Re: [asterisk-users] Confbridge GUI?
> >> If you can provide details, even vague ones, about how you did it, I > >> can update the WMM package. > > > > See http://asterisk.gnat.com/meetme.tgz > > > > That's a gzipped tar of our working directory plus the relevant parts of > > extensions.conf. I xxx'ed out phone numbers and Google interface data. > > The above tarball appears to be no longer available. Sorry. That machine was moved to new hardware and I forgot that I'd put that out there. It's there again. I hope it's useful, and I'll help if I can, but it's not something I can "support". -- _ -- 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] Confbridge GUI?
> If you can provide details, even vague ones, about how you did it, I > can update the WMM package. See http://asterisk.gnat.com/meetme.tgz That's a gzipped tar of our working directory plus the relevant parts of extensions.conf. I xxx'ed out phone numbers and Google interface data. This should help. I hope it's useful. -- _ -- 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] Confbridge GUI?
> I have a very old server that is used only for conferences on > Meetme. To manage the conference rooms we use Web Meetme. Now it is > time to upgrade everything but since Meetme is no longer available I > need to find a replacement GUI to manage the conference rooms. Anyone > know a solution that works with Confbridge? It's straightforward to use web-meetme with Confbridge; we've been doing it here for years. -- _ -- 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] Odd audio issue with video conference
We're experimenting with using Asterisk (14.6.0) for video conferences. This test has three endpoints, a Polycom Trio with its video accessory, and two desktops running Linphone. The video is all H.264. We're using Opus for audio on the Linphone Windows desktops and have tried both G.722 and Siren14 on the Polycom. When we have a two-party conference, everything works fine. But when we add a third party, it gets odd. The two Linphone users can hear each other just fine and the Polycom user can hear the Linphone users fine, but when the Polycom user starts talking, all is OK for about four seconds, then it gets replaced by a hiss for the rest of the period of talking, then goes back to being OK and repeats. This is peculiar. It doesn't sound like an Asterisk bridge issue because it only happens for one particular participant, but it's hard to see how it can be a Polycom issue since it can't tell between two and three participants. Does anybody have any ideas here about what to try next to see if we can diagnose this? -- _ -- 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] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode
I've had two Asterisk crashes today that seem to be caused by errors
where chan->tech_pvt is pointing to something that can't be deallocated
and I think I see a reference count bug in the above function.
It contains:
if (data->chan_old_vsrc) {
ast_channel_unref(data->chan_old_vsrc);
}
Shouldn't this also have:
data->chan_old_vsrc = NULL;
It seems to me that if it doesn't and the next condition also isn't
true, then the next time this same code is executed, it'll decrement
the reference count of the old channel again, which is wrong since it
hasn't been decremented.
What am I missing?
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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4
> There are certain versions of the Linux kernel that have no support > under the older version of ESXI. We started having issues under our > ESXI v4 setup with RH Enterprise and vmware's response was, "It's > not supported" "not supported" and "does not work" are not the same thing. ESXI emulates specific hardware. Most kernels will work with old hardware, so they should work with old ESXI, though there may need to be some configuration changes and there's always the possibility of bugs in ESXI that weren't detected by older kernels. But the question here was *Asterisk*, not kernels. User-level code has *way* fewer dependencies. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4
> The version is licensed and the customer does not want to invest on new > hardware/software at the moment. If the ESXI version is too old I need > to give them definitive proof that the segfaults are caused by that but > since the old elastix has been running there for years they do not quite > believe it. I wouldn't believe it either. I'd be quite surprised if something won't work with any ESXI version. *Perhaps* there's a configuration issue, but I'd be surprised about that 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] Difference between Application Set and Function SET?
> It was only when I ran AsteriskLint over my dialplan that I noticed this: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET > > Hmmm, they both seem to do the same thing. Or don't they? In some sense they do, but one's an application, meaning that it's like a subprogram in a programming-language sense, and the other is a function, which returns a value. -- _ -- 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] CM for menuselect choices
> Use menuselect's command line (--enable and --disable). Great idea! How would you recommend generating the set of --enable and --disable options that differ from the default from a build that was done? -- _ -- 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] CM for menuselect choices
> Of course, you might run into problems if the later release introduces new > options (or deprecates old ones) which then aren't going to be in your > makeopts file That's my question: how do I reflect the changes that I made to the defaults in a way that's not dependent on the exact set of options that each release has? -- _ -- 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] CM for menuselect choices
I'd like to be able to save the choices made in menuselect in a way that they can be tracked in a CM system and applied to a later release of Asterisk using an automated tool like Ansible. What's the best way to 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
[asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000
I had three crashes this morning on a divide-by-zero, for example at abstract_jb.c:1008 in 14.3.0. Does this ring any bell to anybody? -- _ -- 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] More issues with Siren14 datalen == 0 packets
> The feed function in slinfactory explicitly does not allow frames > without a data payload to be added to the queue. It would have prevented > this crash. Ah, so the fix should really be there, righty? > I think the underlying issue is that the data pointer is not NULL when > it sanely should be in the codec implementation. Note that that would still leave a bug in func_speex.c, since it checks for neither case. And, of course, that's not open-source, so I can't fix it. > > Can you suggest ways of searching for other possible occurrences of > > this bug? These crashes are occurring during important conferences and > > are causing significant issues. > > Not really. Frames are used a lot across Asterisk, so you have to follow > the flow based on the features in use. I did some searches and came across one suspicious case: In funcs/app_jack.c:queue_voice_frame That's all I see. -- _ -- 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] More issues with Siren14 datalen == 0 packets
> All patches need to go into JIRA with a license agreement to be > accepted. Understood, but I was using it as an illustration. Note, however, that, from a legal perspective, a patch such as this has no protectable IP (you can't copyright the only way of doing something) and the GNU projects have a formal rule that sufficiently-small patches need no assignments for that reason, which I suggest you may want to adopt as well. > > Why is samples being used as a length instead of datalen? > > Internally a signed linear factory operates in terms of samples, not the > data payload itself. I've also commented on your original issue in > regards to the siren codecs that it should NULL out the data pointer > itself. That is more commonly used. But I don't think that it would have helped in either case, this one or in func_speex.c, because neither tests for a null data pointer either. Can you explain the difference between "datalen" and "samples" in this context, shouldn't they always be related by a (small) linear factor? Should I open a JIRA issue for this as well? Can you suggest ways of searching for other possible occurrences of this bug? These crashes are occurring during important conferences and are causing significant issues. -- _ -- 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] More issues with Siren14 datalen == 0 packets
Another crash with a packet:
$10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 324, offset = 64,
src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318,
uint32 = 2156475160, pad = "\030\063\211\200\000\000\000"}, delivery = {
tv_sec = 1492000520, tv_usec = 225198}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
Note that datalen is zero, but samples aren't.
