Re: [asterisk-users] Substitute Macro() for Gosub in dialplan

Thank you Thank you Thank you! I changed it to: exten => s/555333,1,Gosub(subBusy,s,1()) and it now works like a charm. Really appreciate the help! El sáb, 20 nov 2021 a las 10:55, escribió: > On 11/20/2021 11:51 AM, Richard Reina wrote: > > Since Macro is deprecated

[asterisk-users] Substitute Macro() for Gosub in dialplan

Since Macro is deprecated I am trying to eliminate it from my diaplan. I believe I have successfully done so in the example below. ; dial an internal extension exten => 101,1 Macro(ext,100,Dahdi/15) TO: exten => 101,1,Dial(Dahdi/15,30) So far it seems to work. However I also in my dialplan

Re: [asterisk-users] CURLOPT(useragent) fails with Set requires an '=' to be a valid assignment

There are semicolons in the useragent string you are trying to set. If that is the exact dialplan line then those semicolons are being seen as a start of a comment. Richard On Mon, Dec 14, 2020 at 12:25 PM Jonathan H wrote: > All my other CURLOPT settings like timeout work f

[asterisk-users] Digium TE134 compatibility issues with new Dell server - Zero interrupts

I am getting zero interrupts for a new Digium TE134 Card on a new brand new Dell T40 server with the latest BIOS. Is there something that I am missing or is the card not compatible with Dell servers? (cat /proc/interrupts ; sleep 1 ; cat /proc/interrupts) | grep -i wcte13xp0 16: 0

Re: [asterisk-users] Fwd: call-id on cdr's

use it in the dialplan? > > Thank you in advance > The function CHANNEL(callid) returns exactly what you want. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CHANNEL Richard -- _ -- Bandwidth and Colocation Pro

Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

Argh. That was for chan_pjsip and you are using chan_sip. Be aware that chan_sip is effectively dead. Richard On Thu, May 14, 2020 at 9:50 AM Richard Mudgett wrote: > The other end is sending g729 even though it was not negotiated. The > other end should not do this and it usually

Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

The other end is sending g729 even though it was not negotiated. The other end should not do this and it usually seems that the other ends that do send g729. This was recently fixed. See https://issues.asterisk.org/jira/browse/ASTERISK-28139 Richard On Thu, May 14, 2020 at 1:11 AM John Hughes

[asterisk-users] [asterisk-app-dev] True suppression of DTMF from audio

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP

Re: [asterisk-users] Looking for sample hangup_handler_pop and _wipe using vars

l the test extensions to see what is on the channel's hangup handler stack while the channel is in the Echo application by using the command line commands mentioned on the wiki page. Richard On Mon, Feb 3, 2020 at 7:26 PM David P wrote: > Please point me to samples of popping and wiping hangup

Re: [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?

1 root root 887 nov. 18 20:46 asterisk.key > -rw--- 1 root root 2111 nov. 18 20:47 asterisk.pem > I'd say that asterisk running as the asterisk user has no permission to see the .pem file as only root can see it. Richard > -rw--- 1 root root 161 nov. 18 20:46 ca.cfg > -rw--

Re: [asterisk-users] Stuck "channel"

P MESSAGE requests in the dialplan. It cannot be hung up. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New

Re: [asterisk-users] pattern matching "+"

ions? > You must have multiple patterns to match the various starting sequences you receive. One that begins with + One that begins with 1 One that is for a 10 digit number Richard -- _ -- Bandwidth and Coloca

Re: [asterisk-users] Outbound caller ID ignored

uffers => 12,half > channel => 49-53 > > It will not work. This is a limitation of analog lines. In this case your Asterisk box is pretending to be a POTS phone. Phones do not tell the PSTN who they are as the PSTN already knows who they are. Richard -- __

Re: [asterisk-users] Connected line update prevented

message is not an error. It is a verbose log stating what it did. It is a result of you telling the Dial application to block the initial connected line update because you set Dial's 'I' option flag. Richard -- _ -- Bandwidth a

Re: [asterisk-users] pjsip aor stays in status created

; > > do you think if this can be bug? > It is not a bug. The contact has been "created". It will stay in that state unless you are also going to qualify the endpoint. Asterisk 16 simply renames the state to "NonQualified" to be more explicit. Richard

