Re: [asterisk-users] no audio both ways with ipv6
On Thu, 2021-10-14 at 21:21 +0200, Antony Stone wrote: > On Thursday 14 October 2021 at 19:22:00, hw wrote: > > > Hi, > > > > when asterisk registers with the VOIP provider via ipv6 and when > > local phones don't work with ipv6 but only with ipv4, am I to > > expect issues? > > Do a SIP packet capture and see what the SDP in the INVITE is telling each > end > to expect from the other. Hmm I could try that maybe, as a last resort. > > I'm receiving incoming calls via the provider, asterisk correctly > > dials the phone where the calls are suposed to go to, the phone > > rings --- and when I pick it up, there is no audio in either direction. > > Sounds like the setup is trying to do direct media - which obviously cannot > work between an IPv4-only phone and an IPv6-only provider. > > Make sure Asterisk remains in the audio path and it should "almost transcode" > for you. I thought about that, and I think direct media isn't being used. It works with ipv4, and if it was using direct media, ipv4 wouldn't work, either. IIRC I tried with 'aor (or endpoint?)/direct_media = no'. Unfortunately, I can't really make test calls to try things out. When a call comes in and I pick up the phone, asterisk says it has learned the ipv6 address of the VOIP provider on one side and the ipv4 address on the other, and the channels are joining a simple bridge --- whatever that means. Is there something that would tell me if asterisk is trying to set up direct media or remains in between? > I have audio working over just such an arrangement (in my case, an IPv4-only > provider, and phones connected via IPv6) without problems. I wish I could try with an ipv6 phone, but I couldn't get my Polycom VVX 1500D to work with ipv6 at all. It was suggested on the Polycom forum that the firmware is too old and that I update to the latest, but the latest doesn't run on this phone because the phone is too old. The release before the latest is supposed to work, but that is nowhere to be found, and I didn't get any more answers. I tried Twinkle on my computer, but that doesn't support ipv6 at all. There must be something special going on with ipv6 when it comes to SIP and/or RTP. > > Antony. > > -- > The difference between theory and practice is that in theory there is no > difference, whereas in practice there is. > >Please reply to the list; > please *don't* CC me. > -- _ -- 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] no audio both ways with ipv6
Hi, when asterisk registers with the VOIP provider via ipv6 and when local phones don't work with ipv6 but only with ipv4, am I to expect issues? I'm receiving incoming calls via the provider, asterisk correctly dials the phone where the calls are suposed to go to, the phone rings --- and when I pick it up, there is no audio in either direction. There are no packets showing up in the logs as being rejected by the firewall. I can make outgoing calls just fine, and those are with the VOIP provider on one side and the same ipv4 phone on the other. How can it be that incoming calls have no audio? -- _ -- 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] memory issues
On Fri, 2020-09-25 at 21:32 -0400, Sean Bright wrote: > https://issues.asterisk.org/jira/browse/ASTERISK-28695 > Thanks! The fix doesn't fix it because the cache must be considered; the bufferram isn't so relevant. A few kB more doesn't make much difference. Could/should I re-open this bug report? -- _ -- 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] call an IP camera?
On Sat, 2020-10-03 at 15:51 +0200, Antony Stone wrote: > On Thursday 24 September 2020 at 16:31:33, hw wrote: > > > Hi, > > > > is it possible to "call" an IP camera? I'm thinking about something like > > bridging with a music stream, but instead of streaming audio, bridge with > > the video stream from the camera. > > I'm curious - did you manage to get anywhere with this? Unfornuately not --- would be a cool featuere, though ... -- _ -- 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] call an IP camera?
On Thu, 2020-09-24 at 18:45 +0200, Antony Stone wrote: > On Thursday 24 September 2020 at 18:28:13, hw wrote: > > > On Thu, 2020-09-24 at 16:57 +0200, Antony Stone wrote: > > > I would start with something like > > > https://www.voip-info.org/asterisk-config-musiconholdconf/ > > > https://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf > > > (or any more up to date documentation if you can find it). > > > > > > I've never tried that with video, but given how the media negotiation > > > between Asterisk and SIP devices is handled, I would expect it to work > > > given compatible codecs. > > > > Unfortunately, musiconhold.conf doesn't understand rtsp: > > [test] > > mode=playlist > > entry=rtsp://10.10.30.20/12 > > Have a look at https://www.voip-info.org/asterisk-config-musiconholdconf/ and > the section headings "Stream radio using MPlayer for MOH" and "Example using > asx (mms://)(.wmv) streams. (or “anything” that mplayer can play)." > > Those look promising to me. [test] mode=custom application=ffmpeg -i rtsp://10.10.30.20/12 -map 0:0 -f rawvideo pipe:1 WARNING[100823]: res_musiconhold.c:794 monmp3thread: poll() failed: Interrupted system call Other than getting lots of error messages as above, the command basically works in that ffmpeg pipes the video to STDOUT. I can use 'ffmpeg -i rtsp://10.10.30.20/12 -map 0:0 -f matroska pipe:1 > some_file' and then play the file with mpv (rawvideo doesn't work with mpv --- but should work with a phone?). Why is the system call being interrupted all the time? Because asterisk doesn't take video for music? -- _ -- 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] memory issues
Hi,
ever since I have switched my server from Centos 7 to Fedora 32, asterisk
is showing memory issues and no calls are possible. I'm using the asterisk
that comes with Fedora; before that, I used a self-compiled version on
Centos. The hardware is still the same. Asterisk shows the following
message when trying to make a call:
WARNING[92530]: pbx.c:4644 increase_call_count: Available system memory
(~168MB) is below the configured low watermark (1024MB)
I was thinking that asterisk is leaking memory because the only way to get
asterisk to work again was by restarting the server. Since this is very
annoying and the Fedora bug report remains ignored, I finally started to
investigate. The relevant source is attached in asnip.c.
To see what's going on, I wrote a test program, attached as sysinfo.c. You
can simply compile it with 'cc -O2 sysinfo.c -o sysinfo'.
The output is follows:
./sysinfo
unit size: 1 byte(s)
#
freeram: 153
bufferram: 5
Sum: 158
#
trying to allocate 158
sleeping 5 seconds
allocate twice as much
sleeping 5 seconds
memset allocated memory to 0
sleeping 5 seconds
memory freed
So what asterisk says is about right. When I look at the info from 'cat
/proc/meminfo', I see this:
cat /proc/meminfo
MemTotal: 16361780 kB
MemFree: 151844 kB
MemAvailable: 14883060 kB
Buffers:5468 kB
Cached: 14773136 kB
SwapCached: 224 kB
Active: 2019568 kB
Inactive: 13592124 kB
[...]
which would mean that I have 14GB buffered/cached, and 'free -h' confirms
this.
Apparently, the cache remains persistently occupied. Since I'm currently
performing backups, it's not surprising that the cache is large.
Apparently it means that asterisk fails every time I'm doing something
that fills the cache :(
After understanding that the cache remains full, I figured there might be
way to flush the buffers and the cache. [1] shows how to do this, and
after 'sync; echo 3 > /proc/sys/vm/drop_caches', 'free -h' showed
buff/cache as 1.3Gi and asterisk was working again.
However, the backups are still going on, and it doesn't take long before
the cache is back at 14GB again and asterisk is blocked.
I think asterisk needs to consider the cache as free memory as well. Isn't
the cache supposed to automatically shrink when more memory is required?
As a workaround, I can set minmemfree in asterisk.conf to a low value.
Nonetheless, I guess that should be fixed. I'll update the Fedora bug
report with this.
