recompile DAHDI when there's a new EL6 kernel. When updating a RHEL5/CentOS5 kernel I always had to recompile DAHDI for the new EL5 kernel. HtH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One mailbox for multiple extensions with individual greetings
On 11-05-14 05:10, John T. Bittner wrote: Why don't you use the voicemail copy feature? Create 3 mailboxes 1234, 6789 and 2000 for the shared. VoiceMail(1234@default&2000@default,su) VoiceMail(6789@default&2000@default,su) Set both 1234 and 6789 to email the voicemail to a fake email address and delete after email. A copy of the message for each will be dropped into 2000 and deleted from the original box. Thanks John. I'll give that a try. Cheers, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One mailbox for multiple extensions with individual greetings
Hi, Is there a way in Asterisk 11 to use a single voicemailbox for multiple extensions while still hearing each extension's individual greeting? Use case: someone has 2 numbers and wants all voicemail messages for both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER you would hear "the person at extension 1234 is unavailable" and the message would be stored in mailbox "mymailbox" and when dialing 6789 and NOANSWER you would hear "the person at extension 6789 is unavailable" and that message would also be stored in mailbox "mymailbox". The user then dials an extension to reach mymailbox and hears all messages for both the 1234 and 6789 numbers. I think I can solve it by symlinking /var/spool/asterisk/voicemail/default/6789/INBOX to /var/spool/asterisk/voicemail/default/1234/INBOX (and the other directories too) but it would be nice if this could be done within the dialplan. If that's not possible, would adding an "extension" option to app_voicemail.c solve this by decoupling the extension from the mailbox? Thanks for any pointers. Cheers, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR installation error
On 30-04-14 12:50, [Digital^Dude] ® wrote: make gives this: IIRC Digium's policy is that there's no support on this list for patented technologies like AMR which are possibly not officially licensed. Obviously to prevent any legal liability. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk support for h.324m
On 29-04-14 20:41, [Digital^Dude] ® wrote: Hello, If anyone has successfully compiled asterisk with: app_rtsp codec_amr mp4_play mp4_save app_transcode h324m_call Please share the versions of OS software, and libraries used. Lets make this thread useful so that all tried and tested video resources of asterisk can be found in one place for ease of access and later reference. I haven't but the guys you could talk to are the (former Fontventa?) folks at http://www.medooze.com/products/h324m-stack.aspx Source code/binaries can be viewed/downloaded at: http://sourceforge.net/p/asteriskvideo/ http://sourceforge.net/projects/mcumediaserver/ The old Fontventa AMR patch does not apply to Asterisk 11. I tried to port it to Asterisk 11 but couldn't get a call going. If you are a developer focused on Asterisk 11 and want to have a look at the AMR patch let me know and I'll email it to you. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk issue
On 28-04-14 20:13, Haley,Scott A wrote: That seemed to fix it. Thanks to everyone. https://bugzilla.redhat.com/show_bug.cgi?id=1092150 HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk issue
On 28-04-14 19:49, Haley,Scott A wrote: Now I am getting Permission denied. Have you checked if SELinux is blocking the app? Any blockage should show up as an 'AVC' in /var/log/audit/audit.log You can temporarily set SELinux to permissive with 'setenforce 0' and check if the problem goes away. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and SRTP
On 04/05/2014 07:56 PM, William Wu wrote: Hi experts, I am trying Asterisk SRTP in my environment, and find that when Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay by Asterisk are local ports on the Asterisk server, media from the two clients out of the NAT (for example from Internet) can not reach the ports, and thus the two client can not establish the secure call via Asterisk. I have set up a STUN server and configured in rtp.conf, but seems Asterisk does not do STUN before it opens ports for SRTP. BTW, Non-SRTP call can work though. Anyone can give advice on how to make SRTP work in such an env? I have no problems with a TLS/SRTP call between a GSM with CSipSimple and Asterisk 11.8.1 behind NAT. Have you configured the NAT options in sip.conf? externip=... localnet=... nat=... You may also need to add/change the options below. Check the sip.conf example file to see what these options do and use what's best for your situation. canreinvite=no directmedia=no directrtpsetup=no HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CLI command to see if SRTP is active?
Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TLS/SRTP with Asterisk 11.8.1
On 24-03-14 21:28, Patrick Laimbock wrote: [snip] == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection: So others may find the fix: make sure the server and client certificates have the proper keyUsage. The ast_gen_tls script does not set them and this caused the handshake/verification to fail. The client certificate needs something like: keyUsage = digitalSignature, keyEncipherment extendedKeyUsage = clientAuth The server certificate needs something like: keyUsage = digitalSignature, keyEncipherment extendedKeyUsage = serverAuth HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with TLS/SRTP with Asterisk 11.8.1
Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent "CSipSimple_crespo-19/r2330" for peer encrypted SSL certificate ok == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Mar 24 21:20:42] WARNING[28466]: tcptls.c:272 handle_tcptls_connection: FILE * open failed! [Mar 24 21:20:45] NOTICE[28460]: chan_sip.c:29584 sip_poke_noanswer: Peer 'encrypted' is now UNREACHABLE! Last qualify: 0 SSL certificate ok == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection: FILE * open failed! -- Unregistered SIP 'encrypted' sip.conf looks like this: [general] context=guest allowguest=no allowoverlap=no allowtransfer=no bindaddr=0.0.0.0:5060 udpbindaddr=0.0.0.0:5060 tcpenable=no tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 transport=udp preferred_codec_only=no disallow=all allow=ulaw language=en trustrpid=no dtmfmode=rfc2833 videosupport=no alwaysauthreject=yes directmedia=no jbenable = yes jbforce = no [encrypted] type=friend secret=1234 context=internal callerid="Encrypted" <1002> host=dynamic qualify=yes canreinvite=no dtmfmode=rfc2833 disallow=all allow=alaw allow=ulaw transport=tls encryption=yes $ ls -l /etc/asterisk/keys total 28 -rw-r--r--. 1 asterisk asterisk 1204 mrt 24 16:16 asterisk.crt -r--r-. 1 asterisk asterisk 887 mrt 24 16:16 asterisk.key -r--r-. 1 asterisk asterisk 2091 mrt 24 16:16 asterisk.pem -rw-r--r--. 1 asterisk asterisk 1736 mrt 24 16:16 ca.crt -r. 1 asterisk asterisk 3311 mrt 24 16:16 ca.key -rw-r--r--. 1 asterisk asterisk 1208 mrt 24 16:20 nexus.crt The certs were created with ast_tls_cert as described in the tutorial. I created a nexus.p12 for the phone and imported it before configuring CSipSimple. Does anyone know what's wrong? Pointers much appreciated. Thanks, Patrick [0] https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi + dlink du128ta
On 03/15/2014 10:15 PM, binary wrote: i have tried the misdn from git. my problem is that it needs LCR and it fails to get installed Then you need to fix that. AFAIK there is no other way to use a USB ISDN TA than via mISDN/LCR. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi + dlink du128ta
On 03/14/2014 07:53 AM, binary dreamer wrote: hello everyone, I do have a usb ISDN modem that I would like to make it work with dahdi. is it possible? No. I am running debian 7, with dahdi 2.9, asterisk 11.8 dahdi cannot find it at the moment, unless there is something else to be done. Try building & installing the stuff from http://misdn.eu Works fine with a USB TA with HFC chipset last time I tested it. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux call router
On 11-03-14 12:15, binary dreamer wrote: hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory /usr/src/lcr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory /usr/src/lcr' make: *** [all] Error 2 root@voyage:/usr/src/lcr# could someone help me please? I have not built lcr in a while (on CentOS) but iirc: make sure you get from http://misdn.eu/ the latest isdn4k-utils, mISDN, mISDNuser and lcr. Then build in that order. For lcr I had these build requirements: autoconf automake libtool libtiff-devel mISDNuser-devel ncurses-devel openssl-devel If you still can't figure it out perhaps ask on the ISDN4Linux mailing list: https://www.isdn4linux.de/mailman/listinfo/isdn4linux Cheers, Patrick ps don't build as root, it's bad practice. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashes at "meetme kick all"
On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote: Dear Forum, I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing “meetme kick all” CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which release of asterisk this issue has been fixed. Please help. _https://issues.asterisk.org/jira/browse/ASTERISK-15741_ AFAICT this issue has not been fixed due to inactivity. Note the "Suspended due to lack of activity" remark. Also the 1.6 version mentioned in the bugreport is EOL. Version 10.0.0 you mentioned is also EOL so any bugreport you file against version 10.0.0 will not be acted upon unless you can reproduce it with the latest Asterisk version 11.x.x or 12.x.x. I recommend you upgrade to an Asterisk LTS (long term support) version like the latest 11.x.x (currently 11.7.0). For more information about Asterisk LTS versions go to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can report it at the Asterisk issue tracker at: https://issues.asterisk.org/jira/secure/Dashboard.jspa Before filing a bug please read the information at: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug -- Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