main/slinfactory.c near line 177 doesn't check for datalen of zero,
but copies using samples.
Fixed thusly:
*** slinfactory.c.orig 2017-02-13 15:00:19.0 -0500
--- slinfactory.c 2017-04-12 08:48:16.0 -0400
***
*** 174,178
frame_data = frame_ptr->data.ptr;
! if (frame_ptr->samples <= ineed) {
memcpy(offset, frame_data, frame_ptr->samples *
sizeof(*offset));
sofar += frame_ptr->samples;
--- 174,180
frame_data = frame_ptr->data.ptr;
! if (frame_ptr->datalen == 0)
! ;
! else if (frame_ptr->samples <= ineed) {
memcpy(offset, frame_data, frame_ptr->samples *
sizeof(*offset));
sofar += frame_ptr->samples;
How many more of these cases are there going to be?
Why is samples being used as a length instead of datalen?
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Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0
> I would say this is a bug in func_speex and not in codec_siren14. This
> is because the datalen is zero.
Ah! So, like?
*** func_speex.c.orig 2017-02-13 15:00:19.0 -0500
--- func_speex.c2017-04-06 11:16:03.0 -0400
***
*** 185,189
}
! speex_preprocess(sdi->state, frame->data.ptr, NULL);
snprintf(source, sizeof(source), "%s/speex", frame->src);
if (frame->mallocd & AST_MALLOCD_SRC) {
--- 185,190
}
! if (frame->data.ptr && frame->datalen)
! speex_preprocess(sdi->state, frame->data.ptr, NULL);
snprintf(source, sizeof(source), "%s/speex", frame->src);
if (frame->mallocd & AST_MALLOCD_SRC) {
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[asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0
I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
(gdb) p *frame
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 232, offset = 64,
src = 0x2ac07413e7f8 "siren14tolin32", data = {ptr = 0x3cab9378,
uint32 = 1017877368, pad = "x\223\253<\000\000\000"}, delivery = {
tv_sec = 1491485582, tv_usec = 407272}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
frame->data.ptr is an out-of-range address.
Does this ring a bell to anybody? Without sources of the Siren14 codec,
how would you recommend we debug this?
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[asterisk-users] Asterisk crash when playing a WAV file to G722 SIP
I recently upgraded to Asterisk 14.3.0. When playing a SIP file to a G722 SIP channel (via chan_sip), I get a crash with the following traceback. This is reproducable: #0 0x0036fdc30265 in raise () from /lib64/libc.so.6 #1 0x0036fdc31d10 in abort () from /lib64/libc.so.6 #2 0x0036fdc69beb in __libc_message () from /lib64/libc.so.6 #3 0x0036fdc7174f in _int_free () from /lib64/libc.so.6 #4 0x0036fdc75a4b in free () from /lib64/libc.so.6 #5 0x0050e19e in ast_frame_free (frame=0x5c35, cache=1) at frame.c:171 #6 0x00502bac in ast_readaudio_callback (s=0x6a5df88) at file.c:921 #7 0x00502d19 in ast_fsread_audio (data=0x5c35) at file.c:952 #8 0x004bb3df in __ast_read (chan=0x7ba68f8, dropaudio=0) at channel.c:3848 #9 0x00504e51 in waitstream_core (c=0x7ba68f8, breakon=0x2b9630672bdb "", forward=0x5e56f8 "", reverse=0x5e56f8 "", skip_ms=0, audiofd=-1, cmdfd=-1, context=0x0, cb=0) at file.c:1602 #10 0x005053bf in ast_waitstream (c=0x5c35, breakon=0x5fca ) at file.c:1754 #11 0x2b963067272e in playback_exec (chan=0x7ba68f8, Does this "ring a bell" to anyone? It looks like frame chainin has gotten corrupted somehow, but this should be a straightforward case. -- _ -- 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] UniMRCP and Asterisk 14
> I can't speak for the MRCP guys, but from a difference perspective, > swapping MRCP from Asterisk 13 to Asterisk 14 shouldn't be too > difficult. Most of the changes between the two shouldn't affect most > people's use cases, including projects such as MRCP. I'd definitely > check with their discussion forums though, since it seems that they > don't monitor the asterisk-users mailing lists. I got it working. It indeed wasn't much: *** ./app-unimrcp/Makefile.in.orig 2016-07-29 19:18:09.0 -0400 --- ./app-unimrcp/Makefile.in 2017-03-26 10:43:52.0 -0400 *** *** 132,136 CC = @CC@ CCDEPMODE = @CCDEPMODE@ ! CFLAGS = @CFLAGS@ CPP = @CPP@ CPPFLAGS = @CPPFLAGS@ --- 132,136 CC = @CC@ CCDEPMODE = @CCDEPMODE@ ! CFLAGS = @CFLAGS@ -DAST_MODULE_SELF_SYM=__internal_app_swift CPP = @CPP@ CPPFLAGS = @CPPFLAGS@ *** ./app-unimrcp/app_unimrcp.c.orig2014-07-10 17:04:03.0 -0400 --- ./app-unimrcp/app_unimrcp.c 