[1]:
https://www.tecmint.com/clear-ram-memory-cache-buffer-and-swap-space-on-linux/
#if defined(HAVE_SYSINFO)
if (option_minmemfree) {
/* Make sure that the free system memory is above the configured low watermark */
if (!sysinfo(_info)) {
/* Convert the amount of available RAM from mem_units to MB. The calculation
* was done this way to avoid overflow problems */
uint64_t curfreemem = sys_info.freeram + sys_info.bufferram;
curfreemem *= sys_info.mem_unit;
curfreemem /= 1024 * 1024;
if (curfreemem < option_minmemfree) {
ast_log(LOG_WARNING, "Available system memory (~%" PRIu64 "MB) is below the configured low watermark (%ldMB)\n",
curfreemem, option_minmemfree);
failed = -1;
}
}
}
#endif
#include
#include
#include
#include
#include
#include
#define DISPLAYUNIT (1024 * 1024)
void sleeping(int seconds)
{
printf("sleeping %d seconds\n", seconds);
sleep(seconds);
}
int main(int argc, char argv[])
{
struct sysinfo sys_info;
memset(_info, 0, sizeof(sys_info));
if(!sysinfo(_info)) {
printf(
"unit size:\t%12u byte(s)\n#\nfreeram:\t%12lu\nbufferram:\t%12lu\n\nSum:\t\t%12lu\n",
sys_info.mem_unit,
sys_info.freeram * sys_info.mem_unit / DISPLAYUNIT,
sys_info.bufferram * sys_info.mem_unit / DISPLAYUNIT,
(sys_info.freeram + sys_info.bufferram) * sys_info.mem_unit / DISPLAYUNIT
);
unsigned long some_memory = (sys_info.freeram + sys_info.bufferram) * (unsigned long)sys_info.mem_unit;
printf("#\ntrying to allocate %lu\n", some_memory / DISPLAYUNIT);
void *allocated = malloc((size_t)some_memory);
sleeping(5);
if(allocated) {
printf("allocate twice as much\n");
free(allocated);
allocated = malloc((size_t)(some_memory + some_memory));
sleeping(5);
if(allocated) {
printf("memset allocated memory to 0\n");
memset(allocated, 0, (size_t)(some_memory + some_memory));
sleeping(5);
free(allocated);
printf("memory freed\n");
} else {
printf("larger memory allocation failed\n");
}
} else {
printf("memory allocation failed\n");
}
} else {
printf("error: sysinfo not available\n");
exit(-1);
}
exit(0);
}
--
_
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New
Re: [asterisk-users] call an IP camera?
On Thu, 2020-09-24 at 15:01 +, Ralph L. Miller wrote: > The Grandstream camera product line has SIP output so you can "call" the > camera Good to know, thanks! Unfortunately, I don't have a camera like that. -- _ -- 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] call an IP camera?
On Thu, 2020-09-24 at 16:57 +0200, Antony Stone wrote: > On Thursday 24 September 2020 at 16:31:33, hw wrote: > > > Hi, > > > > is it possible to "call" an IP camera? > > Only if it talks SIP (which some do, generally door entry cameras with a > push > button input and often a lock release output). > > > I'm thinking about something like bridging with a music stream, but > > instead > > of streaming audio, bridge with the video stream from the camera. > > So, maybe you should treat it like a music stream such as music on hold? > > > It would be very cool if I could just call the camera and see what's > > going > > on. Ffmpeg shows the following streams available from the camera: > > > > Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, > > 12 > > fps, 12 tbr, 90k tbn, 24 tbc > > > > Stream #0:0: Video: h264 (Main), yuv420p(progressive), 640x352, 12 > > fps, > > 12 tbr, 90k tbn, 24 tbc > > > > Perhaps it's not even necessary to recode the stream? > > Very likely, but what you're looking at there is the media format; you > also > need some sort of signalling protocol if you're going to call it from > Asterisk. > > I would start with something like > https://www.voip-info.org/asterisk-config-musiconholdconf/ > https://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf > (or any more up to date documentation if you can find it). > > I've never tried that with video, but given how the media negotiation > between > Asterisk and SIP devices is handled, I would expect it to work given > compatible codecs. Unfortunately, musiconhold.conf doesn't understand rtsp: #033[1;37mmoh_parse_options#033[0m: Playlist entries must be a URL or absolute path, 'rtsp://10.10.30.20/12' provided. Asterisk then ignores the configured music class when it's given like this in musiconhold.conf (and plays music from the default class instead): [test] mode=playlist entry=rtsp://10.10.30.20/12 So I guess that musiconhold may be limited to audio only. But who knows? What are the requirements for the URLs that can be used with the 'playlist' option in musiconhold.conf? It's generally possible to stream stuff to devices (like phones), like when using the Playback() dialplan application to stream audio. Is it somehow possible to stream audio from programs into channels from the dialplan or somewhere else without using musiconhold.conf? If that was possible, it might be possible to stream video instead. Does pjsip support video? [1] would indicate that it doesn't. However, that information seems to be over 8 years old :( [1]: https://wiki.asterisk.org/wiki/display/AST/Video+Telephony -- _ -- 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] call an IP camera?
Hi, is it possible to "call" an IP camera? I'm thinking about something like bridging with a music stream, but instead of streaming audio, bridge with the video stream from the camera. It would be very cool if I could just call the camera and see what's going on. Ffmpeg shows the following streams available from the camera: Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 12 fps, 12 tbr, 90k tbn, 24 tbc Stream #0:0: Video: h264 (Main), yuv420p(progressive), 640x352, 12 fps, 12 tbr, 90k tbn, 24 tbc Perhaps it's not even necessary to recode the stream? -- _ -- 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 make a bug report
On Saturday, April 18, 2020 5:42:11 PM CEST Joshua C. Colp wrote: > On Sat, Apr 18, 2020 at 8:47 AM hw wrote: > > Hi, > > > > how do I make a bug report? I filled in the form to make a report and > > https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues > > reported by me. > > If successful then JIRA will redirect you to the newly created issue. It didn't, the form disappeared and nothing further happened. So I have to assume it doesn't work. > > If someone knows how to get asterisk to re-register when using pjsip after > > the > > registration shows as Rejected, like after the internet connection to the > > VOIP > > provider goes away (and comes back), please let me know. This bug makes > > pjsip > > makes basically unusable :( > > There are various options in the outbound registration that controls > behavior. I'd suggest providing your actual configuration. I have put the options that should make asterisk re-register in pjsip_wizard.conf as much as I could find them like this: [easybell_HW] type = wizard sends_auth = yes sends_registrations = yes max_retries = 0 auth_rejection_permanent = no forbidden_retry_interval = 200 transport = transport-tls endpoint/cos_audio = 5 endpoint/cos_video = 4 remote_hosts = secure.sip.easybell.de:5061 aor/qualify_frequency = 30 outbound_auth/username = ... outbound_auth/password = ... endpoint/allow = !all,g722,alaw,ulaw endpoint/context = ingressEasybell endpoint/media_encryption = sdes registration/contact_user = extenHW In pjsip.conf is only the transport: [transport-tls] type=transport protocol=tls bind=192.168.3.50:5061 ca_list_file=/etc/pki/tls/certs/ca-bundle.crt cert_file=/etc/asterisk/cert/newc/mycert.pem priv_key_file=/etc/asterisk/cert/newc/mykey.pem After I finally found out that 'pjsip send register *all' should re-register, I tried it while it was still registered, and it said "Re-register all queue". After that, it kept saying that all the registrations are now "Unregistered". Neither 'pjsip send unregister *all', nor 'pjsip send register *all' have any effect other than giving the message "Unregister all queued" or "Re-register all queue". I had to restart asterisk again to get it to register again. On a side note, asterisk doesn't apply any QoS markers, either ... -- _ -- 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] how to make a bug report
Hi, how do I make a bug report? I filled in the form to make a report and https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues reported by me. If someone knows how to get asterisk to re-register when using pjsip after the registration shows as Rejected, like after the internet connection to the VOIP provider goes away (and comes back), please let me know. This bug makes pjsip makes basically unusable :( -- _ -- 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: how to survive rejected registrations?