Re: [asterisk-users] After updating to 16 "Some non-required modules failed to load"

On Tue, Oct 23, 2018 at 5:07 PM Jonathan H wrote: > Thanks Richard - any idea if these matter? And how to stop the errors: > > cdr_sqlite3_custom declined to load. > cel_sqlite3_custom declined to load > pbx_ael declined to load > > Standard 16.0 build, just updated a

Re: [asterisk-users] After updating to 16 "Some non-required modules failed to load"

; is needed when using chan_sip and res_pjsip_transport_websockets on ; the same system. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon i

Re: [asterisk-users] Asterisk 16.0.0 Now Available

ef=" > http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-15-current.tar.gz > ">download now > But when the page is loaded it downloads the new version. > Should be fixed. Richard -- _ -- Bandwi

Re: [asterisk-users] First attempt with statsd

es instructions on how to build them. https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source > > 2. On a general point of view, is collectd daemon extended with a statsd > plugin able to collect Asterisk Statsd statistics ? > I don't know. Richard > > [

Re: [asterisk-users] Explain module reloading error message

xist > ? > The reload message is incorrect when the statsd.conf file has not changed. I have just put up a patch on gerrit [1] to fix it. There was another fix [2] made about a week ago that fixed a more general problem that affected the reported reload status of any module if it failed. Richard [1

Re: [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"

for chan_sip's behavior and chan_pjsip's asymmetric_rtp_codec=no option is because phone DSP's can only handle a single codec at a time. Technically if the peer's INVITE offered five and we responded with three the peer should immediately renegotiate to narrow the codec choice to one. Richard

Re: [asterisk-users] Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED]

the uniqueid of the oldest associated channel within the Asterisk box. Uniqueid's are unique within an Asterisk box and can be made unique across Asterisk boxes by optionally adding the host name. Richard -- _ -- Bandwidth and Colo

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

pairs. When you dial a local channel it is the ;1 channel that acts as an outgoing channel and the ;2 channel executes dialplan and acts as an incoming channel. This is the initial role between the two channels in a local channel pair. In the transfer scenario I describe above (the an

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

ll shows that DYNAMIC_FEATURES is set. It's just not accessible. > > Any thoughts? > It likely depens on how you are doing the attended transfer. Via DTMF? Via SIP or channel technology protocol? Does the Agent B channel have the DYNAMIC_FEATURES channel variable set on it? Rich

Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)

core set debug atleast X That is why the debug level does not go down. Another thing is that the debug level is global to the system. Thus if you set the level in one connection it affects all connections including future ones. The verbose level is per connection. Richard -- ___

Re: [asterisk-users] Any way of "flattening out" 2 channels back into one?

rmation goes away when the local channels optimize out. The Dial 'L' option currently puts state on the caller and called channels depending on which features are configured (who hears things). If you set the verbose level to 4 you get information in the log about that. Richard [1] https:

Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)

se loggers, they are now only logged as debug messages. Richard -- _ -- 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

Re: [asterisk-users] Recompiling Ast results in a binary with differing SHA256 sums?

ely different "asterisk" ELF binary each time I > recompile asterisk, according to checksum? > > Can someone shed light... > A timestamp is added to the version string when you build Asterisk. Thus every time you recompile Asterisk you get

Re: [asterisk-users] Passing arguments to the 'mailcmd' option in voicemail.conf

d from stdin. ( mailcmd < temp-email-body-file ; rm -f temp-email-body-file ) & Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://com

Re: [asterisk-users] Core show channels concise = deprecated

owChannels Or you can go to the wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AMI+Actions https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_CoreShowChannels Richard -- _ -- Bandwidth and Colocation Pr

Re: [asterisk-users] Asterisk crashing on AAAA lookup

nywhere. I > am using Asterisk 15.4.1. > You have to start asterisk with the -g option to make asterisk create core files. https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Richard -- _ -- Bandwidth and Colocation

Re: [asterisk-users] How to check modules loading order or force such order ?

The most-up-to-date and accurate option documentation for your Asterisk version will be what is installed online with your Asterisk installation. In this case CLI "config show help res_pjsip global endpoint_identifier_order", and "core show help pjsip show identifiers".