2017-03-26 10:49:12.0 -0400 *** *** 57,61 #include "ast_compat_defs.h" ! #if AST_VERSION_AT_LEAST(1,4,0) #define AST_COMPAT_STATIC static ASTERISK_FILE_VERSION(__FILE__, "$Revision: $") --- 57,64 #include "ast_compat_defs.h" ! #if AST_VERSION_AT_LEAST(14,0,0) ! ASTERISK_REGISTER_FILE() ! #define AST_COMPAT_STATIC static ! #elif AST_VERSION_AT_LEAST(1,4,0) #define AST_COMPAT_STATIC static ASTERISK_FILE_VERSION(__FILE__, "$Revision: $") *** ./res-speech-unimrcp/Makefile.in.orig 2016-07-29 19:18:09.0 -0400 --- ./res-speech-unimrcp/Makefile.in2017-03-26 10:43:08.0 -0400 *** *** 130,134 CC = @CC@ CCDEPMODE = @CCDEPMODE@ ! CFLAGS = @CFLAGS@ CPP = @CPP@ CPPFLAGS = @CPPFLAGS@ --- 130,134 CC = @CC@ CCDEPMODE = @CCDEPMODE@ ! CFLAGS = @CFLAGS@ -DAST_MODULE_SELF_SYM=__internal_res_speech_unimrcp CPP = @CPP@ CPPFLAGS = @CPPFLAGS@ *** ./res-speech-unimrcp/res_speech_unimrcp.c.orig 2014-12-10 22:37:36.0 -0500 --- ./res-speech-unimrcp/res_speech_unimrcp.c 2017-03-26 11:31:10.0 -0400 *** *** 29,33 --- 29,41 #define AST_MODULE "res_speech_unimrcp" + #if AST_VERSION_AT_LEAST(14,0,0) + ASTERISK_REGISTER_FILE() + #elif AST_VERSION_AT_LEAST(1,4,0) + #define AST_COMPAT_STATIC static ASTERISK_FILE_VERSION(__FILE__, "$Revision: $") + #else /* 1.2 */ + #define AST_MODULE_LOAD_DECLINE -1 + #define AST_COMPAT_STATIC + #endif #include -- _ -- 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] UniMRCP and Asterisk 14
When I look at the lastest UniMRCP manual, they only mention as high as Asterisk 13. Does anybody know if I need to do anything to allow it to work on Asterisk 14 and, if so, what that is? -- _ -- 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 records and conferences
At least in version 12.2.0, the code in cdr.c appears to create CDR records for each pair of users in a conference. This is quadratic and would seem to be an issue with large conferences. I got two Asterisk crashes when a lot of people tried to dial into a conference. They appear quite related to https://issues.asterisk.org/jira/browse/ASTERISK-24758 There are thousands of CDR entries on the chain. I believe this is due to the quadratic behavior above. Not all of the "last" fields point to the same entry, which is peculiar, though. I think that my crash was caused by a stack overflow in recursive calls to delete the huge CDR chain (over 7,000 entries). Why are all these CDR entries made? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siren7 for Asterisk 13.5
A Siren codec is not currently available and the one for 12 will not work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siren7 for Asterisk 13.5
Alas, until we get off our butts, yes. Sorry about that. Really, we're putting as much effort into fixing things and issues that affect a lot of people. While siren7/siren14/silk are nice, there aren't as many people using them as other affected things at this moment. Is there something nontrivial that needs to be done here other than just recompiling/linking? If so, then I'm likely to run into it as well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siren7 for Asterisk 13.5
What is the proper version of the Siren7 codec to use for Asterisk 13.5.0? Since there's nothing later, does the version for 12.0 work? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siren7 and Asterisk 13
I'm planning on upgrading to Asterisk 13.4 soon and am looking for the corresponding Siren7 codec. Where do I find it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?
This is an interpolated frame from func_jitterbuffer. It's part of packet loss concealment. What scenario exposed this? We were testing for clipping by doing Set(VOLUME(RX)=100) but we were connecting to a ConfBridge that had a jitterbuffer. This occurred when the phone (SIP) hung up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
id = AST_FORMAT_SLINEAR16, fattr = {format_attr = {
0 repeats 64 times}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64,
src = 0x51623b0 func_jitterbuffer interpolation, data = {ptr = 0x0,
uint32 = 0, pad = \000\000\000\000\000\000\000}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
so datalen is 0 and samples nonzero. ast_frame_adjust_volume, however,
iterates over samples, not datalen. Is that correct?
What does it mean to have a packet with a zero datalen anyway?