On Saturday, February 29, 2020 11:29:37 AM CET Administrator wrote: > Le 28/02/2020 à 23:43, hw a écrit : > > On Thursday, February 27, 2020 3:03:47 PM CET hw wrote: > >> Hi, > >> > >> sometimes 'pjsip show registrations' shows registrations to the VOIP > >> provider as Rejected. I have already added > >> > >> > >> max_retries = 0 > >> auth_rejection_permanent = no > >> > >> > >> in pjsip_wizard.conf and still asterisk does not recover. > >> > >> I need asterisk to keep trying to register and to renew the registration > >> without requiring manual intervention. How can I make asterisk do that? > > > > No ideas? > > > > If pjsip is not able to recover after the internet connection has gone > > away > > for a few minutes, it's totally useless. > > A workaround is to have a cron script which looks if your asterisk is > registered and if not to send again the register command Thanks, I'll try that if I can find out which command that is :) This shouldn't be necessary, though. Before switching to PJSIP, there was no problem with registrations going away and not coming back. Is PJSIP still too buggy to be used and not recommended? Maybe I'll make a bug report ... -- _ -- 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: how to survive rejected registrations?
On Thursday, February 27, 2020 3:03:47 PM CET hw wrote: > Hi, > > sometimes 'pjsip show registrations' shows registrations to the VOIP > provider as Rejected. I have already added > > > max_retries = 0 > auth_rejection_permanent = no > > > in pjsip_wizard.conf and still asterisk does not recover. > > I need asterisk to keep trying to register and to renew the registration > without requiring manual intervention. How can I make asterisk do that? No ideas? If pjsip is not able to recover after the internet connection has gone away for a few minutes, it's totally useless. -- _ -- 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] error compiling current git
On Thursday, February 27, 2020 4:29:01 PM CET Kevin Harwell wrote: > On Thu, Feb 27, 2020 at 8:51 AM hw wrote: > > Hi, > > > > compiling the current git version on Centos 7 gives me: > >[CC] res_statsd.c -> res_statsd.o > > > > res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified > > in initializer > > > > .on_valid_pair = ast_rtp_on_valid_pair, > > ^ > > > > res_rtp_asterisk.c:2669:2: warning: initialization from incompatible > > pointer type [enabled by default] > > res_rtp_asterisk.c:2669:2: warning: (near initialization for > > ‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default] > > > >[CC] res_format_attr_g729.c -> res_format_attr_g729.o > > > > Is this to be expected or should I make a bug report? > > When you pulled the lasted code this change would have forced a > re-configure. If you haven't already try doing a full clean and rebuild, > and see if you still have the error: > > $ make distclean > $ ./configure [your options] > $ make Thanks, that worked :) -- _ -- 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] error compiling current git
Hi, compiling the current git version on Centos 7 gives me: [CC] res_statsd.c -> res_statsd.o res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified in initializer .on_valid_pair = ast_rtp_on_valid_pair, ^ res_rtp_asterisk.c:2669:2: warning: initialization from incompatible pointer type [enabled by default] res_rtp_asterisk.c:2669:2: warning: (near initialization for ‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default] [CC] res_format_attr_g729.c -> res_format_attr_g729.o Is this to be expected or should I make a bug report? -- _ -- 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] pjsip: how to survive rejected registrations?
Hi, sometimes 'pjsip show registrations' shows registrations to the VOIP provider as Rejected. I have already added max_retries = 0 auth_rejection_permanent = no in pjsip_wizard.conf and still asterisk does not recover. I need asterisk to keep trying to register and to renew the registration without requiring manual intervention. How can I make asterisk do that? -- _ -- 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 make asterisk set cos values
On Friday, January 31, 2020 12:33:17 PM CET hw wrote: > Hi, > > examining the network traffic with wireshark shows that asterisk does not > set any QoS values at all. > > What do I need to do to make asterisk set QoS values (on Centos 7)? > > The wiki says to use vconfig to set QoS values[1]. What does the > skb-priority need to be set to? How do you use vconfig on interfaces that > are not VLAN interfaces? > > Is it generally impossible to set QoS values on bonding interfaces? > > > [1]: https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service Any ideas? -- _ -- 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 make asterisk set cos values
On Friday, January 31, 2020 12:45:40 PM CET Joshua C. Colp wrote: > On Fri, Jan 31, 2020 at 7:34 AM hw wrote: > > Hi, > > > > examining the network traffic with wireshark shows that asterisk does not > > set > > any QoS values at all. > > > > What do I need to do to make asterisk set QoS values (on Centos 7)? > > > > The wiki says to use vconfig to set QoS values[1]. What does the > > skb-priority > > need to be set to? How do you use vconfig on interfaces that are not VLAN > > interfaces? > > > > Is it generally impossible to set QoS values on bonding interfaces? > > > > > > [1]: https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service > > Further down that page it talks about the options for both chan_sip and > chan_pjsip for setting TOS and CoS values. It can be done in configuration. Well, yes, I have set these options. I am under the impression that asterisk is using default values when these options are not set, but the wiki doesn't say. Do I need to create a VLAN interface? I have also installed libcap and recompiled, and I don't know how to tell if it's actually used or not. I tried with selinux set to permissive to no avail. I could run asterisk as root and see if the values get set, but that might change ownership on files asterisk creates and could cause trouble later. Asterisk even says on the console " == Using SIP RTP Audio CoS mark 5". I can see that packets from phones are marked and the ones from asterisk are not. [xxx] type = wizard accepts_auth = yes accepts_registrations = yes endpoint/cos_audio = 5 endpoint/cos_video = 4 [...] [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cos=3 [...] I never checked this, and now it's time that I want to know for sure and set it up like it's supposed to be. It's silly to have traffic control in place when the packets are not marked so it doesn't work to begin with. -- _ -- 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] how to make asterisk set cos values
Hi, examining the network traffic with wireshark shows that asterisk does not set any QoS values at all. What do I need to do to make asterisk set QoS values (on Centos 7)? The wiki says to use vconfig to set QoS values[1]. What does the skb-priority need to be set to? How do you use vconfig on interfaces that are not VLAN interfaces? Is it generally impossible to set QoS values on bonding interfaces? [1]: https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service -- _ -- 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] delivery verification of instant messages with pjsip
On Thursday, January 30, 2020 2:38:31 PM CET Joshua C. Colp wrote: > On Thu, Jan 30, 2020 at 9:35 AM hw wrote: > > Hi, > > > > when sending IMs from endpoint to endpoint with the MessageSend() > > application, > > I can check the MESSAGE_SEND_STATUS and send another message to the sender > > of > > the message to notify them that their message was not sent when the status > > indicates it. > > > > This works fine with chan_sip. With chan_pjsip, this works differently in > > that MESSAGE_SEND_STATUS is "SUCCESS" after sending the message, and only > > later asterisk figures out that it is "Unable to retrieve contact for > > endpoint > > " when there are no contacts and thus the message never gets > > delivered. > > > > How can I check if the endpoint has contacts --- or preferably --- if the > > message was actually delivered to an endpoint? It would be sufficient to > > get > > it to work with endpoints that are not supposed to have more than one > > contact. > > Making MESSAGE_SEND_STATUS reflect whether the message was sent or not for > PJSIP was merged in 2 days ago[1]. It will be in a future release. If you > don't want to wait you could use device state to know if the device is > reachable (and thus a MESSAGE has a chance of being sent) using the > DEVICE_STATE dialplan function[2]. Perfect answer, thanks! :) I think I'll just update from the git repo then and see if it works. > [1] https://gerrit.asterisk.org/c/asterisk/+/13674 > [2] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_DEVICE_STATE -- _ -- 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] delivery verification of instant messages with pjsip
Hi, when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. This works fine with chan_sip. With chan_pjsip, this works differently in that MESSAGE_SEND_STATUS is "SUCCESS" after sending the message, and only later asterisk figures out that it is "Unable to retrieve contact for endpoint " when there are no contacts and thus the message never gets delivered. How can I check if the endpoint has contacts --- or preferably --- if the message was actually delivered to an endpoint? It would be sufficient to get it to work with endpoints that are not supposed to have more than one contact. -- _ -- 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] solved: PJSIP and Grandstream Wave with TSL and SRTP
Hi, I've got it to work with the following transport: [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 ca_list_file=/etc/pki/tls/certs/ca-bundle.crt cert_file=/etc/asterisk/cert/newc/himinbjorg.adminart.net.pem priv_key_file=/etc/asterisk/cert/newc/himinbjorg.adminart.net.key.pem This is using a self-signed certificate. Note that I omitted 'method='. On Wednesday, January 22, 2020 3:18:23 AM CET hw wrote: > Hi, > > after switching from chan_sip to chan_pjsip, a device running Grandstream > Wave leads to the following error message on the asterisk console: > > > SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:43357 > > > Something with the encryption must have changed with asterisk. How can I > get the device to register again? > > > [transport-tls] > type = transport > protocol = tls > bind = 0.0.0.0:5061 > tos = cs5 > cert_file = /etc/asterisk/cert/asterisk.pem > ca_list_file = /etc/pki/tls/certs/ca-bundle.crt > method = sslv23 > > > 'method = tlsv1' doesn't work, either. -- _ -- 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] Get PJSIP Endpoint Information via REST or similar API?