Re: [asterisk-users] Alias for country in indications.conf

is to define a tone event called alias with a tone cadence of "gb". Richard On Mon, Apr 23, 2018 at 1:08 AM, Patrick Wakano <pwak...@gmail.com> wrote: > Hello list, > Hope you all doing fine! > I've tried to use the 'alias' directive in the indications.conf file but >

Re: [asterisk-users] Asterisk / PRI and Outbound Overlap Dialing

are used to create the specified channels on the next line. channel=1-15,17-31 ; Any options set AFTER the channel line above DO NOT affect those channels. context=other Richard -- _ -- Bandwidth and Colocation Provided by http://ww

Re: [asterisk-users] Strange problem with PRI on 64-bit?

On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson wrote: > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield > wrote: > > In article

Re: [asterisk-users] Which CDR processing for high load ?

n cdr-scv/Master.csv > file ? > Before CDRs get written to the back ends (i.e., permanent storage) they are in memory data structures. Where else could they be before getting written to the back ends? Richard -- _ --

Re: [asterisk-users] Which CDR processing for high load ?

r thread to write to the back ends. You need to be using at least v13.19.1 or v15.2.1 to also have some CDR performance enhancements to help CDR processing of call events. For information about ODBC connection pooling performance problems see [2]. Richard [2] http://blogs.asterisk.org/2016/

Re: [asterisk-users] pjsip trunking configuration issue

aor-single-reg](!) > type=aor > max_contacts=20 > > [1001](endpoint-basic) > auth=auth1001 > aors=1001 > > [auth1001](auth-userpass) > password=password123 > username=1001 > > [1001](aor-single-reg) > > > Extensions.conf > > [from-twilio] &g

Re: [asterisk-users] Running on virtual machine and audio not intelligable

mixing. ConfBridge does not need DAHDI since it does its own mixing in Asterisk. As for timing sources you have several to choose from of which DAHDI is one of them. See menuselect res_timing_xxx modules. Timing is really only needed when playing back sound prompts when nothing is

[asterisk-users] asterisk not transcoding on a re-invite codec change

I have an old setup based in Asterisk 1.8. The carrier is accepting Ulaw in the initial invite, but immediately the call is established they send a re-invite to change to Alaw. This doesn't get transcoded and the user gets no audio from after the re-invite Is/was this a problem for asterisk

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

Re: [asterisk-users] asterisk queues in off-hook mode ?

ed to the queue which should > distribute to connected agents. is this possible on teh actual > app_queue or we would need to implement it using ARI. > > Thanks in advance. > You need to use app_queue[1] with app_agent_pool[2][3][4] for your agents. Richard [1] https://wiki.aster

Re: [asterisk-users] How can I check backtrace files ?

iler can optimize out variables that could make understanding what is going on harder. So it depends upon what happened if an optimized backtrace can help find the root cause or not. It is up to you whether you want to run in production with an optimized build or not. I also recommend

Re: [asterisk-users] Chan Local, Originate and slin

nager API > Can anybody explain how the native format is chosen in these cases? > Version 13.1 is a very old version of Asterisk 13. The current version of Asterisk 13 is 13.18.2. I also recall an issue where local channels tended to use slin192 when there was no need. Howev

Re: [asterisk-users] How to correctly set REDIRECTING to indicate diversion reason

On Tue, Nov 21, 2017 at 5:04 AM, Benoit Panizzon <benoit.paniz...@imp.ch> wrote: > Hi Richard > > Thank you > > > You need to set more redirecting information [1]. > > > > In sip.conf send_diversion=yes needs to be in effect. You also need > > to setup &

Re: [asterisk-users] How to correctly set REDIRECTING to indicate diversion reason

id information (at least the from number) to indicate where you are redirecting from. You should also increment the redirecting count. chan_pjsip has the same requirements. pjsip.conf send_diversion=yes needs to be in effect and you also need to setup the from party id information. Richa

Re: [asterisk-users] Ringing (180) no SDP to progress(183) with SDP transition => no audio.

k that case was because the device was converting ISDN to SIP. I do think that the devices that don't stop local ringback in favor of the incoming RTP stream following the 183 are broken. Unfortunately it is something that is out of your control. Richard -- _