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Re: [asterisk-users] setting outbound caller ID
CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] default features
Question: is there some built-in way to know if macro feature1-ClientA is defined? Something liken ExecIfMacro(feature1-ClientA)?macro(feature1-ClientA):Goto(...). A macro is a context, so DIALPLAN_EXISTS should work if you specify an extension and priority that's in the macro (presumably, s,1). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Jitterbuffer
What are the cons, if any, of enabling a jitterbuffer? Memory and latency. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting source ip adress of incoming INVITE
I'm interested in finding out what the source ip is of an invite in the
dialplan (Asterisk 11).
${CHANNEL(recvip)}
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Re: [asterisk-users] WSS over Asterisk
I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event = failed_to_start Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works fine. Any idea why? Sorry for the delay in answering: I meant to reply and forgot. ws:// uses HTTP and wss:// uses HTTPS so there's no way they can work via the same socket. You have to set up a separate HTTPS socket for wss. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
Committed the fix for this leak on Asterisk v12 branch in -r413452. This leak also applied to Asterisk v11. Thanks. Is this for both the one in the talking callback or the one in handle_cli_confbridge_kick or both (the fix is similar in both)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
I think the bug is in conf_handle_talker_cb. It calls ao2_find but has no
mechanism to decremement the refcount. It appears that the following is
the best fix. I looked at all remaining calls to ao2_find in app_confbridge.c
and they look OK. Do you agree with the below fix?
*** app_confbridge.c.bug2014-05-06 06:42:21.0 -0400
--- app_confbridge.c2014-05-06 06:42:05.0 -0400
*** static int conf_handle_talker_cb(struct
*** 1461,1467
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! struct confbridge_conference *conference = ao2_find(conference_bridges,
conf_name, OBJ_KEY);
struct ast_json *talking_extras;
if (!conference) {
/* Remove the hook since the conference does not exist. */
--- 1461,1468
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! RAII_VAR(struct confbridge_conference *, conference, NULL, ao2_cleanup);
struct ast_json *talking_extras;
+ conference = ao2_find(conference_bridges, conf_name, OBJ_KEY);
if (!conference) {
/* Remove the hook since the conference does not exist. */
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Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
That is definitely a leak and the fix looks good. Thanks. That leak is most likely the one biting you. It definitely is. There is another leak in handle_cli_confbridge_kick() if the participant to kick is not in the conference. Confirmed. I missed that one in my code reading. I just fixed it the same way. Please go ahead and open an issue so proper credit can be given for the patch. I'm not concerned about credit, but would like to get it fixed. I need to figure out what has to happen for me to be able to submit patches, but then I'll have some others to submit too. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
It may show up in 'bridge show all' - but I'd actually expect it not to show up there either. Actually, it does. I have a screen full of bridges with 0 channels. I just tried an experiment where all I have is exten = 329,1,Answer(1000) same = n,Confbridge(1234) with absolutely nothing else going on and those leak too. I need to understand why I'm seeing this and nobody else is. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm seeing this and nobody else is. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
If the reference count on the bridge is off, you should see the conference bridge 'hanging around' after the last participant has left. And how would I be sure this is the case? I did core set debug 1 and didn't see the debug line about destroying the conference, but it doesn't show up in confbridge list. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
Really, I think we're pretty positive there's a ref leak (since otherwise, the CBAnn channel would be long gone). If you can get a ref debug log and the standard Asterisk DEBUG log showing the problem, that would help a lot in finding out what is going on. That can't be done in the 12.2.0 release, just the current SVN, right? Clearly this occurs for me and not in the simple case. So I think what I'll do is see exactly what I have that's causing it and hopefully code inspection of that piece will show the missing ref decrement. I'm away for a few days and so may not be able to get to this until I get back. Thanks for the pointers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CBAnn channel not going away in Asterisk 12
After an upgrade to Asterisk 12, I'm collecting channels. When I enter and then exit a conference room, I see: -- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c -- Channel CBAnn/207-067f;2 left 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c I'd expect those channel to immediately go away, but they just stay around: asterisk*CLI core show channel CBAnn/207-067f;1 -- General -- Name: CBAnn/207-067f;1 Type: CBAnn UniqueID: 1398809161.20186 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (nothing) WriteFormat: unknown ReadFormat: unknown WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h1m3s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:07:04] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;1 asterisk*CLI core show channel CBAnn/207-067f;2 -- General -- Name: CBAnn/207-067f;2 Type: CBAnn UniqueID: 1398809161.20187 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m30s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:09:31] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
The announcer channel joins/leaves the conference as it has sounds to play. If the channel still hangs around after the conference is destroyed then there is a problem. There's a problem. ;-) But thanks for pointing to how that's supposed to be handled. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
If the channel still hangs around after the conference is destroyed then there is a problem. Am I missing something obvious: I'm looking in the confbridge_exec function. I see a conference = NULL line, but no attempt to free that structure, which is what I understand will destroy the playback channel. So where it is freed? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk-12.2.0
What distro are you building on? CentOS 5.10. Both have the libraries listed in install_prereq. Indeed it has all but 2 or 3 of those libraries (none related to uuid), but after running that script, it was still missing what it needed for uuid. Unfortunately, there's no upgrade path from CentOS 5.10 to 6.5. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk-12.2.0
e2fsprogs-devel is the package that provides uuid.h on centos 5 I tried that first and it didn't seem to. I'm pretty sure I needed uuid-dce-devel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem building Asterisk-12.2.0
When I run ./configure, it aborts with: checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) But it *is* installed: [root@asterisk asterisk-12.2.0]# yum list installed | grep uuid uuid.i386 1.5.1-3.el5 installed uuid.x86_64 1.5.1-3.el5 installed uuid-devel.i386 1.5.1-3.el5 installed uuid-devel.x86_64 1.5.1-3.el5 installed So I'm confused ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk-12.2.0
I think you need the libuuid and libuuid-devel packages. yum list available was not showing any such package. I installed a few other packages, including uuid-dce-devel and one of them did the trick, but the install-prereq script wasn't good enough. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
If you really want to do it: 1) create a wrapper to asterisk -r 2) pipe the welcome message to /dev/null 3) ??? 4) profit you didn't modify Asterisk. No you didn't, but you may neverthess have created a derived work. There are two different legal arguments you can make when two pieces of software are tightly coupled in that way: one argues that it's a derived work and the other that it's not. Copyright law when it comes to software is not simple and certainly not obvious. If you want to use a piece of Free Software in a commercial product, you need to consult an attorney. It's really that simple. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