On Monday, January 27, 2020 10:03:27 AM CET Benoit Panizzon wrote: > Hi Gang > > To get our customers more information on how they registered I am > looking for a elegant way to get an information like the CLI command: > > pjsip show endpoint [endpoint] > > I had a got on ARI, but that basically only returns the information if > an endpoint is online or not. > > Is there a API to get similar detailed information as the cli > command? If anything else fails, you could parse the output of "asterisk -x 'pjsip show endpoint '", or send it to the customer. -- _ -- 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 and Grandstream Wave with TSL and SRTP
On Friday, January 24, 2020 6:25:48 PM CET Sean Bright wrote: > On 1/23/2020 6:04 PM, hw wrote: > >> This is what mine looks like which works just fine: > >> > >> [transport-tls] > >> type = transport > >> protocol = tls > >> method= tlsv1_2 > >> cipher= > >> ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES > >> 128 > >> -GCM-SHA256,ECDHE-RSA-AES128-GCM-SHA256,ECDHE-ECDSA-AES256-SHA384,ECDHE- > >> RSA- AES256-SHA384,ECDHE-ECDSA-AES128-SHA256,ECDHE-RSA-AES128-SHA256 > >> cert_file = /etc/letsencrypt/live/specialdomain.com/fullchain.pem > >> priv_key_file = /etc/letsencrypt/live/specialdomain.com/privkey.pem > > > > Thanks, it still says > > > > > > SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> > ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:54937 > > I guess I should have been more clear before - with the above settings > TLS works for other phones, I hadn't tried with Wave. > > I downloaded Wave for iOS and played around a bit and stumbled on a > working configuration. Wave seems to only support TLS 1.0 which is > problematic itself but it is what it is. > > I set up Asterisk 16 on a VM in AWS to test which you can try as well if > you like: > > Domain: sip.seanbright.com > Username: asterisk > Password: asterisk > > Calls are SRTP if offered, and the number dialed just needs to be 1 or > more digits. This is the configuration I ended up with: > > [transport-tls] > type = transport > protocol = tls > method= tlsv1 > cert_file = /etc/letsencrypt/live/sip.seanbright.com/fullchain.pem > priv_key_file = /etc/letsencrypt/live/sip.seanbright.com/privkey.pem > bind = 0.0.0.0:5061 > external_media_address = 52.91.86.158 > external_signaling_address = 52.91.86.158 Ok, I created a new certificate and it still doesn't work with your transport. Is Centos 7 too old to run asterisk on? Is the android device I'm using too old? Why did it work before changing from SIP to PJSIP? Do I need to do anything special when creating the certificate? -- _ -- 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 and Grandstream Wave with TSL and SRTP
On Friday, January 24, 2020 6:25:48 PM CET Sean Bright wrote: > On 1/23/2020 6:04 PM, hw wrote: > >> This is what mine looks like which works just fine: > >> > >> [transport-tls] > >> type = transport > >> protocol = tls > >> method= tlsv1_2 > >> cipher= > >> ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES > >> 128 > >> -GCM-SHA256,ECDHE-RSA-AES128-GCM-SHA256,ECDHE-ECDSA-AES256-SHA384,ECDHE- > >> RSA- AES256-SHA384,ECDHE-ECDSA-AES128-SHA256,ECDHE-RSA-AES128-SHA256 > >> cert_file = /etc/letsencrypt/live/specialdomain.com/fullchain.pem > >> priv_key_file = /etc/letsencrypt/live/specialdomain.com/privkey.pem > > > > Thanks, it still says > > > > > > SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> > ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:54937 > > I guess I should have been more clear before - with the above settings > TLS works for other phones, I hadn't tried with Wave. > > I downloaded Wave for iOS and played around a bit and stumbled on a > working configuration. Wave seems to only support TLS 1.0 which is > problematic itself but it is what it is. > > I set up Asterisk 16 on a VM in AWS to test which you can try as well if > you like: > > Domain: sip.seanbright.com > Username: asterisk > Password: asterisk > > Calls are SRTP if offered, and the number dialed just needs to be 1 or > more digits. This is the configuration I ended up with: > > [transport-tls] > type = transport > protocol = tls > method= tlsv1 > cert_file = /etc/letsencrypt/live/sip.seanbright.com/fullchain.pem > priv_key_file = /etc/letsencrypt/live/sip.seanbright.com/privkey.pem > bind = 0.0.0.0:5061 > external_media_address = 52.91.86.158 > external_signaling_address = 52.91.86.158 Thanks a lot! I tried to register and it worked. It still doesn't work here with tlsv1. Then I noticed that you have priv_key_file set. I don't have that, and I don't remember which of the files that were created when I tried to create the key asterisk is using now is the private key. It seems I'll have to spend another day or so on all the horrible key creation stuff again. -- _ -- 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 and Grandstream Wave with TSL and SRTP
On Thursday, January 23, 2020 11:31:46 PM CET Sean Bright wrote: > On 1/21/2020 9:18 PM, hw wrote: > > [transport-tls] > > type = transport > > protocol = tls > > bind = 0.0.0.0:5061 > > tos = cs5 > > cert_file = /etc/asterisk/cert/asterisk.pem > > ca_list_file = /etc/pki/tls/certs/ca-bundle.crt > > method = sslv23 > > This is what mine looks like which works just fine: > > [transport-tls] > type = transport > protocol = tls > method= tlsv1_2 > cipher= > ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES128 > -GCM-SHA256,ECDHE-RSA-AES128-GCM-SHA256,ECDHE-ECDSA-AES256-SHA384,ECDHE-RSA- > AES256-SHA384,ECDHE-ECDSA-AES128-SHA256,ECDHE-RSA-AES128-SHA256 > cert_file = /etc/letsencrypt/live/specialdomain.com/fullchain.pem > priv_key_file = /etc/letsencrypt/live/specialdomain.com/privkey.pem Thanks, it still says SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> len: 0 peer: 10.10.20.29:54937 Why does it even say ssl3 despite tlsv1_2 is set? Is there a way to see which cipher(s) a client is trying to use? -- _ -- 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 and Grandstream Wave with TSL and SRTP
On Wednesday, January 22, 2020 3:18:23 AM CET hw wrote: > Hi, > > after switching from chan_sip to chan_pjsip, a device running Grandstream > Wave leads to the following error message on the asterisk console: > > > SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:43357 > > > Something with the encryption must have changed with asterisk. How can I > get the device to register again? Linphone doesn't register either, giving the same error message. So this must have to do with something with asterisk. Any ideas? -- _ -- 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] PJSIP and Grandstream Wave with TSL and SRTP
Hi, after switching from chan_sip to chan_pjsip, a device running Grandstream Wave leads to the following error message on the asterisk console: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> len: 0 peer: 10.10.20.29:43357 Something with the encryption must have changed with asterisk. How can I get the device to register again? [transport-tls] type = transport protocol = tls bind = 0.0.0.0:5061 tos = cs5 cert_file = /etc/asterisk/cert/asterisk.pem ca_list_file = /etc/pki/tls/certs/ca-bundle.crt method = sslv23 'method = tlsv1' doesn't work, either. -- _ -- 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] SRTP unprotect failed ...