Re: [asterisk-users] PJSIP console messages with Zoiper

l suppress them on the console. Richard -- _ -- 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:

Re: [asterisk-users] Return without Gosub: stack is empty

he dial application. This dialplan needs to be able to distinguish between the two channels and act accordingly. Using the F() option with a dialplan location is the simplest way to distinguish between the two channels. Richard -- _

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

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

Re: [asterisk-users] Gerrit usage?

quot; based off of the currently checked out "master" branch. This "13" branch is just another branch of master and not a real 13 branch. When you tried to put it up for review to the real 13 branch using "git review 13", git tried to merge your master "13" br

Re: [asterisk-users] Interpreting pjsip.conf

tell pjsip how to send and receive SIP messages. The provided samples may or may not apply to your particular network. * The name of a transport section is completely arbitrary. What makes it a transport section is the "type=transport" line. [1] https://wiki.asterisk.org/wiki/di

Re: [asterisk-users] ERROR during high volume MoH dialplan

e more than 4k simultaneous calls. > * There is no user configurable option to change the excessive ref count trigger value. However, you could change the EXCESSIVE_REF_COUNT define value in the main/astobj2.c file and recompile. Richard --

[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

Re: [asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

On Mon, Aug 28, 2017 at 6:35 PM, Richard Kenner <ken...@gnat.com> wrote: > 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

[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) {

Re: [asterisk-users] ERROR during high volume MoH dialplan

lls/sec and the calls lasting 8 seconds that comes to 4000 active channels. Hitting the FRACK would result in an average of 25 references to the format per channel. This is a simplistic calculation as there are going to be some references that have nothing to do with a call. The number o

Re: [asterisk-users] Detecting DoS attacks via SIP

ch for PeerStatus, but since there's no actual peer in the > attack, I don't seem to get an event from AMI. > > Any ideas? > There is an AMI security class that you can use to monitor the AMI security events. See manager.conf.sample Richard -- __

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

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

Re: [asterisk-users] MoH via AGI broken after upgrade.

sion 13? > > Any advise would be welcome. > The SetMusicOnHold application was deprecated in v1.6 and removed in v13. Use Set(CHANNEL(musicclass)=class) instead to set the music class on the channel. The change was documented in the UPGRADE.txt files. Richard -- ___

Re: [asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?

> > Related, Why can we have multiple Hangup handlers but not Pre-Dial > handlers? > * There is only one dial to execute the called channel pre-dial handler while there are many opportunities to specify hangup handlers. * How do you think you could associate different pre-dial ha

Re: [asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

an > incompatible "codec" on both legs so it shouldn't switch to direct > media. > > Has anyone else seen this issue? > This is an old issue. One of the latest issues is: https://issues.asterisk.org/jira/browse/ASTERISK-25166 Richard -- ___

Re: [asterisk-users] Difference between Application Set and Function SET?

> You are declaring an extension line with a pattern but the pattern only has literal characters so it really isn't a pattern. It takes more CPU to match than the non-pattern form and is more likely an error. Richard [1] ht

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

Re: [asterisk-users] Difference between Application Set and Function SET?

ked anywhere a function can be invoked and not just in dialplan. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/

Re: [asterisk-users] Surrogate channels

surrogate channels are replacement channels for masquerades to swap with your target channel. They are created to die after a masquerade has substituted it for the target channel. If you are seeing them in dialplan then just let them die without doing anything else with them.

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

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

[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? -- _

Re: [asterisk-users] ** in extensions.conf

ead of the ** and that works fine. > > Is there anyway to get the ** to work? I also am using a polycom phone if > that affects things. I'm using asterisk 13.15.0 > A ** extension should work just fine. I expect it is the dialplan in the

[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

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

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

[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,

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 } !