Modifying a program you have legitimately acquired is Fair Dealing. The Law of the Land gives you the right to do that, even if the vendor restricts your exercise of that right in practice by withholding the Source Code. That is false. Modifying a program is creating a derivative work. As purchaser of a copyrighted item, you normally *do not* have that right. And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The author is a recognized expert in software IP law. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
What does violating license of Asterisk means? Does it means I won't be able to use any commercial modules or asterisk commercially? I thought it was open and anyone can change the code? Anyone *can* change the code. But it's licensed software, just like most other software. The difference is that the GPL gives you rights that you don't have for other non-open software. However, in both cases, you have to be sure that you don't violate the terms of the license. If you're unclear as to whether what you propose to do will violate the license, I'd suggest consulting an attorney: nobody on this list (or any other) should be providing you legal advice. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
Of course, any good attorney will never commit to anything. They will never say it is alright to do X, unless X is do nothing No, but a good attorney can give guidance as to likely expectations. As you say, nobody can be sure of something even if it's previously been established law, but a good attorney can point out potential pitfalls on the one hand and identify, on the other, things that are much less likely to be an issue. It's not a guarantee, but you can often get a recommendation about whether or not it's a good idea (not necessarily alright) to do something. Attorneys often have to a take a stand on these matters. If a company needs to use software that performs a specific thing and, say, only three companies provide such, but under different licensing terms, it's the job of that company's legal department to say which, if any, they can be used. Doing nothing will have a cost and risk here too because this example is talking about something that the company needs done. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording conferences with changing bitrate
I'm running 10.7.1 (yes, I know it's old, but this may be a problem in later versions too) and had a conference being recorded via: Set(CONFBRIDGE(bridge,record_conference)=yes) The bridge started out at 8KHz despite one HD device. But when the second came in (G.722), it switched to 16KHz. At that point, the recording file had the bitrate change in the middle. That seems wrong. I'd expect the bitrate of the recording channel to remain unchanged and transcoding to be used to do the recording. But it wasn't. Does this ring a bell with anybody? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jitter buffer on write side of channel
How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integration with skype
For voice, you can use SipToSis. Works flawlessly with Asterisk and the best part, it's free. :) www.mhspot.com/sts/ (site is down right now) And that's related to the problem with it: it hasn't been maintained for quite a while. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disagreements between codec_siren14 and Polycom sources
I'm answering my own email here: There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. The disagreement is in byte-swapping of the encoded stream. Once that's done, things work fine. If anybody wants a codec that can transcode between Siren14 and slin32 (which is better than Digium's codec_siren14 codec which goes to slin and slin16), let me know. I can send a file that calls the Polycom/ITU code. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding issues with siren14
Do you have transcode_via_sln set in asterisk.conf? No, but as I said in a later email, I found the problem: when computing the cost of a path, any downconvert has the same cost. So siren14 - slin - slin32 is the same cost as siren14 - slin16 - slin32 which is wrong. I fixed this by adding the magnitude of the difference in the sampling rate to the cost, but I'm not sure if that's the right solution. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disagreements between codec_siren14 and Polycom sources
There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back correctly with the Digium's siren14 codec. I know the parameters are correct because the file is the same size as that made by the Digium codec. Both sets of decoders/encoders (Digium and Polycom/ITU) are symmetric and can decode what they encode, but neither can read the encoding of the other. Is there some subtle difference between G.722.1C and Siren14? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and slin16 to slin32. So why is slin used as the intermediate instead of slin16? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with .siren14 sound files
I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14. I downloaded the codecs and now it will properly transcode to connect to other phones and play any files that are in .wav format. But when it tries to play any files with .siren14 extensions, I get complete noise coming out. Here's the negotiated SDP: v=0 o=root 1668560220 1668560220 IN IP4 207.10.184.50 s=Asterisk PBX 10.7.1 c=IN IP4 207.10.184.50 t=0 0 m=audio 16204 RTP/AVP 115 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=sendrecv If I rename away the .siren14 files, all is OK. I can't find anything related to this with a search. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Frames with invalid timing info
I'm now getting these errors: [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP even *without* any transcoding. Suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g723 transcoding
It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] clicking sound with alaw codec
I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. The outgoing SDP looks like this: v=0 o=root 1691755711 1691755711 IN IP4 205.232.38.178 s=Asterisk PBX 10.7.1 c=IN IP4 205.232.38.178 t=0 0 m=audio 11432 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv The reply SDP is: v=0 o=default 1359060187 1359060187 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 Any suggestions on how to debug what's causing this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clicking sound with alaw codec
Your sounds might be too loud. We use a lot of custom sounds here and when the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and clicks. Sorry I wasn't clear. This is *always*. I hear it over the call when there's talking and when there's dead silence (e.g., an empty MeetMe room). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clicking sound with alaw codec
When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come to mind in no particular order are: Remember: this is only *one* particular SIP trunk. Use Wireshark to see the difference between a good call and a bad one. If you see a lot of time jumps on the bad call then look at your network/QoS. jumps? Note that a good call is G.729 and bad is G.711, so I wouldn't expect them to be at all similar. We throw a lot more bandwidth than even G.711 down the pipe between the two sites in terms of data each evening, so I don't think it's that kind of issue. I'm thinking in terms of distortion caused by transcoding someplace. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clicking sound with alaw codec
- jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371424, src=RTP [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371434, src=RTP -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clicking sound with alaw codec
Check https://issues.asterisk.org/jira/browse/ASTERISK-12042 I did. But that was with an unofficial G.729. This is with the supplied alaw codec. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with 'i' extension
I'm running Asterisk 10.7.1. In the log, I see:
-- Goto (Conferences,70323,1)
-- Auto fallthrough,
But there is an 'i' extension:
dialplan show i@Conferences
[ Context 'Conferences' created by 'pbx_config' ]
'_[ti]' =1. GotoIf($[${SET(REC=$[${REC}--1])}3]?999) [pbx_config]
2. Set(EFN=conf-invalid) [pbx_config]
3. Goto(200,1)[pbx_config]
What's going on? Shouldn't this go to that extension?