On Thursday, January 16, 2020 4:44:23 PM CET Joshua C. Colp wrote: > On Thu, Jan 16, 2020 at 11:35 AM hw wrote: > > On Tuesday, January 14, 2020 5:29:04 PM CET hw wrote: > > > Hi, > > > > > > I'm getting messages like > > > > > > > > > res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay > > > > check > > > > > failed (index too old), retrying == SRTP unprotect failed on SSRC > > > > 576693764 > > > > > because of authentication failure 10 == SRTP unprotect failed on SSRC > > > 576693764 because of authentication failure 160 [...] > > > > > > > > > ... after a couple minutes during voice calls after which the connection > > > > is > > > > > being aborted. It seems that not all connections are affected but only > > > a > > > few, though that would need further verification. > > > > > > Is this a bug, or due to a bad internet connection (maybe packet loss?)? > > > Can I do something to look into this? > > > > > > asterisk -V > > > Asterisk 16.4.0 > > > > I've upgraded asterisk to version 17 from git, and the problem remains. > > What is the remote endpoint? That's the VOIP provider. I've contacted their support and am still waiting for an answer. > The message itself is occurring because we are receiving encrypted traffic > and failing to decrypt. I've seen this in recent times but it's been > because of the remote endpoint and not Asterisk itself. Since it's > encrypted and visibility into things isn't great, it's hard to point to > precisely what is going on though. Thanks, that's what I've been thinking after all the testing and after seeing a comment to that end in the source. Perhaps encryption is finally becoming more widespread and brings about issues which haven't been noticeable before. -- _ -- 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] SRTP unprotect failed ...
On Tuesday, January 14, 2020 5:29:04 PM CET hw wrote: > Hi, > > I'm getting messages like > > > res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check > failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 > because of authentication failure 10 == SRTP unprotect failed on SSRC > 576693764 because of authentication failure 160 [...] > > > ... after a couple minutes during voice calls after which the connection is > being aborted. It seems that not all connections are affected but only a > few, though that would need further verification. > > Is this a bug, or due to a bad internet connection (maybe packet loss?)? > Can I do something to look into this? > > asterisk -V > Asterisk 16.4.0 I've upgraded asterisk to version 17 from git, and the problem remains. Any ideas? -- _ -- 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] SRTP unprotect failed ...
Hi, I'm getting messages like res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 because of authentication failure 10 == SRTP unprotect failed on SSRC 576693764 because of authentication failure 160 [...] ... after a couple minutes during voice calls after which the connection is being aborted. It seems that not all connections are affected but only a few, though that would need further verification. Is this a bug, or due to a bad internet connection (maybe packet loss?)? Can I do something to look into this? asterisk -V Asterisk 16.4.0 -- _ -- 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] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/6/19 7:23 PM, Michael Maier wrote: On 06.07.19 at 12:16 hwilmer wrote: On 7/6/19 10:40 AM, Michael Maier wrote: On 05.07.19 at 22:02 hw wrote: openssl verify -CAfile ca.pem asterisk.pem asterisk.pem: OK When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers to the SIP provider and there is no error message). Otherwise I'm getting the error message and asterisk does not register. Reading the comments in sip.conf.sample, I would assume that asterisk can not verify the certificate of the SIP provider. Yet openssl s_client -connect secure.sip.easybell.de:5061 I'm using easybell via tls, too - but with pjsip - I had never any problem. Yes, easybell works fine, and their support is great. But don't tell anyone or they might be overwhelmed with customers fleeing the bad support of other providers ... Is there an advantage to using pjsip? What's needed for easybell with pjsip? You know that you don't need an own certificate to connect via tls to the ISP? No, I didn't know that. However, there are local clients connecting to asterisk using encryption, so I suppose my own certificate is required. That's true - but why do you need encryption on your own LAN? Just for fun or are there any particular requirements? I consider it a requirement for when employees end up using their mobile phones over foreign wireless networks, which is something that would be virtually impossible to prevent should the asterisk server be made reachable from the outside. And before that, why shouldn't phone calls always be encrypted for just in case? They are always genuinely private, and it doesn't hurt anything. Setting 'tlscapath' to /etc/pki or to /etc/pki/ca-trust/source/ didn't seem to I'm sorry - I don't know how to handle ca bundles with chan_sip. With pjsip it's ca_list_file=/etc/pki/tls/certs/ca-bundle.crt > in pjsip.transports.conf. Thanks, setting 'tlscafile=/etc/pki/tls/certs/ca-bundle.crt' seems to do the trick. However: First I set 'tlsdontverifyserver=no' and issued a 'sip reload'. There was no error message. I found that suspicious and restarted asterisk, and the error message came back. Only then I added 'tlscafile=/etc/pki/tls/certs/ca-bundle.crt' (which was unset before), and after a 'sip reload', the error message was gone. So far, it hasn't come back even when restarting asterisk. This shows that 'sip reload' doesn't really do a reload in that a certificate which hasn't been verified continues to be accepted after the configuration changed to now require verifying the certificate. This might be a security problem, and if not, it is certainly good for surprises and can create much confusion. Is it supposed to be like this, or should I make a bug report? -- _ -- 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] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/5/19 9:32 PM, John Runyon wrote: On Fri, 5 Jul 2019 at 14:28, hw mailto:h...@gc-24.de>> wrote: I thought about that and checked the configuration I've been using to create the certificate, and I can't see anywhere that it would expire earlier than after 3650 days. Is there another way to check this? openssl verify -CAfile ca.crt server.crt openssl verify -CAfile ca.pem asterisk.pem asterisk.pem: OK When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers to the SIP provider and there is no error message). Otherwise I'm getting the error message and asterisk does not register. Reading the comments in sip.conf.sample, I would assume that asterisk can not verify the certificate of the SIP provider. Yet openssl s_client -connect secure.sip.easybell.de:5061 seems to verify the certificate just fine. Previous tests seemed to show the asterisk is trying to verify its own certificate instead, or as well. What exactly is asterisk trying to verify, and what fails the verification? Suspicious is this: [Jul 5 12:48:00] NOTICE[7015]: chan_sip.c:30416 sip_poke_noanswer: Peer 'aaa' is now UNREACHABLE! Last qualify: 55 == TLS/SSL ECDH initialized (automatic), faster PFS ciphers enabled == TLS/SSL certificate ok [Jul 5 12:48:08] ERROR[1482]: tcptls.c:173 handle_tcptls_connection: Certificate did not verify: unable to get local issuer certificate That's the point at which the certificate suddenly stopped working after the SIP provider became unreachable. Why? -- _ -- 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] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/5/19 9:32 PM, John Runyon wrote: On Fri, 5 Jul 2019 at 14:28, hw mailto:h...@gc-24.de>> wrote: I thought about that and checked the configuration I've been using to create the certificate, and I can't see anywhere that it would expire earlier than after 3650 days. Is there another way to check this? openssl verify -CAfile ca.crt server.crt Which certificate is the one that can not be verified: the one I created or the one used by the SIP provider? How can I find out which certificate the error message is referring to? What is the error message? tcptls.c:173 handle_tcptls_connection: Certificate did not verify: unable to get local issuer certificate So the local issuer certificate must have somehow vanished after a few hours. -- _ -- 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] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/5/19 9:22 PM, Steve Murphy wrote: hw-- I see this kind of behavior when the certificate expires... you've probably checked this, but sometimes we miss little details like that. I thought about that and checked the configuration I've been using to create the certificate, and I can't see anywhere that it would expire earlier than after 3650 days. Is there another way to check this? Which certificate is the one that can not be verified: the one I created or the one used by the SIP provider? How can I find out which certificate the error message is referring to? -- _ -- 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] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/5/19 10:50 AM, Doug Lytle wrote: On 7/4/19 6:40 PM, hw wrote: This has again, and for no reason, ceased to work again after restarting asterisk. No matter what I try, I can't create a certificate asterisk would verify. Have you considered using LetsEncrypt for a valid certificate? Doug What would be the point in making this even more complicated? Today all of a sudden the certificate couldn't be verified anymore even without restarting asterisk. How is it possible that a certificate which was fine for 10 hours and 18 minutes suddenly can not be used anymore? -- _ -- 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] unsolved: Re: solved: how to create a working certificate for using TLS?