[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

Re: [asterisk-users] how to hangup this channel "Message/ast_msg_queu

context pjsip config option. The channel is reused to process each message in dialplan. It is invalid for dialplan to do anything with media on that channel and VoiceMail definitely falls into that category. Richard https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend ht

[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

Re: [asterisk-users] ConfBridge function slight change from 11 to 13

ConfBridge or https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_ConfBridge config show help app_confbridge user_profile template or https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge Richard -- _

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

[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

Re: [asterisk-users] BUG or ???

quot;http://sIte.com:80/api/v1/ > calls?apiKey=UABVAEI=3")} > executes and get answer from the server [{"RequestedCount":0," > MissedCount":7,"Total":7}] > The Set isn't being executed by the ExecIf. However the ${} substitution containing the

Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

n Asterisk server in production with > DEBUG_THREADS enabled ? > No, you should not leave DEBUG_THREADS on in a normal production environment. Only enable it when you are actually hunting for a deadlock. DEBUG_THREADS causes a noticeable drop in performance. Richard --

Re: [asterisk-users] Beep on Attended Transfer

> Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound file to play on a transfer. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: h

Re: [asterisk-users] Execution of pre-bridge handlers

a pre-bridge handler (or any handler for that matter) is a bad thing to do and definitely falls into the "undefined behavior" category. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the n

Re: [asterisk-users] pjsip asterisk 13 add time between DTMF digits

uration and its > interval? > >From the CLI do: core show application SendDTMF Right there in the documentation it says you can specify the intra-tone time and the DTMF duration. You can also look on wiki.asterisk.org for the SendDTMF

Re: [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?

on should be able to use all of the current libpri features. Only DAHDI would really care about the kernel version and I cannot say if that kernel is supported with the latest DAHDI. The easiest way would be to try compiling it. Richard -- __

Re: [asterisk-users] IMPORT from bridged Local channels not importing.

bout in this scenario I describe (four contexts involved)? > In your case, the h exten is run by SIP/origin and Local/dest_ext_num;2. These channels executed dialplan when the call was originally placed. The h exten runs on the respectiv

Re: [asterisk-users] What to do when changing from one asterisk version to another ?

>> > > Correcting myself, make uninstall seems to be what I was after for > Asterisk itself. > I'm still searching for the equivalent make target for pjproject. > pjproject has a make uninstall target as well. Since v13.8, Asterisk has a --with-pjproject-bundled option [1].

Re: [asterisk-users] Non-global variable that follows channel?

On Sun, Nov 27, 2016 at 11:13 AM, Jonathan H <lardconce...@gmail.com> wrote: > Thanks, Richard - your code does indeed work reliably 100% of the > time, and thank you for that explanation. > > I do think the docs at > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Fu

Re: [asterisk-users] Non-global variable that follows channel?

sing variable inheritance. [svtest1] exten = s,1,NoOp() same = n,Answer() same = n,Set(__MY_CALLER=${CHANNEL(name)}) same = n,Dial(Local/s@svtest2,,g) same = n,NoOp(Returned SHARED(sharedVar) = '${SHARED(sharedVar)'} same = n,Hangup() [svtest2] exten = s,1,NoOp()

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

and Local/s@dial-dest;1 When Local/s@dial-dest;2 executes Answer it will allow Local/s@dial-test;1 and ;2 to optimize out because both ends are in a bridge. Thus the H dial option will disappear from the channel chain. Richard -- __

Re: [asterisk-users] res_pjsip parkinglot configuration?

tting but I don't see an equivalent in pjsip.conf > > Do I need to use setvar to set CHANNEL(parkinglot) on my endpoint to do > this now? > Yes. In your type=endpoint section you specify set_var=CHANNEL(parkinlot)=mylot Richard -- _

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On Sat, Sep 10, 2016 at 5:18 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote: > On 10-09-16 09:42, Jonas Kellens wrote: > > > On 10-09-16 00:50, Richard Mudgett wrote: > > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens <jonas.kell...@telenet.be> &

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

g > information about queues (I don't see this message on any other command). > That message is a result of trying to build a string where the buffer is too small to contain it. I would expect that there is a truncated string in the 'queue show' output. You haven't stated which Asteri

Re: [asterisk-users] No ringback heard

t is ringback with a recording interspersed at intervals, you can create a music-on-hold class and have the caller hear that instead. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Comm

Re: [asterisk-users] Toll free pattern matching

finds priority 3 of the generic series. 3, third in generic series There is no priority 4 so the call is hung up. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities [2] https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching --

Re: [asterisk-users] Asterisk 13 High CPU usage

pjsip. Also you should look here for more information: http://blogs.asterisk.org/2016/07/13/asterisk-task-processor-queue-size-warnings/ Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to A

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