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[asterisk-users] Uninitialized variable in main/pbx.c?
I think the below fixes what I reported earlier. Does that seem right? *** pbx.c.old 2013-01-23 21:08:51.0 -0500 --- pbx.c 2013-01-23 21:09:31.0 -0500 *** static enum ast_pbx_result __ast_pbx_run *** 5160,5163 --- 5160,5165 int timeout = 0; + dst_exten[0] = '\0'; + /* loop on priorities in this context/exten */ while (!(res = ast_spawn_extension(c, c-context, c-exten, c-p riority, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uninitialized variable in main/pbx.c?
+ dst_exten[0] = '\0'; Is this 'construct' prefered over dst_exten[0] = 0; or *dst_exten = 0; and why? I'm somewhat of a C pedant here. dst_exten is declared as an array, not a pointer. So if I want to clear the first byte of the array, I'll use array syntax pretty consistently. If it's a pointer, I tend to prefer the pointer syntax, unless I'm also doing something with other than the first byte. So I wouldn't write: *x = 'a'; x[1] = '\0'; but instead x[0] = 'a'; x[1] = '\0'; And I certainly don't like using 0 when I mean the null character, at least not in an assignment. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any timeframe for the release of the Asterisk 11-Lumenvox connector bridge?
I'm starting to think about migrating from an old Asterisk box to a new one and want to use the Asterisk 11 long term support release, but need Lumenvox integration and I don't see the Asterisk 11 connector bridge for Lumenvox available yet. Lumenvox tech support says this is under Digiums control. Can anyone give an idea of how soon it'll be available? I will need this as well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
In this properly trimmed example, there's no record of who said what. When it's relevant, I trim in such a way that that information is preserved. But I would *never* leave in a header, just the identification of the person who typed that part. Most mailers, when you include text from another email, put someting like XYZ wrote: before the included text. So usually it's just a matter of preservating that and adding any that are needed that aren't there. Yes, it takes a few minutes longer, but given that there are probably hundreds of people reading my email, that's an investment that I find *well* worth it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on Confbridge menu item dialplan_exec
I like the example of using that to add somebody to the conference, but what I don't see is how the dialplan can know what conference the menu item was called from. I was hoping that some variable might have been set, but don't see it in the sources. Is the idea to do that outside of the call to Confbridge? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Speex codec
I'm trying to convert from MeetMe to Confbridge and one part of that is
handling the ending of a conference. So I'm taking the suggestion of
originating a call to the conference and doing:
same = n,Playback(conf-will-end-indigits/${WTIME}minutes)
That crashes Asterisk (with no core dump!) in the default configuration.
When I run it manually, I see the error message:
Fatal (internal) error in kiss_fft.c, line 294: KissFFT: max radix supported is
17
If I unload module codec_speex.so, everything works. If I playback files
other than conf-will-end, it also works.
Two questions:
(1) Why is that codec being used in the first place?
(2) Why it is generating that error when it is?
This the Asterisk 10.7.1 release.