On 6/27/19 12:11 PM, hwilmer wrote: On 6/26/19 1:33 PM, hwilmer wrote: Hi, how can I create a self-signed certificate for asterisk which actually works? follow this guide: https://fabianlee.org/2018/02/17/ubuntu-creating-a-trusted-ca-and-san-certificate-using-openssl-on-ubuntu/ This has again, and for no reason, ceased to work again after restarting asterisk. No matter what I try, I can't create a certificate asterisk would verify. Is this a bug in 16.4, or how can I create a certificate that doesn't stop working randomly? -- _ -- 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] configure SRTP port range?
On 2/23/19 5:39 PM, Joshua C. Colp wrote:
On Sat, Feb 23, 2019, at 12:17 PM, hw wrote:
Any source to UDP ports X to Y (1 to 2 by default) allow.
Are you saying that the ports specified in rtp.conf ('rtpstart' and
'rtpend') specify with ports asterisk uses regardless whether RTP or
SRTP is being used? Is that why you speak of "media" (ports)?
(That would have been and would answer my original question: Where to
specify the SRTP ports?)
Yes.
Cool :)
Maybe a hint like "these ports are used for SRTP as well" in the default
rtp.conf would clarify this. (I was actually looking for an srtp.conf
to begin with ...)
What you can't do is limit the rule based on the source of media, except for
circumstances where you know for sure the source.
Note that RTP ports in Asterisk aren't open all the time and only listen when a
call is using it, and they also learn the source of media - blocking out other
sources.
ok
After opening the ports specified in rtp.conf, both RTP and SRTP were
working in the test calls I made. But:
How do clients know which media ports to use? Is asterisk telling them
that?
I. e., can I (basically) rely on the clients to use the media ports in
rtp.conf, or did I just get lucky that by chance the clients happened to
use these ports when I made the test calls?
It's exchanged as part of call setup using SDP. SDP specifies where media
should be sent, the codecs that can be used, and also controls hold/unhold.
Each side provides SDP which is parsed, interpreted, negotiated, and used.
Thank you very much! So I got this to work; next step would be to try
it with clients from outside the local network ... :)
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Re: [asterisk-users] configure SRTP port range?
On 2/23/19 4:19 PM, Joshua C. Colp wrote:
On Sat, Feb 23, 2019, at 11:04 AM, hw wrote:
directmedia is not explicitly enabled; I guess it's the default.
Joshua basically says there is no way to control which ports are being
used for SRTP because that it is "up the endpoint". Such endpoints, in
this case, are mobile phones with software like csipsimple or gs-wave
(or perhaps zoiper), and I see no way in these programs to define which
ports to use for SRTP.
Since I have no way to define which ports endpoints use for SRTP, I
would have to open all UDP ports in the firewall, and I don't want to do
that.
Nat is currently not involved yet. I want to get this to work first and
then look into nat issues.
Only open a range of ports that you really use: for example is you have
maximum 10 simultaneous calls, open only 40 ports (4 ports for each
call, two for RTP and two for RTCP). Then change rtp.conf configuration
reflect the range of ports you using.
So how would I control which ports are being used for SRTP? Some ports
being open on the firewall doesn't mean the phones will automagically
use them, does it?
I think there's confusion over ports. In calls there's two ports and IP
addresses in play. There is the IP address and port that Asterisk listens on
and sends media from. There is also the IP address and port that the endpoint
listens on and sends media from. You can control the Asterisk one as mentioned
using rtp.conf. Therefore the firewall rule for where Asterisk is running would
be:
The confusion probably comes from the canreinvite option which I had
been reading decides whether two clients communicate directly with each
other or have to go via the asterisk server. Today I found that this is
not true --- so that documentation must have been wrong.
It has created confusion because both 'canreinvite=NO' and
'canreinvite=yes' had been working. Today I found that 'directmedia=no'
did not work regardless whether RTP or SRTP was used. That was to be
expected because the firewall didn't have the RTP ports open, either. I
had already been wondering about this because I thought there would have
to be ports open for 'canreinvite=NO' to work.
Any source to UDP ports X to Y (1 to 2 by default) allow.
Are you saying that the ports specified in rtp.conf ('rtpstart' and
'rtpend') specify with ports asterisk uses regardless whether RTP or
SRTP is being used? Is that why you speak of "media" (ports)?
(That would have been and would answer my original question: Where to
specify the SRTP ports?)
What you can't do is limit the rule based on the source of media, except for
circumstances where you know for sure the source.
Note that RTP ports in Asterisk aren't open all the time and only listen when a
call is using it, and they also learn the source of media - blocking out other
sources.
ok
After opening the ports specified in rtp.conf, both RTP and SRTP were
working in the test calls I made. But:
How do clients know which media ports to use? Is asterisk telling them
that?
I. e., can I (basically) rely on the clients to use the media ports in
rtp.conf, or did I just get lucky that by chance the clients happened to
use these ports when I made the test calls?
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Re: [asterisk-users] configure SRTP port range?
On 2/23/19 2:39 PM, Social Boh wrote: *DIrect media with SRTP is not supported. All media when SRTP goes through Asterisk.* So you have to open ports on your firewall and disable directmedia=yes on your configuration. directmedia is not explicitly enabled; I guess it's the default. Joshua basically says there is no way to control which ports are being used for SRTP because that it is "up the endpoint". Such endpoints, in this case, are mobile phones with software like csipsimple or gs-wave (or perhaps zoiper), and I see no way in these programs to define which ports to use for SRTP. Since I have no way to define which ports endpoints use for SRTP, I would have to open all UDP ports in the firewall, and I don't want to do that. Nat is currently not involved yet. I want to get this to work first and then look into nat issues. Only open a range of ports that you really use: for example is you have maximum 10 simultaneous calls, open only 40 ports (4 ports for each call, two for RTP and two for RTCP). Then change rtp.conf configuration reflect the range of ports you using. So how would I control which ports are being used for SRTP? Some ports being open on the firewall doesn't mean the phones will automagically use them, does it? Other option is using another PBX/SWITCH that support SRTP flow direct between endpoints. Which one does that? And does that work through foreign firewalls I have no control over and when NAT becomes involved? -- _ -- 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] configure SRTP port range?