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Re: [asterisk-users] Top Posting
I realize the benefits of bottom-posting, especially when posting inline. But top-posting keeps things in reverse chronological order so any reader could catch up quickly on any missed messages in the chain. A new reader scrolls to the bottom and reads up. What's there to catch up with if you don't first read what the person is replying to? Do you think that everybody remembers every thread. Of what value is it to see something like No, that didn't work. *before* a description of what it was that didn't work. When people reply to an email, it's their responsibility, whether they top-post or bottom-post to remove unnecessary old message and keep just what's necessary to understand the email. One of the problems with top-posting is that it makes it easier to forget to do this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
What's the configuration like for Jitsi in sip.conf? Just fullname and md5secret plus a phones section that reads: [phones](!) type=friend host=dynamic context=SIP_Phones cc_agent_policy=generic cc_monitor_policy=generic disallow=all allow=gsm allow=ulaw allow=g729 allow=h264 What version of Asterisk? 10.7.1 What does the SIP signaling look like? I don't follow. It's just the standard INVITE/Ring/OK. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
What NAT settings are globally in use? nat=yes Do you have directmedia turned off or on? I've tried both ways, but I normally have it off. This really does indeed feel like a weird NAT issue that is probably configuration related (probably both in Jitsi and Asterisk). Except that: (1) It *works* when there's NAT and *doesn't* work when everything has a static IP. (2) I see the RTP packets arriving: if it were NAT, I'd expect *not* to see them. (3) It depends on the direction of the call and on whether it's SIP-SIP or DAHDI-SIP (and directmedia is off). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
Yeah this is so weird that packet captures are really needed. A working call and a non-working call, along with what IP ranges are what. There are *tremendous* numbers of RTP packets, of course. Are those captures really going to be useful? That's the problem. If they *are* going to be useful, then how many packets should I save? I did look at the sip debug output, as I said, and those look the same. I ran into this on a machine that I won't be at for another two weeks, but I can see if I can reproduce it on similar machine. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I'll try to reproduce on this machine and send that off. However, I did look at the SIP signaling and src/dst IP addresses and they're all as expected between the two calls: I really fear that the difference is in the contents of the RTP stream. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I reproduced it on this machine. It's a total of only 1293 packets, taken on this end. First call didn't work: I heard nothing coming inbound. Second call worked, well enough that there was feedback (both phones and the desktop were in the same room). You can find the file at: http://www.gnat.com/~kenner/wierdAsteriskJitsi.pcap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
1. Remove allow=gsm from your sip.conf and reload That did it! Thanks! But why should that have been an issue? 2. Disable ZRTP in Jitsi by going into Options - Accounts - Selecting account - Edit - Security - Uncheck Enable support to encrypt calls. That was one of the first things I tried a few days ago. No change. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd RTP issue
The way you had things configured Asterisk was prioritizing GSM over ULAW, so until Jitsi started responding it sent GSM. I thought I might have seen something like that in the packets, but it didn't look like it showed up in the SDP negotiations, so seemed peculiar to me. Unclear why this only happens with a static IP and not NAT, but oh well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wierd RTP issue
I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it's connected via NAT, all is OK. When it's connected with VPN, the following occurs: The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it's calling. The voice path inbound to Jitsi works fine when it's called from another SIP device. The voice path inbound to Jitsi is silent when it's called from something on the other side of a PRI via DAHDI. I've run Wireshark on my desktop and see the RTP packets coming at the same rate and protocol (g711) in all the cases and sip set debug peer xyz doesn't shed any light on the situation (the SDP data looks similar in the working and non-worknig cases). Does anybody have any ideas what to look at next? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd cracking with SIP-DAHDI
We recently set up a SIP trunk between an office in NY running Asterisk and an office in Paris (running Alcatel). All works fine if a SIP phone on the NY system talks to the Paris PBX. But if something on DAHDI (a PRI or MeetMe) talks to the Paris PBX, there's a low-volume crackling. This isn't clipping because it also occurs when there's no legitimate sound. It's sort of a mild version of what you used to get when a POTS pair had a ground short. This occurs no matter what size originates the call. pings show round trip times of around 100ms, ranging from around 200 to 80 ms. Packet loss is zero. The fact that SIP-SIP works fine suggests the issue isn't related to IP issues. I tried adding a jitter buffer, but that didn't make a difference. I've tried this sending just ULAW and G722 and allowing everything, but no difference. The SDP that comes back from Paris doesn't list any audio codecs and is: v=0 o=default 1350406175 1350406175 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 0 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 m=video 0 RTP/AVP 31 34 34 98 99 104 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:34 H263/9 a=rtpmap:98 h263-1998/9 a=rtpmap:99 H264/9 a=rtpmap:104 MP4V-ES/9 a=sendrecv Does anybody have any ideas as to what I should look at next? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd cracking with SIP-DAHDI
cat proc/interrupts? http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards I'm sorry that I wasn't clear: the PRI is fine. It's been in use for years and hasn't caused any problems. What's new is the SIP connection between the two offices. And another datapoint: the problem only happens for ulaw and alaw, not g729. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd cracking with SIP-DAHDI
I seem to recall seeing somewhere recently where there was a bugfix for ulaw/alaw conversion which would cause poor audio. Hmm. You mean: https://issues.asterisk.org/jira/browse/ASTERISK-1323 That was quite old, but that is what the noise sounds like. Have you tried updating your Asterisk to the latest of whatever major version you are running? I'm running 10.7.1, which is pretty new. I'd prefer not to upgrade unless I know it'll fix it because of the work involved. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on Asterisk memory management
I'm trying to add a Talking: field to the AMI ConfbridgeList event so that my conference room monitoring will work with Confbridge instead of having to stay with MeetMe and there's something I don't understand. When app_confbridge.c calls ast_bridge_features_set_talk_detector, it passes a *copy* of args.conf_name. Why make the copy? Isn't args.conf_name in valid memory throughout the existance of that bridge? I ask because the easiest way to do what I want is to change that parameter to be conference_bridge_user and add a talking field to it (yes, I know I then have to have the callback called unconditionally and test TALKER_DTETECT there). But that can't work if there's a scoping issue with memory and the copy suggests there is, though I don't see it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10.9.0-rc1 : Help with GoSubIf Parsing
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
I've tried with and without spaces the = sign. Same Result. I've
counted my parens and braces.
If there *is* a caller-ID, it should work without spaces. But not if
there isn't. The proper test is:
$[x${CALLERID(num)}=x2024324321]
And this only works if you're *sure* that it'll be just numbers or blank.
Otherwise, use quotes on both sides.
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[asterisk-users] Questions on converting to ConfBridge
I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so it becomes very hard to have web applications that show who's in a conference. There also doesn't seem to be a way to lock conferences or mute or kick out users from the dialplan. And the CLI command needs a channel, not a user index, making scripting via the dialplan that much harder. What am I missing? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any workaround for res_speech_lumenvox.so issue?