On 2/23/19 1:15 PM, Joshua C. Colp wrote: On Sat, Feb 23, 2019, at 8:06 AM, hw wrote: On 2/22/19 7:56 PM, Joshua C. Colp wrote: On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. There is no separate port range used for SRTP, and Asterisk does not control the port that the phone uses for sending to Asterisk. That's up to the endpoint. Thanks! The phones do not have any settings with which I could limit the ports used for SRTP. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? DIrect media with SRTP is not supported. All media when SRTP goes through Asterisk. Well, how are we supposed to handle this in firewalls? I do not want to open all ports for UDP traffic directed to the server. It's expected that traffic to the RTP port range that Asterisk is configured to use is let through to Asterisk to ensure audio flow. The phones don't seem to be using the RTP port range specified in rtp.conf when they are using SRTP. When they are using RTP, they do not send the RTP traffic via asterisk, though they can do that without the ports for this opened in the firewall (perhaps the router uses a conntrack helper for RTP; I'd have to find out). When the phones use SRTP, the ports they're using are all over the place. I'd either have to open all UDP ports for their traffic to go via the server or stick to unencrypted phone calls. There must be some solution for this. That phone calls are encrypted schould be the default, especially since they are all going over the internet nowadays. -- _ -- 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] SRTP with accounts in mysql database
On 2/22/19 6:12 PM, Antony Stone wrote: On Friday 22 February 2019 at 18:05:26, hw wrote: Hi, the ecnryption tutorial[1] says to add 'encryption=yes' into sip.conf for a peer to use SRTP. I have all the account information in a mysql database in a table called `sippeers` asterisk uses. The table doesn't seem to have a column for this option. How can I specify it; where in the database do I put it? Can I just add a column `ecryption` and put 'yes' (or no) into it? Yes - so long as you spell it correctly :) Thanks, it seems to work :) -- _ -- 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] configure SRTP port range?
On 2/22/19 7:56 PM, Joshua C. Colp wrote: On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. There is no separate port range used for SRTP, and Asterisk does not control the port that the phone uses for sending to Asterisk. That's up to the endpoint. Thanks! The phones do not have any settings with which I could limit the ports used for SRTP. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? DIrect media with SRTP is not supported. All media when SRTP goes through Asterisk. Well, how are we supposed to handle this in firewalls? I do not want to open all ports for UDP traffic directed to the server. -- _ -- 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] configure SRTP port range?
Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? That the traffic is being blocked by the firewall is probably the reason why I have no audio when using SRTP ... -- _ -- 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] SRTP with accounts in mysql database
Hi, the ecnryption tutorial[1] says to add 'encryption=yes' into sip.conf for a peer to use SRTP. I have all the account information in a mysql database in a table called `sippeers` asterisk uses. The table doesn't seem to have a column for this option. How can I specify it; where in the database do I put it? Can I just add a column `ecryption` and put 'yes' (or no) into it? [1]: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial -- _ -- 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] Fwd: Ihnen steht ein Kupon bei PIOSPARTSLAP IT Remarketing e.K zur Verfügung
On 2/5/19 3:55 PM, hw wrote: Derzeit interessante Angebote habe ich dort nicht entdeckt. Sorry, this wasn't supposed to go to the list! -- _ -- 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] Fwd: Ihnen steht ein Kupon bei PIOSPARTSLAP IT Remarketing e.K zur Verfügung
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Re: [asterisk-users] how to use a database
On 12/07/2018 04:14 PM, Administrator TOOTAI wrote: Le 07/12/2018 à 15:56, hw a écrit : On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: Le 07/12/2018 à 14:32, hw a écrit : [...] Queues seem to be the only way to have several phones ring at once, or are there other ways? Dial(SIP/Phone1/Phone2&.../Phonex,,) Good to know, thanks! What are the entries needed in the queue_members table when using odbc? Alembic made the primary key so that each queue can only have one entry (What is an interface here?), and there's probably a reason for that. How do you enter several members for a queue? Asterisk seems to either rather crash than to create a queue, or to do nothing. Why you don't just add members dynamically in a queu using AddQueueMember/RemoveQueueMember or even with pause/unpause members ? So far, there's only one queue, and it's members are always the same. With dynamic queue members, how do you solve the problem of automatically recreating queues when restarting asterisk? BTW the above dial string has nothing to do with queue, it just a cmd that rings all phones at once. Yes --- I was looking for a way to do that, and the only way I found was using a queue. I have two cases in one of which a queue is just right while ringing several phones at once and not having a queue would be better in the other. -- _ -- 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] cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.
On 12/07/2018 03:08 PM, Joshua C. Colp wrote: On Fri, Dec 7, 2018, at 9:54 AM, hw wrote: On 12/07/2018 02:25 PM, Joshua C. Colp wrote: On Fri, Dec 7, 2018, at 9:19 AM, hw wrote: Hi, is cdr logging using odbc buggy? I'im only getting an error "cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.". Connecting with isql to the datasource given in cdr_odbc.conf works just fine, and using the database for sippeers also works. This message is output when the "dsn" value provided in cdr_odbc.conf does not match a dsn/class (context name) configured in res_odbc.conf You should confirm they match and if still encountering a problem then provide the configuration. Thanks, it's working now. I've been using the name of the data source rather than the name of the section. It doesn't make any sense to call it data source name (dsn) at places where the name of a section is expected. The cdr_odbc module originates from 15 years ago, so it's unsurprising. Is it not being used much? It also doesn't make sense that the section name in res_odbc.conf should be relevant here. Resources (res) would be configuration information --- the tables for this are created with alembic --- while logging information is something else --- and the tables for it are not created with alembic. The resource module implements the interface to ODBC and provides/manages the connections. Other modules are consumers of it and thus reference it. Then why doesn't alembic create the tables for logging as well as the ones for configuration information? This could use a lot of cleanup and (or at least) much better documentation. If you'd like to contribute documentation improvements they follow the same process as everything[1]. If there's wiki pages that could use improvement just leave a comment and after a few you'll be granted access to edit things. [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process I would have improved a wiki page yesterday if there had been a way to do that. Maybe if there were parts of the wiki for users to easily make comments, more users would contribute. Who has the time and is willing to go through a lengthy contribution process which involves consulting lawyers to figure out if it is advisable to sign the license agreement after verifying if the agreement is even applicable in your country? Finding lawyers knowledgable in international copyright laws is a task in itself, and they're probably rather expensive ... After that, you need to address privacy concerns that could be involved as well ... I'm certainly not going to give my full name and address etc. just to improve a wiki page. It's useless anyway because there's no way to verify if the information is true. -- _ -- 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 use a database
On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: Le 07/12/2018 à 14:32, hw a écrit : [...] Queues seem to be the only way to have several phones ring at once, or are there other ways? Dial(SIP/Phone1/Phone2&.../Phonex,,) Good to know, thanks! What are the entries needed in the queue_members table when using odbc? Alembic made the primary key so that each queue can only have one entry (What is an interface here?), and there's probably a reason for that. How do you enter several members for a queue? Asterisk seems to either rather crash than to create a queue, or to do nothing. -- _ -- 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 use a database
On 12/07/2018 02:45 PM, Marcelo Terres wrote: https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic alembic did not create any tables for logging. -- _ -- 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] cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.
On 12/07/2018 02:25 PM, Joshua C. Colp wrote: On Fri, Dec 7, 2018, at 9:19 AM, hw wrote: Hi, is cdr logging using odbc buggy? I'im only getting an error "cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.". Connecting with isql to the datasource given in cdr_odbc.conf works just fine, and using the database for sippeers also works. This message is output when the "dsn" value provided in cdr_odbc.conf does not match a dsn/class (context name) configured in res_odbc.conf You should confirm they match and if still encountering a problem then provide the configuration. Thanks, it's working now. I've been using the name of the data source rather than the name of the section. It doesn't make any sense to call it data source name (dsn) at places where the name of a section is expected. It also doesn't make sense that the section name in res_odbc.conf should be relevant here. Resources (res) would be configuration information --- the tables for this are created with alembic --- while logging information is something else --- and the tables for it are not created with alembic. This could use a lot of cleanup and (or at least) much better documentation. -- _ -- 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 use a database
On 12/06/2018 10:26 PM, Marcelo Terres wrote: The Asterisk source has a tool to create the db Which one is that? -- _ -- 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 use a database
On 12/06/2018 08:43 PM, Antony Stone wrote: On Thursday 06 December 2018 at 17:49:25, hw wrote: How dynamic are changes made in the database? If by "dynamic" you mean "quickly used" then the answer is "immediately". There's a note in some configuration file saying that dynamic extensions are deprecated and suggesting to use func_odbc instead. This func_odbc seems to be the most awkward way anyone could think of for this, though. I use func_odbc in plenty of situations, but I'm not familiar with it being recommended for managing queues. Did I say anything about using it for queues? Queues seem to be the only way to have several phones ring at once, or are there other ways? Without seeing the "note in some configuration file" that you refer to, though, I don't know what to say about this. It says "However, ; note that using dynamic realtime extensions is not recommended anymore as a ; best practice; instead, you should consider writing a static dialplan with ; proper data abstraction via a tool like func_odbc." in extconfig.conf. For example, if I want to have an extension 'foobar' and want to ring different devices depending on some factors (like time of day, for example), can I modify the entry in the database for the device to ring from 'bar' to 'baz', and baz will ring instead of bar from thereon? Yes. And IIUC the extension would use something like 'Dial(SIP/ODBC_PICK_USER(...))' after defining a query for my ..._PICK_USER function in func_odbc.conf to return what to dial depending on the argument(s) supplied? No comment; I don't use this feature myself. Which feature? How do I make asterisk reload func_odbc.conf? Or is that not needed? Not needed. The whole point of configs in database tables is that they take effect immediately without having to tell Asterisk to reload anything. good We did start off just talking about getting queue_log into a database table, though. That's why I changed the subject. Now I started with CDR logging to the database because queue logging seems to be even more difficult, and it's not working because asterisk says "cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed." while I can connect to the datasource given in cdr_odbc.conf with isql just fine. I made another post about that, though. -- _ -- 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] cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.
Hi, is cdr logging using odbc buggy? I'im only getting an error "cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.". Connecting with isql to the datasource given in cdr_odbc.conf works just fine, and using the database for sippeers also works. The documentation[1] is confusing because it says freeTDS is required and that you must not use multiple database connectors and remains entirely unclear about whether odbc works at all for this and doesn't say what to do when you use odbc for sippeers and want to log CDRs in your database. [1]: https://wiki.asterisk.org/wiki/display/AST/MSSQL+CDR+Backend -- _ -- 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 use a database
On 12/05/2018 05:00 PM, Antony Stone wrote: On Wednesday 05 December 2018 at 15:31:38, hw wrote: I don't see a table for that. You need to create that for yourself. Asterisk can write to database tables, but doesn't help you set them up, for reasons I can't comment on. How do I know what the schema needs to be? Does anybody have a scheme for the queue_log table (and maybe others)? Do I get to see the queries that are being used to write this data, or do I need to form them myself and enter them into some configuration file? How dynamic are changes made in the database? If by "dynamic" you mean "quickly used" then the answer is "immediately". There's a note in some configuration file saying that dynamic extensions are deprecated and suggesting to use func_odbc instead. This func_odbc seems to be the most awkward way anyone could think of for this, though. For example, if I want to have an extension 'foobar' and want to ring different devices depending on some factors (like time of day, for example), can I modify the entry in the database for the device to ring from 'bar' to 'baz', and baz will ring instead of bar from thereon? Yes. And IIUC the extension would use something like 'Dial(SIP/ODBC_PICK_USER(...))' after defining a query for my ..._PICK_USER function in func_odbc.conf to return what to dial depending on the argument(s) supplied? How do I make asterisk reload func_odbc.conf? Or is that not needed? Is it possible to use configuration from both the database and the files at the same time? Yes. So far, that seems to work fine :) -- _ -- 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] how to use a database (was: figuring out what happened to a call)
On 12/05/2018 01:19 PM, Antony Stone wrote: On Wednesday 05 December 2018 at 13:04:57, hw wrote: On 12/04/2018 07:07 PM, Antony Stone wrote: On Tuesday 04 December 2018 at 16:11:39, hw wrote: On 12/01/2018 05:30 PM, Marcelo Terres wrote: Queue_log Thanks! That's not really it; however, how do I make it so that asterisk writes this information right away into a mariadb database instead of into a file so that I could actually use it? Send your queue_log entries to odbc? odbc? Seriously? Yes, it's the preferred method of talking to databases from Asterisk. After reading some documentation, anything else but odbc appears to be more or less deprecated :/ If you want to use the MySQL-specific driver / connector, you can still use that for some things, but Voicemail in a database can only be done via ODBC, for example. It's a setting in extconfig.conf. Does mysql not work? It's mentioned there, too. By all means try it - if it's mentioned, it'll probably work, but ODBC is the more generic and better-supported way of using databases with Asterisk. Since it really seems to be the most reasonable choice, I've set up an odbc connection and used alembic to create tables in a database for asterisk. Now how is this managable? Is there a tool that reads the files I have and enters the configuration into the database? And when changes are to be made, editing configuration files is tremendously easier than going through the tables in the database and try to make the changes there. For now, can I make it so that only the queue_log is written into the database? I don't see a table for that. How dynamic are changes made in the database? For example, if I want to have an extension 'foobar' and want to ring different devices depending on some factors (like time of day, for example), can I modify the entry in the database for the device to ring from 'bar' to 'baz', and baz will ring instead of bar from thereon? That seems to be what this is intended for; in any case, it's what I'm going to need, which is why I went to all these lengths to connect to a database. Is it possible to use configuration from both the database and the files at the same time? That would save me converting all the entries in the files for now. -- _ -- 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] figuring out what happened to a call
On 12/04/2018 07:07 PM, Antony Stone wrote: On Tuesday 04 December 2018 at 16:11:39, hw wrote: On 12/01/2018 05:30 PM, Marcelo Terres wrote: Queue_log Thanks! That's not really it; however, how do I make it so that asterisk writes this information right away into a mariadb database instead of into a file so that I could actually use it? Send your queue_log entries to odbc? odbc? Seriously? It's a setting in extconfig.conf. Does mysql not work? It's mentioned there, too. -- _ -- 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] figuring out what happened to a call
On 12/01/2018 05:30 PM, Marcelo Terres wrote: Queue_log Thanks! That's not really it; however, how do I make it so that asterisk writes this information right away into a mariadb database instead of into a file so that I could actually use it? -- _ -- 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] figuring out what happened to a call
Hi, how can I figure out what happens to inbound calls? The inbound calls I'm particularly interested in make phones that are members of a queue ring; when the call isn't picked up, another phone is dialed and when the call still isn't picked up, asterisk hangs up. I want to know the following: + Who's calling? + What did the caller dial? + Is an inbound call being picked up or not? + Which phone picks it up? + Which of the phones that could be rung for the call are busy so that they can not be used to pick up the call? + How long has a call been going on for (for both the ones that were picked up and the ones that weren't)? I could only figure out who is calling and might be able to figure out what the caller dialed. There seems to be no way to tell how a call is being dealt with, though. -- _ -- 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 problem
Jonathan H schrieb: Well, what immediately stands out is: "FILE * open failed!" Yes, and it doesn´t say which file cannot be opened. I even looked at the source and found that at that point, you can´t simply add some debugging output to find out. Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? It says 'SSL certificate ok' when I 'reload sip'. When it can´t read one of the files involved with the certificate, it says which one. I have it working with microsip and a free TLS cert from LetsEncrypt. When I get to the PC with that on, I can write up what settings I've got if that helps? I´m using a self signed certificate, but that shouldn´t behave any differently than an externally sigend one as long as it checks out, which it apparently does. So yes, it would be nice if you could send me the settings you´re using, thanks :) On 26 August 2016 at 10:47, hw <h...@gc-24.de> wrote: hw schrieb: Hi, I´m trying to get TLS to work with asterisk and client phones, and all I´m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to connect. No client is able to register using TLS. How can I use encrypted connections? Nobody having an idea? Nobody using encryption? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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 -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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 problem
hw schrieb: Hi, I´m trying to get TLS to work with asterisk and client phones, and all I´m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to connect. No client is able to register using TLS. How can I use encrypted connections? Nobody having an idea? Nobody using encryption? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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] TLS problem
Hi, I´m trying to get TLS to work with asterisk and client phones, and all I´m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to connect. No client is able to register using TLS. How can I use encrypted connections? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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