The latest version of res_speech_lumenvox.so doesn't seem to work and nobody seems to know when a version that works will be available. It looks to me like this is some sort of timeout issue. Does anybody have a workaround to allow this to be used? (I know about UniMRCP, but find it quite heavy.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One-way audio with media_address
I'm migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small patch to allow specifying an address for RTP media. That worked. In 10.7.0, this appears to be built in with media_address, but it doesn't work for me. My Asterisk server has multiple addresses, all global address on two different /24's with different routing policies via BGP. I'm connecting to a phone that's over NAT. I have nat=yes in the general section of sip.conf. Everything works fine with the default. But if I specify media_address to be the Asterisk server's address on the other /24, I get one-way audio. I can see with sip debug that the proper address is being given in the SDP data. Audio from the phone is fine. Audio *to* the phone starts out with maybe 1-2 seconds of very garbled audio, then goes quiet. Running traceroute shows that data comes from the phone *to* Asterisk on the desired /24, but goes out with a source address from the other /24 (the default address). I'm not sure if this is the problem or not, but in any event, I think the source address for RTP should be the one in media_address and want it that way for my purposes anyway. Is there a way to configure this to happen? If not, where should I look to make a patch? And is this likely the reason for the one-way audio or is something else the likely cause? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Repeated Asterisk 10.7.0 crashes
I'm getting cycles of repeated crashes which occur and then stop occurring. Looking at the dumps via gdb shows that something peculiar is happening that looks like memory corruption: Program terminated with signal 6, Aborted. #0 0x003686e30285 in raise () from /lib64/libc.so.6 (gdb) up #1 0x003686e31d30 in abort () from /lib64/libc.so.6 (gdb) up #2 0x003686e6971b in __libc_message () from /lib64/libc.so.6 (gdb) up #3 0x003686e71e7e in _int_malloc () from /lib64/libc.so.6 (gdb) up #4 0x003686e7382d in calloc () from /lib64/libc.so.6 (gdb) up #5 0x0054a2a0 in _ast_calloc (num_structs=1, struct_size=88, field_mgr_offset=64, field_mgr_pool_offset=16, pool_size=128, file=0x101010101010101 Address 0x101010101010101 out of bounds, lineno=1235, func=0x58af9e ast_log) at /usr/src/asterisk-10.7.1/include/asterisk/utils.h:495 495 AST_INLINE_API( Once this starts happening, it seems to keep happening, but Asterisk seems to stay up for hours between the cycles, which I can't reliably stop from cycling. Does anybody have any ideas how to debug this? I suspect it may have something to do with either res_speech_lumenvox (which I got from Lumenvox) or res_speech_unimrcp (which looks to be extremely buggy). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Responsibility for res_speech_lumenvox.so
Who's responsible for it? Lumenvox is the only place that distributes it, but they can't do anything with it since they get it from Digium. However, the current version doesn't work with Asterisk 10.7.1 and the latest version of Lumenvox software (it appears that a timeout is being set to zero). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clipping issue with SIP over satellite
You have hardware echo canceling *outside* of your T1 card? No, on the card. The DAHDI layer has some buffering that can help with jitter, but the default buffers can only handle 80ms of jitter. You can increase this by setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms by default. I'm running 1.6.2 and it appears that this is called jitterbuffers there. Is that right? I've set it to 20 and it did indeed help quite a bit, so I tried 30. It sounds like the lack of a proper jitter buffer (of adequate size) is the issue here, since when the audio is directed at endpoints outside of Asterisk that have them, the audio is as you'd expect it to be. Interestingly, that isn't completely true. If it goes out a SIP trunk to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300 (where the T1 goes), it has the same problem. This was leading me to believe that the problem was on the 8300. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clipping issue with SIP over satellite
You have hardware echo canceling *outside* of your T1 card? No, on the card. Then you definitely don't want 'echocancel=no' set, or you'll disable it. When I thought that it was echo cancellers fighting each other, that's exactly what I wanted to do. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Clipping issue with SIP over satellite
I'm having a wierd clipping issue with one employee who's using a phone over a satellite Internet. He was sold that system specifically for use with VoIP. Ping times show average round-trip time as around 700 ms with a range of 560 to 841, so considerable jitter. Things work fine when he's talking to another Asterisk phone or to a SIP trunk provider, but when connecting to a T1, there's clipping where about 1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds like an echo canceller conflict, but I've set echocancel=no in chan_dahdi.conf (I have hardware echo cancelling) and it didn't do anything. I'm forcing his codec to G729 for bandwidth reasons. The phone is an Aastra 6757iCT. Does anybody have any suggestions here? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Licensing question.
But so long as you were careful not to copy any of the code you are going to link against into your Source Code (and why would you, if you were linking against it?), it only *becomes* a derivative work *after* it has been compiled. That's not necessarily true because if you have a work that cannot be used independently (e.g. a plug-in), there are numerous court precedents that say that it indeed is a derived work. This area of the law is very complex and people should really consult an attorney experienced in this area if they care about such things. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
Can please the Powers that Be reconsider and add this option to sip.conf? What Powers that Be? This is open-source software! If you need an option in sip.conf, just add it! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get second cipher in an extension
how can I get the second character/cipher of an extension ?
If I have : exten = 12345,n,NoOP()
How can I get 2 ?
${EXTEN:1:1}
--
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Re: [asterisk-users] Free CNAM
FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 I just tried this on about a dozen numbers I have in various parts of the US (cell, business, and a landline number I've had for decades) and NONE of them were listed in this database. Indeed I can't find one that IS. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
